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[alsa-devel] [PATCH 18/24] sound/pci: coding style fixes: rme9652
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/rme9652/hdsp.c | 530 +++++++++++++++++++++--------------------- sound/pci/rme9652/hdspm.c | 114 +++++----- sound/pci/rme9652/rme9652.c | 146 ++++++------ 3 files changed, 395 insertions(+), 395 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d723543..de05602 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -41,7 +41,7 @@ #include <asm/byteorder.h> #include <asm/current.h> -#include <asm/io.h> +#include <linux/io.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ @@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); 26*3 values are read in ss mode 14*3 in ds mode, with no gap between values */ -#define HDSP_9652_peakBase 7164 +#define HDSP_9652_peakBase 7164 #define HDSP_9652_rmsBase 4096 /* c.f. the hdsp_9632_meters_t struct */ @@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */ #define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */ #define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */ -#define HDSP_SyncRef2 (1<<13) -#define HDSP_SPDIFInputSelect0 (1<<14) -#define HDSP_SPDIFInputSelect1 (1<<15) -#define HDSP_SyncRef0 (1<<16) +#define HDSP_SyncRef2 (1<<13) +#define HDSP_SPDIFInputSelect0 (1<<14) +#define HDSP_SPDIFInputSelect1 (1<<15) +#define HDSP_SyncRef0 (1<<16) #define HDSP_SyncRef1 (1<<17) -#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ +#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ #define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */ #define HDSP_Midi0InterruptEnable (1<<22) #define HDSP_Midi1InterruptEnable (1<<23) @@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_TimecodeSync (1<<27) #define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ #define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */ -#define HDSP_midi0IRQPending (1<<30) +#define HDSP_midi0IRQPending (1<<30) #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) @@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the Multiface still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -460,7 +460,7 @@ struct hdsp { unsigned char qs_in_channels; /* quad speed mode for H9632 */ unsigned char ds_in_channels; unsigned char ss_in_channels; /* different for multiface/digiface */ - unsigned char qs_out_channels; + unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; @@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = { static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ /* Analog */ - 0, 1, 2, 3, 4, 5, 6, 7, + 0, 1, 2, 3, 4, 5, 6, 7, /* ADAT 2 */ - 16, 17, 18, 19, 20, 21, 22, 23, + 16, 17, 18, 19, 20, 21, 22, 23, /* SPDIF */ 24, 25, -1, -1, -1, -1, -1, -1, -1, -1 @@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ - -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; @@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ @@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d static struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, - .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, + .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, .subvendor = PCI_ANY_ID, .subdevice = PCI_ANY_ID, }, /* RME Hammerfall-DSP */ @@ -599,15 +599,15 @@ MODULE_DEVICE_TABLE(pci, snd_hdsp_ids); /* prototypes */ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp); static int snd_hdsp_create_pcm(struct snd_card *card, struct hdsp *hdsp); -static int snd_hdsp_enable_io (struct hdsp *hdsp); -static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp); -static void snd_hdsp_initialize_channels (struct hdsp *hdsp); +static int snd_hdsp_enable_io(struct hdsp *hdsp); +static void snd_hdsp_initialize_midi_flush(struct hdsp *hdsp); +static void snd_hdsp_initialize_channels(struct hdsp *hdsp); static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout); static int hdsp_autosync_ref(struct hdsp *hdsp); static int snd_hdsp_set_defaults(struct hdsp *hdsp); -static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp); +static void snd_hdsp_9652_enable_mixer(struct hdsp *hdsp); -static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) +static int hdsp_playback_to_output_key(struct hdsp *hdsp, int in, int out) { switch (hdsp->io_type) { case Multiface: @@ -624,7 +624,7 @@ static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) } } -static int hdsp_input_to_output_key (struct hdsp *hdsp, int in, int out) +static int hdsp_input_to_output_key(struct hdsp *hdsp, int in, int out) { switch (hdsp->io_type) { case Multiface: @@ -651,7 +651,7 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) return readl (hdsp->iobase + reg); } -static int hdsp_check_for_iobox (struct hdsp *hdsp) +static int hdsp_check_for_iobox(struct hdsp *hdsp) { if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; @@ -670,19 +670,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { unsigned long flags; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + snd_printk ("Hammerfall-DSP: loading firmware\n"); hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); return -EIO; } - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - + for (i = 0; i < 24413; ++i) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { @@ -692,7 +692,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { } ssleep(3); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); return -EIO; @@ -705,15 +705,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { #endif hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); - + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); spin_lock_irqsave(&hdsp->lock, flags); snd_hdsp_set_defaults(hdsp); - spin_unlock_irqrestore(&hdsp->lock, flags); + spin_unlock_irqrestore(&hdsp->lock, flags); } - + hdsp->state |= HDSP_FirmwareLoaded; return 0; @@ -722,7 +722,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) @@ -738,7 +738,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); } else { hdsp->io_type = Digiface; - } + } } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) @@ -754,19 +754,19 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) static int hdsp_request_fw_loader(struct hdsp *hdsp); #endif -static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) +static int hdsp_check_for_firmware(struct hdsp *hdsp, int load_on_demand) { if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { hdsp->state &= ~HDSP_FirmwareLoaded; - if (! load_on_demand) + if (!load_on_demand) return -EIO; snd_printk(KERN_ERR "Hammerfall-DSP: firmware not present.\n"); /* try to load firmware */ - if (! (hdsp->state & HDSP_FirmwareCached)) { + if (!(hdsp->state & HDSP_FirmwareCached)) { #ifdef HDSP_FW_LOADER - if (! hdsp_request_fw_loader(hdsp)) + if (!hdsp_request_fw_loader(hdsp)) return 0; #endif snd_printk(KERN_ERR @@ -786,13 +786,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout) -{ +{ int i; /* the fifoStatus registers reports on how many words are available in the command FIFO. */ - + for (i = 0; i < timeout; i++) { if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count) @@ -824,11 +824,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short if (addr >= HDSP_MATRIX_MIXER_SIZE) return -1; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) { /* from martin bjornsen: - + "You can only write dwords to the mixer memory which contain two mixer values in the low and high @@ -847,7 +847,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short hdsp->mixer_matrix[addr] = data; - + /* `addr' addresses a 16-bit wide address, but the address space accessed via hdsp_write uses byte offsets. put another way, addr @@ -856,17 +856,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short to access 0 to 2703 ... */ ad = addr/2; - - hdsp_write (hdsp, 4096 + (ad*4), - (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + + + hdsp_write (hdsp, 4096 + (ad*4), + (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + hdsp->mixer_matrix[addr&0x7fe]); - + return 0; } else { ad = (addr << 16) + data; - + if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT)) return -1; @@ -902,7 +902,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) if (status & HDSP_SPDIFErrorFlag) return 0; - + switch (rate_bits) { case HDSP_spdifFrequency32KHz: return 32000; case HDSP_spdifFrequency44_1KHz: return 44100; @@ -910,13 +910,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) case HDSP_spdifFrequency64KHz: return 64000; case HDSP_spdifFrequency88_2KHz: return 88200; case HDSP_spdifFrequency96KHz: return 96000; - case HDSP_spdifFrequency128KHz: + case HDSP_spdifFrequency128KHz: if (hdsp->io_type == H9632) return 128000; break; - case HDSP_spdifFrequency176_4KHz: + case HDSP_spdifFrequency176_4KHz: if (hdsp->io_type == H9632) return 176400; break; - case HDSP_spdifFrequency192KHz: + case HDSP_spdifFrequency192KHz: if (hdsp->io_type == H9632) return 192000; break; default: @@ -1027,7 +1027,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -1053,35 +1053,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) there is no need for it (e.g. during module initialization). */ - - if (!(hdsp->control_register & HDSP_ClockModeMaster)) { + + if (!(hdsp->control_register & HDSP_ClockModeMaster)) { if (called_internally) { /* request from ctl or card initialization */ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n"); return -1; - } else { + } else { /* hw_param request while in AutoSync mode */ int external_freq = hdsp_external_sample_rate(hdsp); int spdif_freq = hdsp_spdif_sample_rate(hdsp); - + if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n"); else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) - snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); else if (rate != external_freq) { snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n"); return -1; - } - } + } + } } current_rate = hdsp->system_sample_rate; /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -1089,7 +1089,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) if (rate > 96000 && hdsp->io_type != H9632) return -EINVAL; - + switch (rate) { case 32000: if (current_rate > 48000) @@ -1179,7 +1179,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) break; } } - + hdsp->system_sample_rate = rate; return 0; @@ -1189,7 +1189,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) MIDI ----------------------------------------------------------------------------*/ -static unsigned char snd_hdsp_midi_read_byte (struct hdsp *hdsp, int id) +static unsigned char snd_hdsp_midi_read_byte(struct hdsp *hdsp, int id) { /* the hardware already does the relevant bit-mask with 0xff */ if (id) @@ -1198,7 +1198,7 @@ static unsigned char snd_hdsp_midi_read_byte (struct hdsp *hdsp, int id) return hdsp_read(hdsp, HDSP_midiDataIn0); } -static void snd_hdsp_midi_write_byte (struct hdsp *hdsp, int id, int val) +static void snd_hdsp_midi_write_byte(struct hdsp *hdsp, int id, int val) { /* the hardware already does the relevant bit-mask with 0xff */ if (id) @@ -1207,7 +1207,7 @@ static void snd_hdsp_midi_write_byte (struct hdsp *hdsp, int id, int val) hdsp_write(hdsp, HDSP_midiDataOut0, val); } -static int snd_hdsp_midi_input_available (struct hdsp *hdsp, int id) +static int snd_hdsp_midi_input_available(struct hdsp *hdsp, int id) { if (id) return (hdsp_read(hdsp, HDSP_midiStatusIn1) & 0xff); @@ -1215,7 +1215,7 @@ static int snd_hdsp_midi_input_available (struct hdsp *hdsp, int id) return (hdsp_read(hdsp, HDSP_midiStatusIn0) & 0xff); } -static int snd_hdsp_midi_output_possible (struct hdsp *hdsp, int id) +static int snd_hdsp_midi_output_possible(struct hdsp *hdsp, int id) { int fifo_bytes_used; @@ -1230,13 +1230,13 @@ static int snd_hdsp_midi_output_possible (struct hdsp *hdsp, int id) return 0; } -static void snd_hdsp_flush_midi_input (struct hdsp *hdsp, int id) +static void snd_hdsp_flush_midi_input(struct hdsp *hdsp, int id) { while (snd_hdsp_midi_input_available (hdsp, id)) snd_hdsp_midi_read_byte (hdsp, id); } -static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) +static int snd_hdsp_midi_output_write(struct hdsp_midi *hmidi) { unsigned long flags; int n_pending; @@ -1245,16 +1245,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output) { if (!snd_rawmidi_transmit_empty (hmidi->output)) { if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]); } } @@ -1264,7 +1264,7 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) return 0; } -static int snd_hdsp_midi_input_read (struct hdsp_midi *hmidi) +static int snd_hdsp_midi_input_read(struct hdsp_midi *hmidi) { unsigned char buf[128]; /* this buffer is designed to match the MIDI input FIFO size */ unsigned long flags; @@ -1325,14 +1325,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data) { struct hdsp_midi *hmidi = (struct hdsp_midi *) data; unsigned long flags; - + snd_hdsp_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1435,7 +1435,7 @@ static struct snd_rawmidi_ops snd_hdsp_midi_input = .trigger = snd_hdsp_midi_input_trigger, }; -static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id) +static int snd_hdsp_create_midi(struct snd_card *card, struct hdsp *hdsp, int id) { char buf[32]; @@ -1501,7 +1501,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif); return 0; } @@ -1511,7 +1511,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif; @@ -1530,7 +1530,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream); return 0; } @@ -1540,7 +1540,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif_stream; @@ -1602,7 +1602,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp); return 0; } @@ -1612,7 +1612,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3); @@ -1649,7 +1649,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp); return 0; } @@ -1659,7 +1659,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1693,7 +1693,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp); return 0; } @@ -1703,7 +1703,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1737,7 +1737,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp); return 0; } @@ -1747,7 +1747,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1781,7 +1781,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp); return 0; } @@ -1791,7 +1791,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1828,7 +1828,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_spdif_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1858,7 +1858,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct ucontrol->value.enumerated.item[0] = 9; break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1882,7 +1882,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate; return 0; } @@ -1899,10 +1899,10 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; + uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -1912,7 +1912,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_external_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1940,9 +1940,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru break; case 192000: ucontrol->value.enumerated.item[0] = 9; - break; + break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1968,7 +1968,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp) static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Master", "Slave" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 2; @@ -1981,7 +1981,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp); return 0; } @@ -2018,7 +2018,7 @@ static int hdsp_clock_source(struct hdsp *hdsp) case 192000: return 9; default: - return 3; + return 3; } } else { return 0; @@ -2032,7 +2032,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) case HDSP_CLOCK_SOURCE_AUTOSYNC: if (hdsp_external_sample_rate(hdsp) != 0) { if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) { - hdsp->control_register &= ~HDSP_ClockModeMaster; + hdsp->control_register &= ~HDSP_ClockModeMaster; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); return 0; } @@ -2043,7 +2043,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) break; case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ: rate = 44100; - break; + break; case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ: rate = 48000; break; @@ -2078,13 +2078,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ { static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; if (hdsp->io_type == H9632) uinfo->value.enumerated.items = 10; else - uinfo->value.enumerated.items = 7; + uinfo->value.enumerated.items = 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2094,7 +2094,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp); return 0; } @@ -2104,7 +2104,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2130,7 +2130,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; return 0; } @@ -2165,7 +2165,7 @@ static int hdsp_da_gain(struct hdsp *hdsp) case HDSP_DAGainMinus10dBV: return 2; default: - return 1; + return 1; } } @@ -2180,8 +2180,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_DAGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_DAGainMinus10dBV; - break; + hdsp->control_register |= HDSP_DAGainMinus10dBV; + break; default: return -1; @@ -2193,7 +2193,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2206,7 +2206,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp); return 0; } @@ -2216,7 +2216,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2250,7 +2250,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp) case HDSP_ADGainLowGain: return 2; default: - return 1; + return 1; } } @@ -2262,11 +2262,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_ADGainMinus10dBV; break; case 1: - hdsp->control_register |= HDSP_ADGainPlus4dBu; + hdsp->control_register |= HDSP_ADGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_ADGainLowGain; - break; + hdsp->control_register |= HDSP_ADGainLowGain; + break; default: return -1; @@ -2278,7 +2278,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2291,7 +2291,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp); return 0; } @@ -2301,7 +2301,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2335,7 +2335,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp) case HDSP_PhoneGainMinus12dB: return 2; default: - return 0; + return 0; } } @@ -2347,11 +2347,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_PhoneGain0dB; break; case 1: - hdsp->control_register |= HDSP_PhoneGainMinus6dB; + hdsp->control_register |= HDSP_PhoneGainMinus6dB; break; case 2: - hdsp->control_register |= HDSP_PhoneGainMinus12dB; - break; + hdsp->control_register |= HDSP_PhoneGainMinus12dB; + break; default: return -1; @@ -2363,7 +2363,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2376,7 +2376,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp); return 0; } @@ -2386,7 +2386,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2432,7 +2432,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode) static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp); return 0; } @@ -2442,7 +2442,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2488,7 +2488,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode) static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp); return 0; } @@ -2498,7 +2498,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2576,7 +2576,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd { static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2595,7 +2595,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd uinfo->value.enumerated.items = 0; break; } - + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2605,7 +2605,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp); return 0; } @@ -2615,7 +2615,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; @@ -2664,7 +2664,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) case HDSP_SelSyncRef_SPDIF: return HDSP_AUTOSYNC_FROM_SPDIF; case HDSP_SelSyncRefMask: - return HDSP_AUTOSYNC_FROM_NONE; + return HDSP_AUTOSYNC_FROM_NONE; case HDSP_SelSyncRef_ADAT1: return HDSP_AUTOSYNC_FROM_ADAT1; case HDSP_SelSyncRef_ADAT2: @@ -2680,7 +2680,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 7; @@ -2693,7 +2693,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp); return 0; } @@ -2727,7 +2727,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp_line_out(hdsp); spin_unlock_irq(&hdsp->lock); @@ -2739,7 +2739,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2773,7 +2773,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise) static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->precise_ptr; spin_unlock_irq(&hdsp->lock); @@ -2785,7 +2785,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2819,7 +2819,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet) static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; spin_unlock_irq(&hdsp->lock); @@ -2831,7 +2831,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2873,12 +2873,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem source = ucontrol->value.integer.value[0]; destination = ucontrol->value.integer.value[1]; - + if (source >= hdsp->max_channels) addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination); else addr = hdsp_input_to_output_key(hdsp,source, destination); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr); spin_unlock_irq(&hdsp->lock); @@ -2926,7 +2926,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; + static char *texts[] = {"No Lock", "Lock", "Sync" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2971,7 +2971,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp) int status = hdsp_read(hdsp, HDSP_statusRegister); if (status & HDSP_SPDIFErrorFlag) return 0; - else { + else { if (status & HDSP_SPDIFSync) return 2; else @@ -3007,7 +3007,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp) return 1; } else return 0; -} +} static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3025,17 +3025,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc } static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx) -{ +{ int status = hdsp_read(hdsp, HDSP_statusRegister); - + if (status & (HDSP_Lock0>>idx)) { if (status & (HDSP_Sync0>>idx)) return 2; else - return 1; + return 1; } else return 0; -} +} static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3053,7 +3053,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn break; case Multiface: case H9632: - if (offset >= 1) + if (offset >= 1) return -EINVAL; break; default: @@ -3115,7 +3115,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); return 0; } @@ -3125,7 +3125,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -3170,7 +3170,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = { .get = snd_hdsp_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -3188,7 +3188,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0), HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0), HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), -/* 'Sample Clock Source' complies with the alsa control naming scheme */ +/* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3240,7 +3240,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) return err; } } - + /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */ if (hdsp->io_type == H9632) { for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) { @@ -3259,7 +3259,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3298,7 +3298,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } } } - + status = hdsp_read(hdsp, HDSP_statusRegister); status2 = hdsp_read(hdsp, HDSP_status2Register); @@ -3362,17 +3362,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ: clock_source = "Internal 192 kHz"; - break; + break; default: - clock_source = "Error"; + clock_source = "Error"; } snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source); - + if (hdsp_system_clock_mode(hdsp)) system_clock_mode = "Slave"; else system_clock_mode = "Master"; - + switch (hdsp_pref_sync_ref (hdsp)) { case HDSP_SYNC_FROM_WORD: pref_sync_ref = "Word Clock"; @@ -3397,7 +3397,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref); - + switch (hdsp_autosync_ref (hdsp)) { case HDSP_AUTOSYNC_FROM_WORD: autosync_ref = "Word Clock"; @@ -3410,7 +3410,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_AUTOSYNC_FROM_NONE: autosync_ref = "None"; - break; + break; case HDSP_AUTOSYNC_FROM_ADAT1: autosync_ref = "ADAT1"; break; @@ -3425,14 +3425,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref); - + snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp)); - + snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); - + snd_iprintf(buffer, "\n"); switch (hdsp_spdif_in(hdsp)) { @@ -3452,7 +3452,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "IEC958 input: ???\n"); break; } - + if (hdsp->control_register & HDSP_SPDIFOpticalOut) snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); else @@ -3510,13 +3510,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf (buffer, "SPDIF: No Lock\n"); else snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock"); - + x = status2 & HDSP_wc_sync; if (status2 & HDSP_wc_lock) snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock"); else snd_iprintf (buffer, "Word Clock: No Lock\n"); - + x = status & HDSP_TimecodeSync; if (status & HDSP_TimecodeLock) snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock"); @@ -3524,11 +3524,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "ADAT Sync: No Lock\n"); snd_iprintf(buffer, "\n"); - + /* Informations about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; - + switch (hdsp_ad_gain(hdsp)) { case 0: tmp = "-10 dBV"; @@ -3554,7 +3554,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf(buffer, "DA Gain : %s\n", tmp); - + switch (hdsp_phone_gain(hdsp)) { case 0: tmp = "0 dB"; @@ -3568,8 +3568,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } snd_iprintf(buffer, "Phones Gain : %s\n", tmp); - snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); - + snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); + if (hdsp->control_register & HDSP_AnalogExtensionBoard) snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n"); else @@ -3583,7 +3583,7 @@ static void snd_hdsp_proc_init(struct hdsp *hdsp) { struct snd_info_entry *entry; - if (! snd_card_proc_new(hdsp->card, "hdsp", &entry)) + if (!snd_card_proc_new(hdsp->card, "hdsp", &entry)) snd_info_set_text_ops(entry, hdsp, snd_hdsp_proc_read); } @@ -3632,18 +3632,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax Master clock mode maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - Enable line out. + Enable line out. */ - hdsp->control_register = HDSP_ClockModeMaster | - HDSP_SPDIFInputCoaxial | - hdsp_encode_latency(7) | + hdsp->control_register = HDSP_ClockModeMaster | + HDSP_SPDIFInputCoaxial | + hdsp_encode_latency(7) | HDSP_LineOut; - + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3661,7 +3661,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) hdsp_compute_period_size(hdsp); /* silence everything */ - + for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i) hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN; @@ -3669,7 +3669,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN)) return -EIO; } - + /* H9632 specific defaults */ if (hdsp->io_type == H9632) { hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB); @@ -3687,12 +3687,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) static void hdsp_midi_tasklet(unsigned long arg) { struct hdsp *hdsp = (struct hdsp *)arg; - + if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); if (hdsp->midi[1].pending) snd_hdsp_midi_input_read (&hdsp->midi[1]); -} +} static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) { @@ -3704,7 +3704,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) unsigned int midi0status; unsigned int midi1status; int schedule = 0; - + status = hdsp_read(hdsp, HDSP_statusRegister); audio = status & HDSP_audioIRQPending; @@ -3718,15 +3718,15 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; - + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - + if (hdsp->playback_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); } - + if (midi0 && midi0status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ @@ -3769,10 +3769,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp, if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels)) return NULL; - + if ((mapped_channel = hdsp->channel_map[channel]) < 0) return NULL; - + if (stream == SNDRV_PCM_STREAM_CAPTURE) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else @@ -3910,7 +3910,7 @@ static int snd_hdsp_hw_params(struct snd_pcm_substream *substream, */ spin_lock_irq(&hdsp->lock); - if (! hdsp->clock_source_locked) { + if (!hdsp->clock_source_locked) { if ((err = hdsp_set_rate(hdsp, params_rate(params), 0)) < 0) { spin_unlock_irq(&hdsp->lock); _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE); @@ -3965,7 +3965,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; int running; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; @@ -4016,7 +4016,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) hdsp_silence_playback(hdsp); } - _ok: +_ok: snd_pcm_trigger_done(substream, substream); if (!hdsp->running && running) hdsp_start_audio(hdsp); @@ -4059,10 +4059,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4088,10 +4088,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4170,7 +4170,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_in_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_in_channels, @@ -4201,7 +4201,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_out_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_out_channels, @@ -4318,8 +4318,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; - } - + } + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -4536,7 +4536,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_playbackRmsLevel + i * 8)) return -EFAULT; - if (copy_u64_le(&peak_rms->input_rms[i], + if (copy_u64_le(&peak_rms->input_rms[i], hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_inputRmsLevel + i * 8)) return -EFAULT; @@ -4546,7 +4546,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; switch (cmd) { @@ -4571,9 +4571,9 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_config_info info; unsigned long flags; int i; - + if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); + snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); return -EINVAL; } spin_lock_irqsave(&hdsp->lock, flags); @@ -4601,7 +4601,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp); info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); - + } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4612,7 +4612,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne } case SNDRV_HDSP_IOCTL_GET_9632_AEB: { struct hdsp_9632_aeb h9632_aeb; - + if (hdsp->io_type != H9632) return -EINVAL; h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS; h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS; @@ -4623,7 +4623,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne case SNDRV_HDSP_IOCTL_GET_VERSION: { struct hdsp_version hdsp_version; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; if (hdsp->io_type == Undefined) { if ((err = hdsp_get_iobox_version(hdsp)) < 0) @@ -4639,7 +4639,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_firmware __user *firmware; u32 __user *firmware_data; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */ if (hdsp->io_type == Undefined) return -EINVAL; @@ -4652,25 +4652,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (get_user(firmware_data, &firmware->firmware_data)) return -EFAULT; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0) return -EFAULT; - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; - - snd_hdsp_initialize_channels(hdsp); + + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - + if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n"); return err; @@ -4717,10 +4717,10 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) { struct snd_hwdep *hw; int err; - + if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0) return err; - + hdsp->hwdep = hw; hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); @@ -4728,7 +4728,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) hw->ops.open = snd_hdsp_hwdep_dummy_op; hw->ops.ioctl = snd_hdsp_hwdep_ioctl; hw->ops.release = snd_hdsp_hwdep_dummy_op; - + return 0; } @@ -4752,33 +4752,33 @@ static int snd_hdsp_create_pcm(struct snd_card *card, struct hdsp *hdsp) return 0; } -static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp) +static void snd_hdsp_9652_enable_mixer(struct hdsp *hdsp) { hdsp->control2_register |= HDSP_9652_ENABLE_MIXER; hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); } -static int snd_hdsp_enable_io (struct hdsp *hdsp) +static int snd_hdsp_enable_io(struct hdsp *hdsp) { int i; - + if (hdsp_fifo_wait (hdsp, 0, 100)) { snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n"); return -EIO; } - + for (i = 0; i < hdsp->max_channels; ++i) { hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1); hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1); } - + return 0; } static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { int status, aebi_channels, aebo_channels; - + switch (hdsp->io_type) { case Digiface: hdsp->card_name = "RME Hammerfall DSP + Digiface"; @@ -4791,7 +4791,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS; break; - + case H9632: status = hdsp_read(hdsp, HDSP_statusRegister); /* HDSP_AEBx bits are low when AEB are connected */ @@ -4811,14 +4811,14 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; - + default: /* should never get here */ break; } } -static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp) +static void snd_hdsp_initialize_midi_flush(struct hdsp *hdsp) { snd_hdsp_flush_midi_input (hdsp, 0); snd_hdsp_flush_midi_input (hdsp, 1); @@ -4827,12 +4827,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp) static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp) { int err; - + if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n"); return err; } - + if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n"); @@ -4863,19 +4863,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n"); return err; } - + if (!(hdsp->state & HDSP_InitializationComplete)) { strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - + if ((err = snd_card_register(card)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n"); return err; } hdsp->state |= HDSP_InitializationComplete; } - + return 0; } @@ -4886,7 +4886,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) const char *fwfile; const struct firmware *fw; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp->io_type == Undefined) { @@ -4895,7 +4895,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; } - + /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { case Multiface: @@ -4929,12 +4929,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache)); release_firmware(fw); - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; @@ -4981,14 +4981,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card, hdsp->max_channels = 26; hdsp->card = card; - + spin_lock_init(&hdsp->lock); tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); - + pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; - + /* From Martin Bjoernsen : "It is important that the card's latency timer register in the PCI configuration space is set to a value much larger @@ -4997,7 +4997,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, to its maximum 255 to avoid problems with some computers." */ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF); - + strcpy(card->driver, "H-DSP"); strcpy(card->mixername, "Xilinx FPGA"); @@ -5011,7 +5011,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, } else { hdsp->card_name = "RME HDSP 9632"; hdsp->max_channels = 16; - is_9632 = 1; + is_9632 = 1; } if ((err = pci_enable_device(pci)) < 0) @@ -5040,7 +5040,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) return err; - + if (!is_9652 && !is_9632) { /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); @@ -5063,35 +5063,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; return 0; } else { - snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; - else + else hdsp->io_type = Digiface; } } - + if ((err = snd_hdsp_enable_io(hdsp)) != 0) return err; - + if (is_9652) hdsp->io_type = H9652; - + if (is_9632) hdsp->io_type = H9632; if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; - + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - hdsp->state |= HDSP_FirmwareLoaded; + hdsp->state |= HDSP_FirmwareLoaded; if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0) return err; - return 0; + return 0; } static int snd_hdsp_free(struct hdsp *hdsp) @@ -5107,13 +5107,13 @@ static int snd_hdsp_free(struct hdsp *hdsp) free_irq(hdsp->irq, (void *)hdsp); snd_hdsp_free_buffers(hdsp); - + if (hdsp->iobase) iounmap(hdsp->iobase); if (hdsp->port) pci_release_regions(hdsp->pci); - + pci_disable_device(hdsp->pci); return 0; } @@ -5156,7 +5156,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, } strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); if ((err = snd_card_register(card)) < 0) { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 98762f9..16ca8ab 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -29,7 +29,7 @@ #include <linux/moduleparam.h> #include <linux/slab.h> #include <linux/pci.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/control.h> @@ -84,7 +84,7 @@ MODULE_DESCRIPTION("RME HDSPM"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); -/* --- Write registers. --- +/* --- Write registers. --- These are defined as byte-offsets from the iobase value. */ #define HDSPM_controlRegister 64 @@ -96,10 +96,10 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_eeprom_wr 384 /* for AES32 */ /* DMA enable for 64 channels, only Bit 0 is relevant */ -#define HDSPM_outputEnableBase 512 /* 512-767 input DMA */ +#define HDSPM_outputEnableBase 512 /* 512-767 input DMA */ #define HDSPM_inputEnableBase 768 /* 768-1023 output DMA */ -/* 16 page addresses for each of the 64 channels DMA buffer in and out +/* 16 page addresses for each of the 64 channels DMA buffer in and out (each 64k=16*4k) Buffer must be 4k aligned (which is default i386 ????) */ #define HDSPM_pageAddressBufferOut 8192 #define HDSPM_pageAddressBufferIn (HDSPM_pageAddressBufferOut+64*16*4) @@ -122,15 +122,15 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_midiDataIn1 364 /* status is data bytes in MIDI-FIFO (0-128) */ -#define HDSPM_midiStatusOut0 384 -#define HDSPM_midiStatusOut1 388 -#define HDSPM_midiStatusIn0 392 -#define HDSPM_midiStatusIn1 396 +#define HDSPM_midiStatusOut0 384 +#define HDSPM_midiStatusOut1 388 +#define HDSPM_midiStatusIn0 392 +#define HDSPM_midiStatusIn1 396 /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ #define HDSPM_MADI_peakrmsbase 4096 /* 4096-8191 2x64x32Bit Meters */ @@ -156,7 +156,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); 56channelMODE=0 */ /* MADI ONLY*/ #define HDSPM_Emphasis (1<<10) /* Emphasis */ /* AES32 ONLY */ -#define HDSPM_AutoInp (1<<11) /* Auto Input (takeover) == Safe Mode, +#define HDSPM_AutoInp (1<<11) /* Auto Input (takeover) == Safe Mode, 0=off, 1=on */ /* MADI ONLY */ #define HDSPM_Dolby (1<<11) /* Dolby = "NonAudio" ?? */ /* AES32 ONLY */ @@ -171,7 +171,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_SyncRef3 (1<<25) #define HDSPM_SMUX (1<<18) /* Frame ??? */ /* MADI ONY */ -#define HDSPM_clr_tms (1<<19) /* clear track marker, do not use +#define HDSPM_clr_tms (1<<19) /* clear track marker, do not use AES additional bits in lower 5 Audiodatabits ??? */ #define HDSPM_taxi_reset (1<<20) /* ??? */ /* MADI ONLY ? */ @@ -272,7 +272,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */ #define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */ - /* since 64byte accurate last 6 bits + /* since 64byte accurate last 6 bits are not used */ #define HDSPM_madiSync (1<<18) /* MADI is in sync */ @@ -344,7 +344,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); /* status */ #define HDSPM_AES32_wcLock 0x0200000 #define HDSPM_AES32_wcFreq_bit 22 -/* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function +/* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ #define HDSPM_AES32_syncref_bit 16 /* (status >> HDSPM_AES32_syncref_bit) & 0xF gives sync source */ @@ -403,7 +403,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); /* the size of the area we need to allocate for DMA transfers. the size is the same regardless of the number of channels, and - also the latency to use. + also the latency to use. for one direction !!! */ #define HDSPM_DMA_AREA_BYTES (HDSPM_MAX_CHANNELS * HDSPM_CHANNEL_BUFFER_BYTES) @@ -563,8 +563,8 @@ static inline unsigned int hdspm_read(struct hdspm * hdspm, unsigned int reg) return readl(hdspm->iobase + reg); } -/* for each output channel (chan) I have an Input (in) and Playback (pb) Fader - mixer is write only on hardware so we have to cache him for read +/* for each output channel (chan) I have an Input (in) and Playback (pb) Fader + mixer is write only on hardware so we have to cache him for read each fader is a u32, but uses only the first 16 bit */ static inline int hdspm_read_in_gain(struct hdspm * hdspm, unsigned int chan, @@ -764,7 +764,7 @@ static snd_pcm_uframes_t hdspm_hw_pointer(struct hdspm * hdspm) /* hwpointer comes in bytes and is 64Bytes accurate (by docu since PCI Burst) - i have experimented that it is at most 64 Byte to much for playing + i have experimented that it is at most 64 Byte to much for playing so substraction of 64 byte should be ok for ALSA, but use it only for application where you know what you do since if you come to near with record pointer it can be a disaster */ @@ -832,16 +832,16 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) rate /= 2; /* RME says n = 104857600000000, but in the windows MADI driver, I see: -// return 104857600000000 / rate; // 100 MHz - return 110100480000000 / rate; // 105 MHz - */ +// return 104857600000000 / rate; //100 MHz + return 110100480000000 / rate; //105 MHz + */ /* n = 104857600000000ULL; */ /* = 2^20 * 10^8 */ n = 110100480000000ULL; /* Value checked for AES32 and MADI */ div64_32(&n, rate, &r); @@ -864,13 +864,13 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally) if (!(hdspm->control_register & HDSPM_ClockModeMaster)) { - /* SLAVE --- */ + /* SLAVE --- */ if (called_internally) { - /* request from ctl or card initialization - just make a warning an remember setting + /* request from ctl or card initialization + just make a warning an remember setting for future master mode switching */ - + snd_printk(KERN_WARNING "HDSPM: " "Warning: device is not running " "as a clock master.\n"); @@ -907,7 +907,7 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally) Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do is to flag rate - changes in the read/write routines. + changes in the read/write routines. */ if (current_rate <= 48000) @@ -975,10 +975,10 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally) /* For AES32, need to set DDS value in FREQ register For MADI, also apparently */ hdspm_set_dds_value(hdspm, rate); - + if (hdspm->is_aes32 && rate != current_rate) hdspm_write(hdspm, HDSPM_eeprom_wr, 0); - + /* For AES32 and for MADI (at least rev 204), channel_map needs to * always be channel_map_madi_ss, whatever the sample rate */ hdspm->channel_map = channel_map_madi_ss; @@ -1015,7 +1015,7 @@ static void all_in_all_mixer(struct hdspm * hdspm, int sgain) MIDI ----------------------------------------------------------------------------*/ -static inline unsigned char snd_hdspm_midi_read_byte (struct hdspm *hdspm, +static inline unsigned char snd_hdspm_midi_read_byte(struct hdspm *hdspm, int id) { /* the hardware already does the relevant bit-mask with 0xff */ @@ -1025,7 +1025,7 @@ static inline unsigned char snd_hdspm_midi_read_byte (struct hdspm *hdspm, return hdspm_read(hdspm, HDSPM_midiDataIn0); } -static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id, +static inline void snd_hdspm_midi_write_byte(struct hdspm *hdspm, int id, int val) { /* the hardware already does the relevant bit-mask with 0xff */ @@ -1035,7 +1035,7 @@ static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id, hdspm_write(hdspm, HDSPM_midiDataOut0, val); } -static inline int snd_hdspm_midi_input_available (struct hdspm *hdspm, int id) +static inline int snd_hdspm_midi_input_available(struct hdspm *hdspm, int id) { if (id) return (hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xff); @@ -1043,7 +1043,7 @@ static inline int snd_hdspm_midi_input_available (struct hdspm *hdspm, int id) return (hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xff); } -static inline int snd_hdspm_midi_output_possible (struct hdspm *hdspm, int id) +static inline int snd_hdspm_midi_output_possible(struct hdspm *hdspm, int id) { int fifo_bytes_used; @@ -1065,7 +1065,7 @@ static void snd_hdspm_flush_midi_input(struct hdspm *hdspm, int id) snd_hdspm_midi_read_byte (hdspm, id); } -static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi) +static int snd_hdspm_midi_output_write(struct hdspm_midi *hmidi) { unsigned long flags; int n_pending; @@ -1074,7 +1074,7 @@ static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi) unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output && !snd_rawmidi_transmit_empty (hmidi->output)) { @@ -1083,11 +1083,11 @@ static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi) if (n_pending > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending); if (to_write > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdspm_midi_write_byte (hmidi->hdspm, hmidi->id, buf[i]); @@ -1098,7 +1098,7 @@ static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi) return 0; } -static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi) +static int snd_hdspm_midi_input_read(struct hdspm_midi *hmidi) { unsigned char buf[128]; /* this buffer is designed to match the MIDI * input FIFO size @@ -1167,14 +1167,14 @@ static void snd_hdspm_midi_output_timer(unsigned long data) { struct hdspm_midi *hmidi = (struct hdspm_midi *) data; unsigned long flags; - + snd_hdspm_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1278,7 +1278,7 @@ static struct snd_rawmidi_ops snd_hdspm_midi_input = .trigger = snd_hdspm_midi_input_trigger, }; -static int __devinit snd_hdspm_create_midi (struct snd_card *card, +static int __devinit snd_hdspm_create_midi(struct snd_card *card, struct hdspm *hdspm, int id) { int err; @@ -1312,12 +1312,12 @@ static int __devinit snd_hdspm_create_midi (struct snd_card *card, static void hdspm_midi_tasklet(unsigned long arg) { struct hdspm *hdspm = (struct hdspm *)arg; - + if (hdspm->midi[0].pending) snd_hdspm_midi_input_read (&hdspm->midi[0]); if (hdspm->midi[1].pending) snd_hdspm_midi_input_read (&hdspm->midi[1]); -} +} /*----------------------------------------------------------------------------- @@ -2475,8 +2475,8 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol, /* Simple Mixer deprecated since to much faders ??? MIXER interface says output (source, destination, value) - where source > MAX_channels are playback channels - on MADICARD + where source > MAX_channels are playback channels + on MADICARD - playback mixer matrix: [channelout+64] [output] [value] - input(thru) mixer matrix: [channelin] [output] [value] (better do 2 kontrols for seperation ?) @@ -2586,7 +2586,7 @@ static int snd_hdspm_put_mixer(struct snd_kcontrol *kcontrol, /* The simple mixer control(s) provide gain control for the basic 1:1 mappings of playback streams to output - streams. + streams. */ #define HDSPM_PLAYBACK_MIXER \ @@ -2634,7 +2634,7 @@ static int snd_hdspm_get_playback_mixer(struct snd_kcontrol *kcontrol, snd_printdd("get pb mixer index %d, channel %d, mapped_channel %d, " "value %d\n", ucontrol->id.index, channel, mapped_channel, - ucontrol->value.integer.value[0]); + ucontrol->value.integer.value[0]); */ return 0; } @@ -2906,7 +2906,7 @@ static int snd_hdspm_create_controls(struct snd_card *card, struct hdspm * hdspm } } - /* Channel playback mixer as default control + /* Channel playback mixer as default control Note: the whole matrix would be 128*HDSPM_MIXER_CHANNELS Faders, thats too * big for any alsamixer they are accesible via special IOCTL on hwdep and the mixer 2dimensional mixer control @@ -2934,7 +2934,7 @@ static int snd_hdspm_create_controls(struct snd_card *card, struct hdspm * hdspm } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3341,7 +3341,7 @@ static void __devinit snd_hdspm_proc_init(struct hdspm * hdspm) } /*------------------------------------------------------------ - hdspm intitialize + hdspm intitialize ------------------------------------------------------------*/ static int snd_hdspm_set_defaults(struct hdspm * hdspm) @@ -3417,7 +3417,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) /*------------------------------------------------------------ - interrupt + interrupt ------------------------------------------------------------*/ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) @@ -3481,7 +3481,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) } /*------------------------------------------------------------ - pcm interface + pcm interface ------------------------------------------------------------*/ @@ -3719,7 +3719,7 @@ static int snd_hdspm_hw_free(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* params_channels(params) should be enough, + /* params_channels(params) should be enough, but to get sure in case of error */ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i) snd_hdspm_enable_out(hdspm, i, 0); @@ -3827,7 +3827,7 @@ static int snd_hdspm_trigger(struct snd_pcm_substream *substream, int cmd) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) hdspm_silence_playback(hdspm); } - _ok: +_ok: snd_pcm_trigger_done(substream, substream); if (!hdspm->running && running) hdspm_start_audio(hdspm); @@ -4124,13 +4124,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, /* maybe there is a chance to memorymap in future * so dont touch just copy */ - if(copy_to_user_fromio((void __user *)rms.peak, + if (copy_to_user_fromio((void __user *)rms.peak, hdspm->iobase+HDSPM_MADI_peakrmsbase, - sizeof(struct hdspm_peak_rms)) != 0 ) + sizeof(struct hdspm_peak_rms)) != 0) return -EFAULT; break; - + case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO: @@ -4222,7 +4222,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, /*------------------------------------------------------------ - memory interface + memory interface ------------------------------------------------------------*/ static int __devinit snd_hdspm_preallocate_memory(struct hdspm * hdspm) { diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 2570907..7a4fdad 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1,5 +1,5 @@ /* - * ALSA driver for RME Digi9652 audio interfaces + * ALSA driver for RME Digi9652 audio interfaces * * Copyright (c) 1999 IEM - Winfried Ritsch * Copyright (c) 1999-2001 Paul Davis @@ -35,7 +35,7 @@ #include <sound/initval.h> #include <asm/current.h> -#include <asm/io.h> +#include <linux/io.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -60,7 +60,7 @@ MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall}," capture, one for playback. Both the ADAT and S/PDIF channels appear to the host CPU in the same block of memory. There is no functional difference between them in terms of access. - + The Hammerfall Light is identical to the Hammerfall, except that it has 2 sets 18 channels (16 ADAT + 2 S/PDIF) for capture and playback. */ @@ -115,7 +115,7 @@ MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall}," /* Bits 31,30,29 are bits 5,4,3 of h/w pointer position on later Rev G EEPROMS and Rev 1.5 cards or later. -*/ +*/ #define RME9652_REV15_buf_pos(x) ((((x)&0xE0000000)>>26)|((x)&RME9652_buf_pos)) @@ -186,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall}," #define RME9652_CHANNEL_BUFFER_BYTES (4*RME9652_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the 9636 still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -204,7 +204,7 @@ struct snd_rme9652 { int irq; unsigned long port; void __iomem *iobase; - + int precise_ptr; u32 control_register; /* cached value */ @@ -217,7 +217,7 @@ struct snd_rme9652 { size_t hw_offsetmask; /* &-with status register to get real hw_offset */ size_t prev_hw_offset; /* previous hw offset */ - size_t max_jitter; /* maximum jitter in frames for + size_t max_jitter; /* maximum jitter in frames for hw pointer */ size_t period_bytes; /* guess what this is */ @@ -266,7 +266,7 @@ static char channel_map_9652_ss[26] = { }; static char channel_map_9636_ss[26] = { - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, /* channels 16 and 17 are S/PDIF */ 24, 25, /* channels 18-25 don't exist */ @@ -367,7 +367,7 @@ static inline void rme9652_compute_period_size(struct snd_rme9652 *rme9652) i = rme9652->control_register & RME9652_latency; rme9652->period_bytes = 1 << ((rme9652_decode_latency(i) + 8)); - rme9652->hw_offsetmask = + rme9652->hw_offsetmask = (rme9652->period_bytes * 2 - 1) & RME9652_buf_pos; rme9652->max_jitter = 80; } @@ -426,7 +426,7 @@ static inline void rme9652_reset_hw_pointer(struct snd_rme9652 *rme9652) /* reset the FIFO pointer to zero. We do this by writing to 8 registers, each of which is a 32bit wide register, and set them all to zero. Note that s->iobase is a pointer to - int32, not pointer to char. + int32, not pointer to char. */ for (i = 0; i < 8; i++) { @@ -494,9 +494,9 @@ static int rme9652_set_rate(struct snd_rme9652 *rme9652, int rate) /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -605,7 +605,7 @@ static void rme9652_set_thru(struct snd_rme9652 *rme9652, int channel, int enabl rme9652_write(rme9652, RME9652_thru_base + mapped_channel * 4, - enable ? 1 : 0); + enable ? 1 : 0); } } @@ -619,7 +619,7 @@ static int rme9652_set_passthru(struct snd_rme9652 *rme9652, int onoff) */ rme9652->control_register = - RME9652_inp_0 | + RME9652_inp_0 | rme9652_encode_latency(7) | RME9652_start_bit; @@ -630,24 +630,24 @@ static int rme9652_set_passthru(struct snd_rme9652 *rme9652, int onoff) rme9652->passthru = 1; } else { rme9652_set_thru(rme9652, -1, 0); - rme9652_stop(rme9652); + rme9652_stop(rme9652); rme9652->passthru = 0; } return 0; } -static void rme9652_spdif_set_bit (struct snd_rme9652 *rme9652, int mask, int onoff) +static void rme9652_spdif_set_bit(struct snd_rme9652 *rme9652, int mask, int onoff) { - if (onoff) + if (onoff) rme9652->control_register |= mask; - else + else rme9652->control_register &= ~mask; - + rme9652_write(rme9652, RME9652_control_register, rme9652->control_register); } -static void rme9652_spdif_write_byte (struct snd_rme9652 *rme9652, const int val) +static void rme9652_spdif_write_byte(struct snd_rme9652 *rme9652, const int val) { long mask; long i; @@ -655,7 +655,7 @@ static void rme9652_spdif_write_byte (struct snd_rme9652 *rme9652, const int val for (i = 0, mask = 0x80; i < 8; i++, mask >>= 1) { if (val & mask) rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_WRITE, 1); - else + else rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_WRITE, 0); rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_CLOCK, 1); @@ -663,7 +663,7 @@ static void rme9652_spdif_write_byte (struct snd_rme9652 *rme9652, const int val } } -static int rme9652_spdif_read_byte (struct snd_rme9652 *rme9652) +static int rme9652_spdif_read_byte(struct snd_rme9652 *rme9652) { long mask; long val; @@ -681,7 +681,7 @@ static int rme9652_spdif_read_byte (struct snd_rme9652 *rme9652) return val; } -static void rme9652_write_spdif_codec (struct snd_rme9652 *rme9652, const int address, const int data) +static void rme9652_write_spdif_codec(struct snd_rme9652 *rme9652, const int address, const int data) { rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 1); rme9652_spdif_write_byte (rme9652, 0x20); @@ -691,7 +691,7 @@ static void rme9652_write_spdif_codec (struct snd_rme9652 *rme9652, const int ad } -static int rme9652_spdif_read_codec (struct snd_rme9652 *rme9652, const int address) +static int rme9652_spdif_read_codec(struct snd_rme9652 *rme9652, const int address) { int ret; @@ -708,7 +708,7 @@ static int rme9652_spdif_read_codec (struct snd_rme9652 *rme9652, const int addr return ret; } -static void rme9652_initialize_spdif_receiver (struct snd_rme9652 *rme9652) +static void rme9652_initialize_spdif_receiver(struct snd_rme9652 *rme9652) { /* XXX what unsets this ? */ @@ -726,19 +726,19 @@ static inline int rme9652_spdif_sample_rate(struct snd_rme9652 *s) if (rme9652_read(s, RME9652_status_register) & RME9652_ERF) { return -1; /* error condition */ } - + if (s->hw_rev == 15) { int x, y, ret; - + x = rme9652_spdif_read_codec (s, 30); - if (x != 0) + if (x != 0) y = 48000 * 64 / x; else y = 0; - if (y > 30400 && y < 33600) ret = 32000; + if (y > 30400 && y < 33600) ret = 32000; else if (y > 41900 && y < 46000) ret = 44100; else if (y > 46000 && y < 50400) ret = 48000; else if (y > 60800 && y < 67200) ret = 64000; @@ -819,7 +819,7 @@ static int snd_rme9652_control_spdif_info(struct snd_kcontrol *kcontrol, struct static int snd_rme9652_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + snd_rme9652_convert_to_aes(&ucontrol->value.iec958, rme9652->creg_spdif); return 0; } @@ -829,7 +829,7 @@ static int snd_rme9652_control_spdif_put(struct snd_kcontrol *kcontrol, struct s struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_rme9652_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&rme9652->lock); change = val != rme9652->creg_spdif; @@ -848,7 +848,7 @@ static int snd_rme9652_control_spdif_stream_info(struct snd_kcontrol *kcontrol, static int snd_rme9652_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + snd_rme9652_convert_to_aes(&ucontrol->value.iec958, rme9652->creg_spdif_stream); return 0; } @@ -858,7 +858,7 @@ static int snd_rme9652_control_spdif_stream_put(struct snd_kcontrol *kcontrol, s struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_rme9652_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&rme9652->lock); change = val != rme9652->creg_spdif_stream; @@ -891,7 +891,7 @@ static int snd_rme9652_control_spdif_mask_get(struct snd_kcontrol *kcontrol, str static unsigned int rme9652_adat1_in(struct snd_rme9652 *rme9652) { if (rme9652->control_register & RME9652_ADAT1_INTERNAL) - return 1; + return 1; return 0; } @@ -936,7 +936,7 @@ static int snd_rme9652_info_adat1_in(struct snd_kcontrol *kcontrol, struct snd_c static int snd_rme9652_get_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme9652->lock); ucontrol->value.enumerated.item[0] = rme9652_adat1_in(rme9652); spin_unlock_irq(&rme9652->lock); @@ -948,7 +948,7 @@ static int snd_rme9652_put_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ct struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_rme9652_use_is_exclusive(rme9652)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % 2; @@ -1007,7 +1007,7 @@ static int snd_rme9652_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_c static int snd_rme9652_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme9652->lock); ucontrol->value.enumerated.item[0] = rme9652_spdif_in(rme9652); spin_unlock_irq(&rme9652->lock); @@ -1019,7 +1019,7 @@ static int snd_rme9652_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ct struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_rme9652_use_is_exclusive(rme9652)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % 3; @@ -1069,7 +1069,7 @@ static int rme9652_set_spdif_output(struct snd_rme9652 *rme9652, int out) static int snd_rme9652_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme9652->lock); ucontrol->value.integer.value[0] = rme9652_spdif_out(rme9652); spin_unlock_irq(&rme9652->lock); @@ -1081,7 +1081,7 @@ static int snd_rme9652_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_c struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_rme9652_use_is_exclusive(rme9652)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1156,7 +1156,7 @@ static int snd_rme9652_info_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ static int snd_rme9652_get_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme9652->lock); ucontrol->value.enumerated.item[0] = rme9652_sync_mode(rme9652); spin_unlock_irq(&rme9652->lock); @@ -1168,7 +1168,7 @@ static int snd_rme9652_put_sync_mode(struct snd_kcontrol *kcontrol, struct snd_c struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + val = ucontrol->value.enumerated.item[0] % 3; spin_lock_irq(&rme9652->lock); change = (int)val != rme9652_sync_mode(rme9652); @@ -1248,7 +1248,7 @@ static int snd_rme9652_info_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ static int snd_rme9652_get_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme9652->lock); ucontrol->value.enumerated.item[0] = rme9652_sync_pref(rme9652); spin_unlock_irq(&rme9652->lock); @@ -1260,7 +1260,7 @@ static int snd_rme9652_put_sync_pref(struct snd_kcontrol *kcontrol, struct snd_c struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_rme9652_use_is_exclusive(rme9652)) return -EBUSY; max = rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3; @@ -1308,7 +1308,7 @@ static int snd_rme9652_put_thru(struct snd_kcontrol *kcontrol, struct snd_ctl_el if (ucontrol->value.integer.value[chn]) thru_bits |= 1 << chn; } - + spin_lock_irq(&rme9652->lock); change = thru_bits ^ rme9652->thru_bits; if (change) { @@ -1379,7 +1379,7 @@ static int snd_rme9652_info_spdif_rate(struct snd_kcontrol *kcontrol, struct snd static int snd_rme9652_get_spdif_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&rme9652->lock); ucontrol->value.integer.value[0] = rme9652_spdif_sample_rate(rme9652); spin_unlock_irq(&rme9652->lock); @@ -1409,11 +1409,11 @@ static int snd_rme9652_get_adat_sync(struct snd_kcontrol *kcontrol, struct snd_c { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); unsigned int mask1, mask2, val; - + switch (kcontrol->private_value) { - case 0: mask1 = RME9652_lock_0; mask2 = RME9652_sync_0; break; - case 1: mask1 = RME9652_lock_1; mask2 = RME9652_sync_1; break; - case 2: mask1 = RME9652_lock_2; mask2 = RME9652_sync_2; break; + case 0: mask1 = RME9652_lock_0; mask2 = RME9652_sync_0; break; + case 1: mask1 = RME9652_lock_1; mask2 = RME9652_sync_1; break; + case 2: mask1 = RME9652_lock_2; mask2 = RME9652_sync_2; break; default: return -EINVAL; } val = rme9652_read(rme9652, RME9652_status_register); @@ -1433,8 +1433,8 @@ static int snd_rme9652_get_adat_sync(struct snd_kcontrol *kcontrol, struct snd_c static int snd_rme9652_get_tc_valid(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = + + ucontrol->value.integer.value[0] = (rme9652_read(rme9652, RME9652_status_register) & RME9652_tc_valid) ? 1 : 0; return 0; } @@ -1520,7 +1520,7 @@ static struct snd_kcontrol_new snd_rme9652_controls[] = { .get = snd_rme9652_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -1582,7 +1582,7 @@ static int snd_rme9652_create_controls(struct snd_card *card, struct snd_rme9652 } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -1606,10 +1606,10 @@ snd_rme9652_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buff snd_iprintf(buffer, "\n"); - x = 1 << (6 + rme9652_decode_latency(rme9652->control_register & + x = 1 << (6 + rme9652_decode_latency(rme9652->control_register & RME9652_latency)); - snd_iprintf(buffer, "Latency: %d samples (2 periods of %lu bytes)\n", + snd_iprintf(buffer, "Latency: %d samples (2 periods of %lu bytes)\n", x, (unsigned long) rme9652->period_bytes); snd_iprintf(buffer, "Hardware pointer (frames): %ld\n", rme9652_hw_pointer(rme9652)); @@ -1655,7 +1655,7 @@ snd_rme9652_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buff snd_iprintf(buffer, "\n"); - switch (rme9652_decode_spdif_in(rme9652->control_register & + switch (rme9652_decode_spdif_in(rme9652->control_register & RME9652_inp)) { case RME9652_SPDIFIN_OPTICAL: snd_iprintf(buffer, "IEC958 input: ADAT1\n"); @@ -1762,7 +1762,7 @@ static void __devinit snd_rme9652_proc_init(struct snd_rme9652 *rme9652) { struct snd_info_entry *entry; - if (! snd_card_proc_new(rme9652->card, "rme9652", &entry)) + if (!snd_card_proc_new(rme9652->card, "rme9652", &entry)) snd_info_set_text_ops(entry, rme9652, snd_rme9652_proc_read); } @@ -1828,11 +1828,11 @@ static void snd_rme9652_set_defaults(struct snd_rme9652 *rme9652) /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax autosync clock mode maximum latency (7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - + if rev 1.5, initialize the S/PDIF receiver. */ @@ -1893,11 +1893,11 @@ static char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652, if (snd_BUG_ON(channel < 0 || channel >= RME9652_NCHANNELS)) return NULL; - + if ((mapped_channel = rme9652->channel_map[channel]) < 0) { return NULL; } - + if (stream == SNDRV_PCM_STREAM_CAPTURE) { return rme9652->capture_buffer + (mapped_channel * RME9652_CHANNEL_BUFFER_BYTES); @@ -2142,10 +2142,10 @@ static int snd_rme9652_trigger(struct snd_pcm_substream *substream, rme9652_silence_playback(rme9652); } } else { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) rme9652_silence_playback(rme9652); } - _ok: +_ok: snd_pcm_trigger_done(substream, substream); if (!rme9652->running && running) rme9652_start(rme9652); @@ -2178,9 +2178,9 @@ static struct snd_pcm_hardware snd_rme9652_playback_subinfo = SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_DOUBLE), .formats = SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | + .rates = (SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 96000, @@ -2201,9 +2201,9 @@ static struct snd_pcm_hardware snd_rme9652_capture_subinfo = SNDRV_PCM_INFO_NONINTERLEAVED | SNDRV_PCM_INFO_SYNC_START), .formats = SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | + .rates = (SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 96000, @@ -2478,7 +2478,7 @@ static int __devinit snd_rme9652_create(struct snd_card *card, snd_printk(KERN_ERR "unable to remap region 0x%lx-0x%lx\n", rme9652->port, rme9652->port + RME9652_IO_EXTENT - 1); return -EBUSY; } - + if (request_irq(pci->irq, snd_rme9652_interrupt, IRQF_SHARED, "rme9652", rme9652)) { snd_printk(KERN_ERR "unable to request IRQ %d\n", pci->irq); @@ -2503,7 +2503,7 @@ static int __devinit snd_rme9652_create(struct snd_card *card, "Light", which does not have the expansion board. This method comes from information received from Mathhias Clausen at RME. Display the EEPROM and h/w revID where - relevant. + relevant. */ switch (rev) { @@ -2616,7 +2616,7 @@ static int __devinit snd_rme9652_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, rme9652->port, rme9652->irq); - + if ((err = snd_card_register(card)) < 0) { snd_card_free(card); return err;

1 0

[alsa-devel] [PATCH 17/24] sound/pci: coding style fixes: pcxhr
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/pcxhr/pcxhr.c | 44 ++++++++++++++++++++-------------------- sound/pci/pcxhr/pcxhr_core.c | 22 ++++++++++---------- sound/pci/pcxhr/pcxhr_hwdep.c | 16 +++++++------- sound/pci/pcxhr/pcxhr_mixer.c | 16 +++++++------- 4 files changed, 49 insertions(+), 49 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 0e06c6c..bebf866 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -197,7 +197,7 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, if (err < 0) { snd_printk(KERN_ERR "error CMD_ACCESS_IO_WRITE for PLL register : %x!\n", - err ); + err); return err; } } @@ -603,12 +603,12 @@ static void pcxhr_trigger_tasklet(unsigned long arg) /* put the streams into the running state now (increment pointer by interrupt) */ spin_lock_irqsave(&mgr->lock, flags); - for ( i =0; i < mgr->num_cards; i++) { + for (i =0; i < mgr->num_cards; i++) { struct pcxhr_stream *stream; chip = mgr->chip[i]; - for(j = 0; j < chip->nb_streams_capt; j++) { + for (j = 0; j < chip->nb_streams_capt; j++) { stream = &chip->capture_stream[j]; - if(stream->status == PCXHR_STREAM_STATUS_STARTED) + if (stream->status == PCXHR_STREAM_STATUS_STARTED) stream->status = PCXHR_STREAM_STATUS_RUNNING; } for (j = 0; j < chip->nb_streams_play; j++) { @@ -723,7 +723,7 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) subs->runtime->buffer_size); /* - if(subs->runtime->period_size <= PCXHR_GRANULARITY) { + if (subs->runtime->period_size <= PCXHR_GRANULARITY) { snd_printk(KERN_ERR "pcxhr_prepare : error period_size too small (%x)\n", (unsigned int)subs->runtime->period_size); return -EINVAL; @@ -735,11 +735,11 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) do { /* if the stream was stopped before, format and buffer were reset */ /* - if(stream->status == PCXHR_STREAM_STATUS_STOPPED) { + if (stream->status == PCXHR_STREAM_STATUS_STOPPED) { err = pcxhr_set_format(stream); - if(err) break; + if (err) break; err = pcxhr_update_r_buffer(stream); - if(err) break; + if (err) break; } */ @@ -790,7 +790,7 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs, /* set the format to the board */ /* err = pcxhr_set_format(stream); - if(err) { + if (err) { mutex_unlock(&mgr->setup_mutex); return err; } @@ -853,7 +853,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) /* copy the struct snd_pcm_hardware struct */ runtime->hw = pcxhr_caps; - if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) { + if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) { snd_printdd("pcxhr_open playback chip%d subs%d\n", chip->chip_idx, subs->number); stream = &chip->playback_stream[subs->number]; @@ -1025,7 +1025,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, }; mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { + if (!chip) { snd_printk(KERN_ERR "cannot allocate chip\n"); return -ENOMEM; } @@ -1078,7 +1078,7 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer /* calc cpu load of the dsp */ pcxhr_init_rmh(&rmh, CMD_GET_DSP_RESOURCES); - if( ! pcxhr_send_msg(mgr, &rmh) ) { + if (!pcxhr_send_msg(mgr, &rmh)) { int cur = rmh.stat[0]; int ref = rmh.stat[1]; if (ref > 0) { @@ -1108,7 +1108,7 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer rmh.stat_len = PCXHR_SIZE_MAX_STATUS; rmh.dsp_stat = 0; rmh.cmd_idx = CMD_LAST_INDEX; - if( ! pcxhr_send_msg(mgr, &rmh) ) { + if (!pcxhr_send_msg(mgr, &rmh)) { int i; for (i = 0; i < rmh.stat_len; i++) snd_iprintf(buffer, "debug[%02d] = %06x\n", i, rmh.stat[i]); @@ -1147,9 +1147,9 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "info", &entry)) + if (!snd_card_proc_new(chip->card, "info", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_info); - if (! snd_card_proc_new(chip->card, "sync", &entry)) + if (!snd_card_proc_new(chip->card, "sync", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_sync); } /* end of proc interface */ @@ -1167,7 +1167,7 @@ static int pcxhr_free(struct pcxhr_mgr *mgr) } /* reset board if some firmware was loaded */ - if(mgr->dsp_loaded) { + if (mgr->dsp_loaded) { pcxhr_reset_board(mgr); snd_printdd("reset pcxhr !\n"); } @@ -1205,7 +1205,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id if (dev >= SNDRV_CARDS) return -ENODEV; - if (! enable[dev]) { + if (!enable[dev]) { dev++; return -ENOENT; } @@ -1224,7 +1224,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id /* alloc card manager */ mgr = kzalloc(sizeof(*mgr), GFP_KERNEL); - if (! mgr) { + if (!mgr) { pci_disable_device(pci); return -ENOMEM; } @@ -1271,15 +1271,15 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id /* init taslket */ tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, (unsigned long) mgr); tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, (unsigned long) mgr); - mgr->prmh = kmalloc(sizeof(*mgr->prmh) + + mgr->prmh = kmalloc(sizeof(*mgr->prmh) + sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS), GFP_KERNEL); - if (! mgr->prmh) { + if (!mgr->prmh) { pcxhr_free(mgr); return -ENOMEM; } - for (i=0; i < PCXHR_MAX_CARDS; i++) { + for (i = 0; i < PCXHR_MAX_CARDS; i++) { struct snd_card *card; char tmpid[16]; int idx; @@ -1296,7 +1296,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : card_name, i); card = snd_card_new(idx, tmpid, THIS_MODULE, 0); - if (! card) { + if (!card) { snd_printk(KERN_ERR "cannot allocate the card %d\n", i); pcxhr_free(mgr); return -ENOMEM; diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 7143259..16575f4 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -23,7 +23,7 @@ #include <linux/delay.h> #include <linux/firmware.h> #include <linux/interrupt.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include "pcxhr.h" #include "pcxhr_mixer.h" @@ -246,9 +246,9 @@ void pcxhr_reset_dsp(struct pcxhr_mgr *mgr) /* let's reset the DSP */ PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, 0); - msleep( PCXHR_WAIT_DEFAULT ); /* wait 2 msec */ + msleep(PCXHR_WAIT_DEFAULT); /* wait 2 msec */ PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, 3); - msleep( PCXHR_WAIT_DEFAULT ); /* wait 2 msec */ + msleep(PCXHR_WAIT_DEFAULT); /* wait 2 msec */ /* reset mailbox */ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0, 0); @@ -275,7 +275,7 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin chipsc = PCXHR_INPL(mgr, PCXHR_PLX_CHIPSC); /* REV01 cards do not support the PCXHR_CHIPSC_GPI_USERI bit anymore */ /* this bit will always be 1; no possibility to test presence of first xilinx */ - if(second) { + if (second) { if ((chipsc & PCXHR_CHIPSC_GPI_USERI) == 0) { snd_printk(KERN_ERR "error loading first xilinx\n"); return -EINVAL; @@ -283,7 +283,7 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin /* activate second xilinx */ chipsc |= PCXHR_CHIPSC_RESET_XILINX; PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc); - msleep( PCXHR_WAIT_DEFAULT ); /* wait 2 msec */ + msleep(PCXHR_WAIT_DEFAULT); /* wait 2 msec */ } image = xilinx->data; for (i = 0; i < xilinx->size; i++, image++) { @@ -304,7 +304,7 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | PCXHR_CHIPSC_DATA_IN); PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc); /* wait 2 msec (time to boot the xilinx before any access) */ - msleep( PCXHR_WAIT_DEFAULT ); + msleep(PCXHR_WAIT_DEFAULT); return 0; } @@ -627,7 +627,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) PCXHR_OUTPB(mgr, PCXHR_DSP_TXM, (data>>8)&0xFF); PCXHR_OUTPB(mgr, PCXHR_DSP_TXL, (data&0xFF)); - for (i=1; i < rmh->cmd_len; i++) { + for (i = 1; i < rmh->cmd_len; i++) { /* send other words */ data = rmh->cmd[i]; #ifdef CONFIG_SND_DEBUG_VERBOSE @@ -706,7 +706,7 @@ void pcxhr_set_pipe_cmd_params(struct pcxhr_rmh *rmh, int capture, snd_BUG_ON(param2 > MASK_FIRST_FIELD); rmh->cmd[0] |= param2; } - if(param3) { + if (param3) { snd_BUG_ON(param3 > MASK_DSP_WORD); rmh->cmd[1] = param3; rmh->cmd_len = 2; @@ -841,7 +841,7 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) pcxhr_init_rmh(&rmh, CMD_SEND_IRQA); err = pcxhr_send_msg(mgr, &rmh); if (err) { - snd_printk(KERN_ERR "error pipe start (CMD_SEND_IRQA) err=%x!\n", err ); + snd_printk(KERN_ERR "error pipe start (CMD_SEND_IRQA) err=%x!\n", err); return err; } return 0; @@ -1154,7 +1154,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) spin_lock(&mgr->lock); reg = PCXHR_INPL(mgr, PCXHR_PLX_IRQCS); - if (! (reg & PCXHR_IRQCS_ACTIVE_PCIDB)) { + if (!(reg & PCXHR_IRQCS_ACTIVE_PCIDB)) { spin_unlock(&mgr->lock); return IRQ_NONE; /* this device did not cause the interrupt */ } @@ -1211,7 +1211,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) */ /* better resynchronize all streams next interrupt : */ mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; - + mgr->src_it_dsp = reg; tasklet_hi_schedule(&mgr->msg_taskq); } diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 96640d9..9d04175 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -24,7 +24,7 @@ #include <linux/vmalloc.h> #include <linux/firmware.h> #include <linux/pci.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/hwdep.h> #include "pcxhr.h" @@ -104,7 +104,7 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) mgr->board_has_analog = 1; /* analog addon board available */ else /* analog addon board not available -> no support for instance */ - return -EINVAL; + return -EINVAL; /* unmute inputs */ err = pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, @@ -144,7 +144,7 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr) /* * allocate a playback/capture pipe (pcmp0/pcmc0) */ -static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe, +static int pcxhr_dsp_allocate_pipe(struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe, int is_capture, int pin) { int stream_count, audio_count; @@ -166,10 +166,10 @@ static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pi pipe->first_audio = pin; /* define pipe (P_PCM_ONLY_MASK (0x020000) is not necessary) */ pcxhr_init_rmh(&rmh, CMD_RES_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, audio_count, stream_count); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, audio_count, stream_count); err = pcxhr_send_msg(mgr, &rmh); if (err < 0) { - snd_printk(KERN_ERR "error pipe allocation (CMD_RES_PIPE) err=%x!\n", err ); + snd_printk(KERN_ERR "error pipe allocation (CMD_RES_PIPE) err=%x!\n", err); return err; } pipe->status = PCXHR_PIPE_DEFINED; @@ -181,7 +181,7 @@ static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pi * free playback/capture pipe (pcmp0/pcmc0) */ #if 0 -static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe) +static int pcxhr_dsp_free_pipe(struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe) { struct pcxhr_rmh rmh; int capture_mask = 0; @@ -223,7 +223,7 @@ static int pcxhr_config_pipes(struct pcxhr_mgr *mgr) err = pcxhr_dsp_allocate_pipe( mgr, pipe, 0, i*2); if (err) return err; - for(j = 0; j < chip->nb_streams_play; j++) + for (j = 0; j < chip->nb_streams_play; j++) chip->playback_stream[j].pipe = pipe; } for (j = 0; j < chip->nb_streams_capt; j++) { @@ -392,7 +392,7 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, fw.size = dsp->length; fw.data = vmalloc(fw.size); - if (! fw.data) { + if (!fw.data) { snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%lu bytes)\n", (unsigned long)fw.size); return -ENOMEM; diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index aabc7bc..fd7e60c 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -173,7 +173,7 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int i, changed = 0; mutex_lock(&chip->mgr->mixer_mutex); - for(i = 0; i < 2; i++) { + for (i = 0; i < 2; i++) { if (chip->analog_playback_active[i] != ucontrol->value.integer.value[i]) { chip->analog_playback_active[i] = @@ -284,7 +284,7 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); - if(err<0) { + if (err<0) { snd_printk(KERN_DEBUG "error update_audio_level card(%d) err(%x)\n", chip->chip_idx, err); return -EINVAL; @@ -445,7 +445,7 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[i]) { chip->monitoring_volume[i] = !!ucontrol->value.integer.value[i]; - if(chip->monitoring_active[i]) + if (chip->monitoring_active[i]) /* update monitoring volume and mute */ /* do only when monitoring is unmuted */ pcxhr_update_audio_pipe_level(chip, 0, i); @@ -566,7 +566,7 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) rmh.cmd[1] = codec; rmh.cmd[2] = (CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x54); err = pcxhr_send_msg(chip->mgr, &rmh); - if(err) + if (err) return err; rmh.cmd[2] = (CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x49); err = pcxhr_send_msg(chip->mgr, &rmh); @@ -722,7 +722,7 @@ static int pcxhr_clock_rate_get(struct snd_kcontrol *kcontrol, int i, err, rate; mutex_lock(&mgr->mixer_mutex); - for(i = 0; i < 3 + mgr->capture_chips; i++) { + for (i = 0; i < 3 + mgr->capture_chips; i++) { if (i == PCXHR_CLOCK_TYPE_INTERNAL) rate = mgr->sample_rate_real; else { @@ -802,7 +802,7 @@ static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v int i, err; mutex_lock(&chip->mgr->mixer_mutex); - for(i = 0; i < 5; i++) { + for (i = 0; i < 5; i++) { if (kcontrol->private_value == 0) /* playback */ aes_bits = chip->aes_bits[i]; else { /* capture */ @@ -835,7 +835,7 @@ static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigne for (i = 0; i < 8; i++) { if ((old_bits & 0x01) != (new_bits & 0x01)) { cmd = chip->chip_idx & 0x03; /* chip index 0..3 */ - if(chip->chip_idx > 3) + if (chip->chip_idx > 3) /* new bit used if chip_idx>3 (PCX1222HR) */ cmd |= 1 << 22; cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */ @@ -971,7 +971,7 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr) snd_ctl_new1(&pcxhr_control_output_switch, chip))) < 0) return err; - + temp = snd_pcxhr_pcm_vol; temp.name = "PCM Playback Volume"; temp.count = PCXHR_PLAYBACK_STREAMS;

1 0

[alsa-devel] [PATCH 16/24] sound/pci: coding style fixes: nm256
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/nm256/nm256.c | 66 +++++++++++++++++++++--------------------- sound/pci/nm256/nm256_coef.c | 4 +- sound/pci/oxygen/oxygen_io.c | 2 +- 3 files changed, 36 insertions(+), 36 deletions(-) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 50c9f8a..b12e322 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1,4 +1,4 @@ -/* +/* * Driver for NeoMagic 256AV and 256ZX chipsets. * Copyright (c) 2000 by Takashi Iwai <tiwai(a)suse.de> * @@ -23,8 +23,8 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - -#include <asm/io.h> + +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -190,7 +190,7 @@ struct nm256_stream { struct snd_pcm_substream *substream; int running; int suspended; - + u32 buf; /* offset from chip->buffer */ int bufsize; /* buffer size in bytes */ void __iomem *bufptr; /* mapped pointer */ @@ -205,7 +205,7 @@ struct nm256_stream { }; struct nm256 { - + struct snd_card *card; void __iomem *cport; /* control port */ @@ -375,11 +375,11 @@ snd_nm256_load_coefficient(struct nm256 *chip, int stream, int number) if (stream == SNDRV_PCM_STREAM_CAPTURE) number += 8; - if (! chip->use_cache) { + if (!chip->use_cache) { snd_nm256_load_one_coefficient(chip, stream, addr, number); return; } - if (! chip->coeffs_current) { + if (!chip->coeffs_current) { snd_nm256_write_buffer(chip, coefficients, chip->all_coeff_buf, NM_TOTAL_COEFF_COUNT * 4); chip->coeffs_current = 1; @@ -400,7 +400,7 @@ static unsigned int samplerates[8] = { 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, }; static struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(samplerates), + .count = ARRAY_SIZE(samplerates), .list = samplerates, .mask = 0, }; @@ -571,7 +571,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) s->suspended = 0; /* fallthru */ case SNDRV_PCM_TRIGGER_START: - if (! s->running) { + if (!s->running) { snd_nm256_playback_start(chip, s, substream); s->running = 1; } @@ -607,7 +607,7 @@ snd_nm256_capture_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - if (! s->running) { + if (!s->running) { snd_nm256_capture_start(chip, s, substream); s->running = 1; } @@ -680,7 +680,7 @@ snd_nm256_capture_pointer(struct snd_pcm_substream *substream) if (snd_BUG_ON(!s)) return 0; curp = snd_nm256_readl(chip, NM_RBUFFER_CURRP) - (unsigned long)s->buf; - curp %= s->dma_size; + curp %= s->dma_size; return bytes_to_frames(substream->runtime, curp); } @@ -956,8 +956,8 @@ snd_nm256_pcm(struct nm256 *chip, int device) } -/* - * Initialize the hardware. +/* + * Initialize the hardware. */ static void snd_nm256_init_chip(struct nm256 *chip) @@ -966,8 +966,8 @@ snd_nm256_init_chip(struct nm256 *chip) snd_nm256_writeb(chip, 0x0, 0x11); snd_nm256_writew(chip, 0x214, 0); /* stop sounds.. */ - //snd_nm256_playback_stop(chip); - //snd_nm256_capture_stop(chip); + /* snd_nm256_playback_stop(chip); */ + /* snd_nm256_capture_stop(chip); */ } @@ -997,8 +997,8 @@ snd_nm256_intr_check(struct nm256 *chip) return IRQ_NONE; } -/* - * Handle a potential interrupt for the device referred to by DEV_ID. +/* + * Handle a potential interrupt for the device referred to by DEV_ID. * * I don't like the cut-n-paste job here either between the two routines, * but there are sufficient differences between the two interrupt handlers @@ -1147,8 +1147,8 @@ snd_nm256_ac97_ready(struct nm256 *chip) testaddr = chip->mixer_status_offset; testb = chip->mixer_status_mask; - /* - * Loop around waiting for the mixer to become ready. + /* + * Loop around waiting for the mixer to become ready. */ while (timeout-- > 0) { if ((snd_nm256_readw(chip, testaddr) & testb) == 0) @@ -1158,7 +1158,7 @@ snd_nm256_ac97_ready(struct nm256 *chip) return 0; } -/* +/* * Initial register values to be written to the AC97 mixer. * While most of these are identical to the reset values, we do this * so that we have most of the register contents cached--this avoids @@ -1187,7 +1187,7 @@ static struct initialValues nm256_ac97_init_val[] = { AC97_REC_SEL, 0x0000 }, { AC97_REC_GAIN, 0x0B0B }, { AC97_GENERAL_PURPOSE, 0x0000 }, - { AC97_3D_CONTROL, 0x8000 }, + { AC97_3D_CONTROL, 0x8000 }, { AC97_VENDOR_ID1, 0x8384 }, { AC97_VENDOR_ID2, 0x7609 }, }; @@ -1217,7 +1217,7 @@ snd_nm256_ac97_read(struct snd_ac97 *ac97, unsigned short reg) return chip->ac97_regs[idx]; } -/* +/* */ static void snd_nm256_ac97_write(struct snd_ac97 *ac97, @@ -1273,16 +1273,16 @@ snd_nm256_ac97_reset(struct snd_ac97 *ac97) /* Reset the mixer. 'Tis magic! */ snd_nm256_writeb(chip, 0x6c0, 1); - if (! chip->reset_workaround) { + if (!chip->reset_workaround) { /* Dell latitude LS will lock up by this */ snd_nm256_writeb(chip, 0x6cc, 0x87); } - if (! chip->reset_workaround_2) { + if (!chip->reset_workaround_2) { /* Dell latitude CSx will lock up by this */ snd_nm256_writeb(chip, 0x6cc, 0x80); snd_nm256_writeb(chip, 0x6cc, 0x0); } - if (! chip->in_resume) { + if (!chip->in_resume) { int i; for (i = 0; i < ARRAY_SIZE(nm256_ac97_init_val); i++) { /* preload the cache, so as to avoid even a single @@ -1309,7 +1309,7 @@ snd_nm256_mixer(struct nm256 *chip) chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val), sizeof(short), GFP_KERNEL); - if (! chip->ac97_regs) + if (!chip->ac97_regs) return -ENOMEM; if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) @@ -1323,14 +1323,14 @@ snd_nm256_mixer(struct nm256 *chip) err = snd_ac97_mixer(pbus, &ac97, &chip->ac97); if (err < 0) return err; - if (! (chip->ac97->id & (0xf0000000))) { + if (!(chip->ac97->id & (0xf0000000))) { /* looks like an invalid id */ sprintf(chip->card->mixername, "%s AC97", chip->card->driver); } return 0; } -/* +/* * See if the signature left by the NM256 BIOS is intact; if so, we use * the associated address as the end of our audio buffer in the video * RAM. @@ -1498,7 +1498,7 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize = playback_bufsize * 1024; chip->streams[SNDRV_PCM_STREAM_CAPTURE].bufsize = capture_bufsize * 1024; - /* + /* * The NM256 has two memory ports. The first port is nothing * more than a chunk of video RAM, which is used as the I/O ring * buffer. The second port has the actual juicy stuff (like the @@ -1529,7 +1529,7 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, /* Ok, try to see if this is a non-AC97 version of the hardware. */ pval = snd_nm256_readw(chip, NM_MIXER_PRESENCE); if ((pval & NM_PRESENCE_MASK) != NM_PRESENCE_VALUE) { - if (! force_ac97) { + if (!force_ac97) { printk(KERN_ERR "nm256: no ac97 is found!\n"); printk(KERN_ERR " force the driver to load by " "passing in the module parameter\n"); @@ -1555,7 +1555,7 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, chip->mixer_status_offset = NM2_MIXER_STATUS_OFFSET; chip->mixer_status_mask = NM2_MIXER_READY_MASK; } - + chip->buffer_size = chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize + chip->streams[SNDRV_PCM_STREAM_CAPTURE].bufsize; if (chip->use_cache) @@ -1614,8 +1614,8 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, snd_nm256_init_chip(chip); - // pci_set_master(pci); /* needed? */ - + /* pci_set_master(pci); needed? */ + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) goto __error; diff --git a/sound/pci/nm256/nm256_coef.c b/sound/pci/nm256/nm256_coef.c index 747d5d6..a75e037 100644 --- a/sound/pci/nm256/nm256_coef.c +++ b/sound/pci/nm256/nm256_coef.c @@ -1,6 +1,6 @@ #define NM_TOTAL_COEFF_COUNT 0x3158 -static char coefficients[NM_TOTAL_COEFF_COUNT * 4] = { +static char coefficients[NM_TOTAL_COEFF_COUNT * 4] = { 0xFF, 0xFF, 0x2F, 0x00, 0x4B, 0xFF, 0xA5, 0x01, 0xEF, 0xFC, 0x21, 0x05, 0x87, 0xF7, 0x62, 0x11, 0xE9, 0x45, 0x5E, 0xF9, 0xB5, 0x01, 0xDE, 0xFF, 0xA4, 0xFF, 0x60, 0x00, 0xCA, 0xFF, 0x0D, 0x00, 0xFD, @@ -4594,7 +4594,7 @@ static char coefficients[NM_TOTAL_COEFF_COUNT * 4] = { 0x9A, 0xFF, 0x0B, 0x00, 0x00, 0x00, 0x00, 0x00, 0x05, 0x00, 0xAD, 0xFF, 0x48, 0x01, 0xE2, 0xFC, 0xDA, 0x05, 0x97, 0xF6, 0x0B, 0x0F, 0x79, 0x42, 0xC7, 0x03, 0x9E, 0xFA, 0x71, 0x04, 0x28, 0xFD, 0x63, - 0x01, 0x8D, 0xFF, 0x0F, 0x00 + 0x01, 0x8D, 0xFF, 0x0F, 0x00 }; static u16 diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 83f135f..17618c1 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -20,7 +20,7 @@ #include <linux/delay.h> #include <linux/sched.h> #include <sound/core.h> -#include <asm/io.h> +#include <linux/io.h> #include "oxygen.h" u8 oxygen_read8(struct oxygen *chip, unsigned int reg)

1 0

[alsa-devel] [PATCH 15/24] sound/pci: coding style fixes: mixart
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/mixart/mixart.c | 148 +++++++++++++++++++------------------- sound/pci/mixart/mixart_core.c | 100 +++++++++++++------------- sound/pci/mixart/mixart_hwdep.c | 128 +++++++++++++++++----------------- sound/pci/mixart/mixart_hwdep.h | 28 ++++---- sound/pci/mixart/mixart_mixer.c | 56 +++++++------- 5 files changed, 230 insertions(+), 230 deletions(-) diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce6..9db4225 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -80,10 +80,10 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, switch(pipe->status) { case PIPE_RUNNING: case PIPE_CLOCK_SET: - if(start) return 0; /* already started */ + if (start) return 0; /* already started */ break; case PIPE_STOPPED: - if(!start) return 0; /* already stopped */ + if (!start) return 0; /* already stopped */ break; default: snd_printk(KERN_ERR "error mixart_set_pipe_state called with wrong pipe->status!\n"); @@ -100,7 +100,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, request.size = sizeof(system_msg_uid); err = snd_mixart_send_msg_wait_notif(mgr, &request, system_msg_uid); - if(err) { + if (err) { snd_printk(KERN_ERR "error : MSG_SYSTEM_WAIT_SYNCHRO_CMD was not notified !\n"); return err; } @@ -111,7 +111,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, group_state.pipe_count = 1; group_state.pipe_uid[0] = pipe->group_uid; - if(start) + if (start) request.message_id = MSG_STREAM_START_STREAM_GRP_PACKET; else request.message_id = MSG_STREAM_STOP_STREAM_GRP_PACKET; @@ -126,7 +126,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, return -EINVAL; } - if(start) { + if (start) { u32 stat; group_state.pipe_count = 0; /* in case of start same command once again with pipe_count=0 */ @@ -171,10 +171,10 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_CLOCK_SET: break; case PIPE_RUNNING: - if(rate != 0) + if (rate != 0) break; default: - if(rate == 0) + if (rate == 0) return 0; /* nothing to do */ else { snd_printk(KERN_ERR "error mixart_set_clock(%d) called with wrong pipe->status !\n", rate); @@ -202,7 +202,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, return -EINVAL; } - if(rate) pipe->status = PIPE_CLOCK_SET; + if (rate) pipe->status = PIPE_CLOCK_SET; else pipe->status = PIPE_RUNNING; return 0; @@ -220,7 +220,7 @@ snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, struct mixart_pipe *pipe; struct mixart_msg request; - if(capture) { + if (capture) { if (pcm_number == MIXART_PCM_ANALOG) { pipe = &(chip->pipe_in_ana); /* analog inputs */ } else { @@ -239,12 +239,12 @@ snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, } /* a new stream is opened and there are already all streams in use */ - if( (monitoring == 0) && (pipe->references >= stream_count) ) { + if ((monitoring == 0) && (pipe->references >= stream_count)) { return NULL; } /* pipe is not yet defined */ - if( pipe->status == PIPE_UNDEFINED ) { + if (pipe->status == PIPE_UNDEFINED) { int err, i; struct { struct mixart_streaming_group_req sgroup_req; @@ -268,11 +268,11 @@ snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, buf->sgroup_req.latency = 256; buf->sgroup_req.connector = pipe->uid_left_connector; /* the left connector */ - for (i=0; i<stream_count; i++) { + for (i = 0; i < stream_count; i++) { int j; struct mixart_flowinfo *flowinfo; struct mixart_bufferinfo *bufferinfo; - + /* we don't yet know the format, so config 16 bit pcm audio for instance */ buf->sgroup_req.stream_info[i].size_max_byte_frame = 1024; buf->sgroup_req.stream_info[i].size_max_sample_frame = 256; @@ -280,7 +280,7 @@ snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, /* find the right bufferinfo_array */ j = (chip->chip_idx * MIXART_MAX_STREAM_PER_CARD) + (pcm_number * (MIXART_PLAYBACK_STREAMS + MIXART_CAPTURE_STREAMS)) + i; - if(capture) j += MIXART_PLAYBACK_STREAMS; /* in the array capture is behind playback */ + if (capture) j += MIXART_PLAYBACK_STREAMS; /* in the array capture is behind playback */ buf->sgroup_req.flow_entry[i] = j; @@ -294,13 +294,13 @@ snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, /* construct the identifier of the stream buffer received in the interrupts ! */ bufferinfo[j].buffer_id = (chip->chip_idx << MIXART_NOTIFY_CARD_OFFSET) + (pcm_number << MIXART_NOTIFY_PCM_OFFSET ) + i; - if(capture) { + if (capture) { bufferinfo[j].buffer_id |= MIXART_NOTIFY_CAPT_MASK; } } err = snd_mixart_send_msg(chip->mgr, &request, sizeof(buf->sgroup_resp), &buf->sgroup_resp); - if((err < 0) || (buf->sgroup_resp.status != 0)) { + if ((err < 0) || (buf->sgroup_resp.status != 0)) { snd_printk(KERN_ERR "error MSG_STREAM_ADD_**PUT_GROUP err=%x stat=%x !\n", err, buf->sgroup_resp.status); kfree(buf); return NULL; @@ -314,7 +314,7 @@ snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, kfree(buf); } - if(monitoring) pipe->monitoring = 1; + if (monitoring) pipe->monitoring = 1; else pipe->references++; return pipe; @@ -326,28 +326,28 @@ int snd_mixart_kill_ref_pipe(struct mixart_mgr *mgr, { int err = 0; - if(pipe->status == PIPE_UNDEFINED) + if (pipe->status == PIPE_UNDEFINED) return 0; - if(monitoring) + if (monitoring) pipe->monitoring = 0; else pipe->references--; - if((pipe->references <= 0) && (pipe->monitoring == 0)) { + if ((pipe->references <= 0) && (pipe->monitoring == 0)) { struct mixart_msg request; struct mixart_delete_group_resp delete_resp; /* release the clock */ err = mixart_set_clock( mgr, pipe, 0); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "mixart_set_clock(0) return error!\n"); } /* stop the pipe */ err = mixart_set_pipe_state(mgr, pipe, 0); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "error stopping pipe!\n"); } @@ -376,7 +376,7 @@ static int mixart_set_stream_state(struct mixart_stream *stream, int start) struct mixart_stream_state_req stream_state_req; struct mixart_msg request; - if(!stream->substream) + if (!stream->substream) return -EINVAL; memset(&stream_state_req, 0, sizeof(stream_state_req)); @@ -416,7 +416,7 @@ static int snd_mixart_trigger(struct snd_pcm_substream *subs, int cmd) snd_printdd("SNDRV_PCM_TRIGGER_START\n"); /* START_STREAM */ - if( mixart_set_stream_state(stream, 1) ) + if (mixart_set_stream_state(stream, 1)) return -EINVAL; stream->status = MIXART_STREAM_STATUS_RUNNING; @@ -425,7 +425,7 @@ static int snd_mixart_trigger(struct snd_pcm_substream *subs, int cmd) case SNDRV_PCM_TRIGGER_STOP: /* STOP_STREAM */ - if( mixart_set_stream_state(stream, 0) ) + if (mixart_set_stream_state(stream, 0)) return -EINVAL; stream->status = MIXART_STREAM_STATUS_OPEN; @@ -479,12 +479,12 @@ static int snd_mixart_prepare(struct snd_pcm_substream *subs) /* only the first stream can choose the sample rate */ /* the further opened streams will be limited to its frequency (see open) */ - if(chip->mgr->ref_count_rate == 1) + if (chip->mgr->ref_count_rate == 1) chip->mgr->sample_rate = subs->runtime->rate; /* set the clock only once (first stream) on the same pipe */ - if(stream->pipe->references == 1) { - if( mixart_set_clock(chip->mgr, stream->pipe, subs->runtime->rate) ) + if (stream->pipe->references == 1) { + if (mixart_set_clock(chip->mgr, stream->pipe, subs->runtime->rate)) return -EINVAL; } @@ -508,7 +508,7 @@ static int mixart_set_format(struct mixart_stream *stream, snd_pcm_format_t form stream_param.number_of_channel = stream->channels; stream_param.sampling_freq = chip->mgr->sample_rate; - if(stream_param.sampling_freq == 0) + if (stream_param.sampling_freq == 0) stream_param.sampling_freq = 44100; /* if frequency not yet defined, use some default */ switch(format){ @@ -564,7 +564,7 @@ static int mixart_set_format(struct mixart_stream *stream, snd_pcm_format_t form request.size = sizeof(stream_param); err = snd_mixart_send_msg(chip->mgr, &request, sizeof(resp), &resp); - if((err < 0) || resp.error_code) { + if ((err < 0) || resp.error_code) { snd_printk(KERN_ERR "MSG_STREAM_SET_INPUT_STAGE_PARAM err=%x; resp=%x\n", err, resp.error_code); return -EINVAL; } @@ -594,9 +594,9 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, mutex_lock(&mgr->setup_mutex); /* update the stream levels */ - if( stream->pcm_number <= MIXART_PCM_DIGITAL ) { + if (stream->pcm_number <= MIXART_PCM_DIGITAL) { int is_aes = stream->pcm_number > MIXART_PCM_ANALOG; - if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) + if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) mixart_update_playback_stream_level(chip, is_aes, subs->number); else mixart_update_capture_stream_level( chip, is_aes); @@ -606,7 +606,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, /* set the format to the board */ err = mixart_set_format(stream, format); - if(err < 0) { + if (err < 0) { return err; } @@ -616,10 +616,10 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, if (err > 0) { struct mixart_bufferinfo *bufferinfo; int i = (chip->chip_idx * MIXART_MAX_STREAM_PER_CARD) + (stream->pcm_number * (MIXART_PLAYBACK_STREAMS+MIXART_CAPTURE_STREAMS)) + subs->number; - if( subs->stream == SNDRV_PCM_STREAM_CAPTURE ) { + if (subs->stream == SNDRV_PCM_STREAM_CAPTURE) { i += MIXART_PLAYBACK_STREAMS; /* in array capture is behind playback */ } - + bufferinfo = (struct mixart_bufferinfo *)chip->mgr->bufferinfo.area; bufferinfo[i].buffer_address = subs->runtime->dma_addr; bufferinfo[i].available_length = subs->runtime->dma_bytes; @@ -704,7 +704,7 @@ static int snd_mixart_playback_open(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - if ( pcm == chip->pcm ) { + if (pcm == chip->pcm) { pcm_number = MIXART_PCM_ANALOG; runtime->hw = snd_mixart_analog_caps; } else { @@ -734,7 +734,7 @@ static int snd_mixart_playback_open(struct snd_pcm_substream *subs) /* start the pipe if necessary */ err = mixart_set_pipe_state(chip->mgr, pipe, 1); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "error starting pipe!\n"); snd_mixart_kill_ref_pipe(chip->mgr, pipe, 0); err = -EINVAL; @@ -753,13 +753,13 @@ static int snd_mixart_playback_open(struct snd_pcm_substream *subs) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64); /* if a sample rate is already used, another stream cannot change */ - if(mgr->ref_count_rate++) { - if(mgr->sample_rate) { + if (mgr->ref_count_rate++) { + if (mgr->sample_rate) { runtime->hw.rate_min = runtime->hw.rate_max = mgr->sample_rate; } } - _exit_open: +_exit_open: mutex_unlock(&mgr->setup_mutex); return err; @@ -779,7 +779,7 @@ static int snd_mixart_capture_open(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - if ( pcm == chip->pcm ) { + if (pcm == chip->pcm) { pcm_number = MIXART_PCM_ANALOG; runtime->hw = snd_mixart_analog_caps; } else { @@ -812,7 +812,7 @@ static int snd_mixart_capture_open(struct snd_pcm_substream *subs) /* start the pipe if necessary */ err = mixart_set_pipe_state(chip->mgr, pipe, 1); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "error starting pipe!\n"); snd_mixart_kill_ref_pipe(chip->mgr, pipe, 0); err = -EINVAL; @@ -831,13 +831,13 @@ static int snd_mixart_capture_open(struct snd_pcm_substream *subs) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64); /* if a sample rate is already used, another stream cannot change */ - if(mgr->ref_count_rate++) { - if(mgr->sample_rate) { + if (mgr->ref_count_rate++) { + if (mgr->sample_rate) { runtime->hw.rate_min = runtime->hw.rate_max = mgr->sample_rate; } } - _exit_open: +_exit_open: mutex_unlock(&mgr->setup_mutex); return err; @@ -856,7 +856,7 @@ static int snd_mixart_close(struct snd_pcm_substream *subs) snd_printdd("snd_mixart_close C%d/P%d/Sub%d\n", chip->chip_idx, stream->pcm_number, subs->number); /* sample rate released */ - if(--mgr->ref_count_rate == 0) { + if (--mgr->ref_count_rate == 0) { mgr->sample_rate = 0; } @@ -1011,7 +1011,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * }; mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { + if (!chip) { snd_printk(KERN_ERR "cannot allocate chip\n"); return -ENOMEM; } @@ -1038,7 +1038,7 @@ int snd_mixart_create_pcm(struct snd_mixart* chip) if (err < 0) return err; - if(chip->mgr->board_type == MIXART_DAUGHTER_TYPE_AES) { + if (chip->mgr->board_type == MIXART_DAUGHTER_TYPE_AES) { err = snd_mixart_pcm_digital(chip); if (err < 0) @@ -1068,7 +1068,7 @@ static int snd_mixart_free(struct mixart_mgr *mgr) free_irq(mgr->irq, mgr); /* reset board if some firmware was loaded */ - if(mgr->dsp_loaded) { + if (mgr->dsp_loaded) { snd_mixart_reset_board(mgr); snd_printdd("reset miXart !\n"); } @@ -1081,12 +1081,12 @@ static int snd_mixart_free(struct mixart_mgr *mgr) pci_release_regions(mgr->pci); /* free flowarray */ - if(mgr->flowinfo.area) { + if (mgr->flowinfo.area) { snd_dma_free_pages(&mgr->flowinfo); mgr->flowinfo.area = NULL; } /* free bufferarray */ - if(mgr->bufferinfo.area) { + if (mgr->bufferinfo.area) { snd_dma_free_pages(&mgr->bufferinfo); mgr->bufferinfo.area = NULL; } @@ -1120,7 +1120,7 @@ static long long snd_mixart_BA0_llseek(struct snd_info_entry *entry, default: return -EINVAL; } - if(file->f_pos > MIXART_BA0_SIZE) + if (file->f_pos > MIXART_BA0_SIZE) file->f_pos = MIXART_BA0_SIZE; return file->f_pos; } @@ -1146,7 +1146,7 @@ static long long snd_mixart_BA1_llseek(struct snd_info_entry *entry, default: return -EINVAL; } - if(file->f_pos > MIXART_BA1_SIZE) + if (file->f_pos > MIXART_BA1_SIZE) file->f_pos = MIXART_BA1_SIZE; return file->f_pos; } @@ -1161,11 +1161,11 @@ static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private struct mixart_mgr *mgr = entry->private_data; count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (count <= 0) return 0; - if(pos + count > MIXART_BA0_SIZE) + if (pos + count > MIXART_BA0_SIZE) count = (long)(MIXART_BA0_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count)) + if (copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count)) return -EFAULT; return count; } @@ -1180,11 +1180,11 @@ static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private struct mixart_mgr *mgr = entry->private_data; count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (count <= 0) return 0; - if(pos + count > MIXART_BA1_SIZE) + if (pos + count > MIXART_BA1_SIZE) count = (long)(MIXART_BA1_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count)) + if (copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count)) return -EFAULT; return count; } @@ -1200,25 +1200,25 @@ static struct snd_info_entry_ops snd_mixart_proc_ops_BA1 = { }; -static void snd_mixart_proc_read(struct snd_info_entry *entry, +static void snd_mixart_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_mixart *chip = entry->private_data; - u32 ref; + struct snd_mixart *chip = entry->private_data; + u32 ref; snd_iprintf(buffer, "Digigram miXart (alsa card %d)\n\n", chip->chip_idx); /* stats available when embedded OS is running */ if (chip->mgr->dsp_loaded & ( 1 << MIXART_MOTHERBOARD_ELF_INDEX)) { snd_iprintf(buffer, "- hardware -\n"); - switch (chip->mgr->board_type ) { + switch (chip->mgr->board_type) { case MIXART_DAUGHTER_TYPE_NONE : snd_iprintf(buffer, "\tmiXart8 (no daughter board)\n\n"); break; case MIXART_DAUGHTER_TYPE_AES : snd_iprintf(buffer, "\tmiXart8 AES/EBU\n\n"); break; case MIXART_DAUGHTER_TYPE_COBRANET : snd_iprintf(buffer, "\tmiXart8 Cobranet\n\n"); break; default: snd_iprintf(buffer, "\tUNKNOWN!\n\n"); break; } - snd_iprintf(buffer, "- system load -\n"); + snd_iprintf(buffer, "- system load -\n"); /* get perf reference */ @@ -1241,18 +1241,18 @@ static void __devinit snd_mixart_proc_init(struct snd_mixart *chip) struct snd_info_entry *entry; /* text interface to read perf and temp meters */ - if (! snd_card_proc_new(chip->card, "board_info", &entry)) { + if (!snd_card_proc_new(chip->card, "board_info", &entry)) { entry->private_data = chip; entry->c.text.read = snd_mixart_proc_read; } - if (! snd_card_proc_new(chip->card, "mixart_BA0", &entry)) { + if (!snd_card_proc_new(chip->card, "mixart_BA0", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; - entry->private_data = chip->mgr; + entry->private_data = chip->mgr; entry->c.ops = &snd_mixart_proc_ops_BA0; entry->size = MIXART_BA0_SIZE; } - if (! snd_card_proc_new(chip->card, "mixart_BA1", &entry)) { + if (!snd_card_proc_new(chip->card, "mixart_BA1", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = chip->mgr; entry->c.ops = &snd_mixart_proc_ops_BA1; @@ -1278,7 +1278,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, */ if (dev >= SNDRV_CARDS) return -ENODEV; - if (! enable[dev]) { + if (!enable[dev]) { dev++; return -ENOENT; } @@ -1298,7 +1298,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, /* */ mgr = kzalloc(sizeof(*mgr), GFP_KERNEL); - if (! mgr) { + if (!mgr) { pci_disable_device(pci); return -ENOMEM; } @@ -1367,7 +1367,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : "MIXART", i); card = snd_card_new(idx, tmpid, THIS_MODULE, 0); - if (! card) { + if (!card) { snd_printk(KERN_ERR "cannot allocate the card %d\n", i); snd_mixart_free(mgr); return -ENOMEM; @@ -1382,7 +1382,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, return err; } - if(i==0) { + if (i==0) { /* init proc interface only for chip0 */ snd_mixart_proc_init(mgr->chip[i]); } @@ -1398,7 +1398,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, /* create array of streaminfo */ size = PAGE_ALIGN( (MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS * - sizeof(struct mixart_flowinfo)) ); + sizeof(struct mixart_flowinfo))); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), size, &mgr->flowinfo) < 0) { snd_mixart_free(mgr); @@ -1409,7 +1409,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, /* create array of bufferinfo */ size = PAGE_ALIGN( (MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS * - sizeof(struct mixart_bufferinfo)) ); + sizeof(struct mixart_bufferinfo))); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), size, &mgr->bufferinfo) < 0) { snd_mixart_free(mgr); diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index b9a06c2..26b5f46 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -23,7 +23,7 @@ #include <linux/interrupt.h> #include <linux/mutex.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include "mixart.h" #include "mixart_hwdep.h" @@ -65,7 +65,7 @@ static int retrieve_msg_frame(struct mixart_mgr *mgr, u32 *msg_frame) /* increment the tail index */ tailptr += 4; - if( tailptr >= (MSG_OUTBOUND_POST_STACK+MSG_BOUND_STACK_SIZE) ) + if (tailptr >= (MSG_OUTBOUND_POST_STACK+MSG_BOUND_STACK_SIZE)) tailptr = MSG_OUTBOUND_POST_STACK; writel_be(tailptr, MIXART_MEM(mgr, MSG_OUTBOUND_POST_TAIL)); @@ -73,7 +73,7 @@ static int retrieve_msg_frame(struct mixart_mgr *mgr, u32 *msg_frame) } static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, - u32 msg_frame_address ) + u32 msg_frame_address) { unsigned long flags; u32 headptr; @@ -92,7 +92,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, resp->uid.object_id = readl_be(MIXART_MEM(mgr, msg_frame_address + 8)); /* uidDest */ resp->uid.desc = readl_be(MIXART_MEM(mgr, msg_frame_address + 12)); /* */ - if( (size < MSG_DESCRIPTOR_SIZE) || (resp->size < (size - MSG_DESCRIPTOR_SIZE))) { + if ((size < MSG_DESCRIPTOR_SIZE) || (resp->size < (size - MSG_DESCRIPTOR_SIZE))) { err = -EINVAL; snd_printk(KERN_ERR "problem with response size = %d\n", size); goto _clean_exit; @@ -105,7 +105,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, /* swap if necessary */ #ifndef __BIG_ENDIAN size /= 4; /* u32 size */ - for(i=0; i < size; i++) { + for (i = 0; i < size; i++) { ((u32*)resp->data)[i] = be32_to_cpu(((u32*)resp->data)[i]); } #endif @@ -115,7 +115,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, */ headptr = readl_be(MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD)); - if( (headptr < MSG_OUTBOUND_FREE_STACK) || ( headptr >= (MSG_OUTBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE))) { + if ((headptr < MSG_OUTBOUND_FREE_STACK) || ( headptr >= (MSG_OUTBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE))) { err = -EINVAL; goto _clean_exit; } @@ -125,12 +125,12 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, /* increment the outbound free head */ headptr += 4; - if( headptr >= (MSG_OUTBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE) ) + if (headptr >= (MSG_OUTBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE)) headptr = MSG_OUTBOUND_FREE_STACK; writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD)); - _clean_exit: +_clean_exit: spin_unlock_irqrestore(&mgr->msg_lock, flags); return err; @@ -141,7 +141,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, * send a message to miXart. return: the msg_frame used for this message */ /* call with mgr->msg_lock held! */ -static int send_msg( struct mixart_mgr *mgr, +static int send_msg(struct mixart_mgr *mgr, struct mixart_msg *msg, int max_answersize, int mark_pending, @@ -165,7 +165,7 @@ static int send_msg( struct mixart_mgr *mgr, return -EBUSY; } - if( (tailptr < MSG_INBOUND_FREE_STACK) || (tailptr >= (MSG_INBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE))) { + if ((tailptr < MSG_INBOUND_FREE_STACK) || (tailptr >= (MSG_INBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE))) { return -EINVAL; } @@ -174,7 +174,7 @@ static int send_msg( struct mixart_mgr *mgr, /* increment the inbound free tail */ tailptr += 4; - if( tailptr >= (MSG_INBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE) ) + if (tailptr >= (MSG_INBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE)) tailptr = MSG_INBOUND_FREE_STACK; writel_be(tailptr, MIXART_MEM(mgr, MSG_INBOUND_FREE_TAIL)); @@ -182,24 +182,24 @@ static int send_msg( struct mixart_mgr *mgr, /* TODO : use memcpy_toio() with intermediate buffer to copy the message */ /* copy message descriptor to card memory */ - writel_be( msg->size + MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address) ); /* size of descriptor + request */ - writel_be( msg->message_id , MIXART_MEM(mgr, msg_frame_address + 4) ); /* dwMessageID */ - writel_be( msg->uid.object_id, MIXART_MEM(mgr, msg_frame_address + 8) ); /* uidDest */ - writel_be( msg->uid.desc, MIXART_MEM(mgr, msg_frame_address + 12) ); /* */ - writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 16) ); /* SizeHeader */ - writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 20) ); /* OffsetDLL_T16 */ - writel_be( msg->size, MIXART_MEM(mgr, msg_frame_address + 24) ); /* SizeDLL_T16 */ - writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 28) ); /* OffsetDLL_DRV */ - writel_be( 0, MIXART_MEM(mgr, msg_frame_address + 32) ); /* SizeDLL_DRV */ - writel_be( MSG_DESCRIPTOR_SIZE + max_answersize, MIXART_MEM(mgr, msg_frame_address + 36) ); /* dwExpectedAnswerSize */ + writel_be( msg->size + MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address)); /* size of descriptor + request */ + writel_be( msg->message_id , MIXART_MEM(mgr, msg_frame_address + 4)); /* dwMessageID */ + writel_be( msg->uid.object_id, MIXART_MEM(mgr, msg_frame_address + 8)); /* uidDest */ + writel_be( msg->uid.desc, MIXART_MEM(mgr, msg_frame_address + 12)); /* */ + writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 16)); /* SizeHeader */ + writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 20)); /* OffsetDLL_T16 */ + writel_be( msg->size, MIXART_MEM(mgr, msg_frame_address + 24)); /* SizeDLL_T16 */ + writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 28)); /* OffsetDLL_DRV */ + writel_be( 0, MIXART_MEM(mgr, msg_frame_address + 32)); /* SizeDLL_DRV */ + writel_be( MSG_DESCRIPTOR_SIZE + max_answersize, MIXART_MEM(mgr, msg_frame_address + 36)); /* dwExpectedAnswerSize */ /* copy message data to card memory */ - for( i=0; i < msg->size; i+=4 ) { - writel_be( *(u32*)(msg->data + i), MIXART_MEM(mgr, MSG_HEADER_SIZE + msg_frame_address + i) ); + for (i = 0; i < msg->size; i += 4) { + writel_be( *(u32*)(msg->data + i), MIXART_MEM(mgr, MSG_HEADER_SIZE + msg_frame_address + i) ); } - if( mark_pending ) { - if( *msg_event ) { + if (mark_pending) { + if (*msg_event) { /* the pending event is the notification we wait for ! */ mgr->pending_event = *msg_event; } @@ -218,7 +218,7 @@ static int send_msg( struct mixart_mgr *mgr, /* post the frame */ headptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_POST_HEAD)); - if( (headptr < MSG_INBOUND_POST_STACK) || (headptr >= (MSG_INBOUND_POST_STACK+MSG_BOUND_STACK_SIZE))) { + if ((headptr < MSG_INBOUND_POST_STACK) || (headptr >= (MSG_INBOUND_POST_STACK+MSG_BOUND_STACK_SIZE))) { return -EINVAL; } @@ -226,7 +226,7 @@ static int send_msg( struct mixart_mgr *mgr, /* increment the inbound post head */ headptr += 4; - if( headptr >= (MSG_INBOUND_POST_STACK+MSG_BOUND_STACK_SIZE) ) + if (headptr >= (MSG_INBOUND_POST_STACK+MSG_BOUND_STACK_SIZE)) headptr = MSG_INBOUND_POST_STACK; writel_be(headptr, MIXART_MEM(mgr, MSG_INBOUND_POST_HEAD)); @@ -262,7 +262,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int timeout = schedule_timeout(MSG_TIMEOUT_JIFFIES); remove_wait_queue(&mgr->msg_sleep, &wait); - if (! timeout) { + if (!timeout) { /* error - no ack */ mutex_unlock(&mgr->msg_mutex); snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame); @@ -277,7 +277,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int err = get_msg(mgr, &resp, msg_frame); - if( request->message_id != resp.message_id ) + if (request->message_id != resp.message_id) snd_printk(KERN_ERR "REPONSE ERROR!\n"); mutex_unlock(&mgr->msg_mutex); @@ -306,7 +306,7 @@ int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr, spin_lock_irq(&mgr->msg_lock); /* send the message */ err = send_msg(mgr, request, MSG_DEFAULT_SIZE, 1, &notif_event); /* send and mark the notification event pending */ - if(err) { + if (err) { spin_unlock_irq(&mgr->msg_lock); mutex_unlock(&mgr->msg_mutex); return err; @@ -318,7 +318,7 @@ int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr, timeout = schedule_timeout(MSG_TIMEOUT_JIFFIES); remove_wait_queue(&mgr->msg_sleep, &wait); - if (! timeout) { + if (!timeout) { /* error - no ack */ mutex_unlock(&mgr->msg_mutex); snd_printk(KERN_ERR "error: notification %x not received\n", notif_event); @@ -354,7 +354,7 @@ static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4]; void snd_mixart_msg_tasklet(unsigned long arg) { - struct mixart_mgr *mgr = ( struct mixart_mgr*)(arg); + struct mixart_mgr *mgr = (struct mixart_mgr*)(arg); struct mixart_msg resp; u32 msg, addr, type; int err; @@ -377,7 +377,7 @@ void snd_mixart_msg_tasklet(unsigned long arg) resp.data = mixart_msg_data; resp.size = sizeof(mixart_msg_data); err = get_msg(mgr, &resp, addr); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "tasklet: error(%d) reading mf %x\n", err, msg); break; } @@ -387,7 +387,7 @@ void snd_mixart_msg_tasklet(unsigned long arg) case MSG_STREAM_START_OUTPUT_STAGE_PACKET: case MSG_STREAM_STOP_INPUT_STAGE_PACKET: case MSG_STREAM_STOP_OUTPUT_STAGE_PACKET: - if(mixart_msg_data[0]) + if (mixart_msg_data[0]) snd_printk(KERN_ERR "tasklet : error MSG_STREAM_ST***_***PUT_STAGE_PACKET status=%x\n", mixart_msg_data[0]); break; default: @@ -425,7 +425,7 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) spin_lock(&mgr->lock); it_reg = readl_le(MIXART_REG(mgr, MIXART_PCI_OMISR_OFFSET)); - if( !(it_reg & MIXART_OIDI) ) { + if (!(it_reg & MIXART_OIDI)) { /* this device did not cause the interrupt */ spin_unlock(&mgr->lock); return IRQ_NONE; @@ -439,7 +439,7 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) writel(it_reg, MIXART_REG(mgr, MIXART_PCI_ODBR_OFFSET)); /* clear interrupt */ - writel_le( MIXART_OIDI, MIXART_REG(mgr, MIXART_PCI_OMISR_OFFSET) ); + writel_le( MIXART_OIDI, MIXART_REG(mgr, MIXART_PCI_OMISR_OFFSET)); /* process interrupt */ while (retrieve_msg_frame(mgr, &msg)) { @@ -450,17 +450,17 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) resp.data = mixart_msg_data; resp.size = sizeof(mixart_msg_data); err = get_msg(mgr, &resp, msg & ~MSG_TYPE_MASK); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "interrupt: error(%d) reading mf %x\n", err, msg); break; } - if(resp.message_id == MSG_SERVICES_TIMER_NOTIFY) { + if (resp.message_id == MSG_SERVICES_TIMER_NOTIFY) { int i; struct mixart_timer_notify *notify; notify = (struct mixart_timer_notify *)mixart_msg_data; - for(i=0; i<notify->stream_count; i++) { + for (i = 0; i < notify->stream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; unsigned int chip_number = (buffer_id & MIXART_NOTIFY_CARD_MASK) >> MIXART_NOTIFY_CARD_OFFSET; /* card0 to 3 */ @@ -503,9 +503,9 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) stream->abs_period_elapsed = new_elapse_pos; } } - stream->buf_period_frag = (u32)( sample_count - stream->abs_period_elapsed ); + stream->buf_period_frag = (u32)(sample_count - stream->abs_period_elapsed); - if(elapsed) { + if (elapsed) { spin_unlock(&mgr->lock); snd_pcm_period_elapsed(stream->substream); spin_lock(&mgr->lock); @@ -514,12 +514,12 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) } break; } - if(resp.message_id == MSG_SERVICES_REPORT_TRACES) { - if(resp.size > 1) { + if (resp.message_id == MSG_SERVICES_REPORT_TRACES) { + if (resp.size > 1) { #ifndef __BIG_ENDIAN /* Traces are text: the swapped msg_data has to be swapped back ! */ int i; - for(i=0; i<(resp.size/4); i++) { + for (i = 0; i < (resp.size/4); i++) { (mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]); } #endif @@ -533,7 +533,7 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) break; case MSG_TYPE_NOTIFY: - if(msg & MSG_CANCEL_NOTIFY_MASK) { + if (msg & MSG_CANCEL_NOTIFY_MASK) { msg &= ~MSG_CANCEL_NOTIFY_MASK; snd_printk(KERN_ERR "canceled notification %x !\n", msg); } @@ -541,7 +541,7 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ spin_lock(&mgr->msg_lock); - if( (msg & ~MSG_TYPE_MASK) == mgr->pending_event ) { + if ((msg & ~MSG_TYPE_MASK) == mgr->pending_event) { wake_up(&mgr->msg_sleep); mgr->pending_event = 0; } @@ -573,11 +573,11 @@ irqreturn_t snd_mixart_interrupt(int irq, void *dev_id) void snd_mixart_init_mailbox(struct mixart_mgr *mgr) { - writel( 0, MIXART_MEM( mgr, MSG_HOST_RSC_PROTECTION ) ); - writel( 0, MIXART_MEM( mgr, MSG_AGENT_RSC_PROTECTION ) ); + writel( 0, MIXART_MEM( mgr, MSG_HOST_RSC_PROTECTION )); + writel( 0, MIXART_MEM( mgr, MSG_AGENT_RSC_PROTECTION )); /* allow outbound messagebox to generate interrupts */ - if(mgr->irq >= 0) { + if (mgr->irq >= 0) { writel_le( MIXART_ALLOW_OUTBOUND_DOORBELL, MIXART_REG( mgr, MIXART_PCI_OMIMR_OFFSET)); } return; @@ -593,6 +593,6 @@ void snd_mixart_exit_mailbox(struct mixart_mgr *mgr) void snd_mixart_reset_board(struct mixart_mgr *mgr) { /* reset miXart */ - writel_be( 1, MIXART_REG(mgr, MIXART_BA1_BRUTAL_RESET_OFFSET) ); + writel_be( 1, MIXART_REG(mgr, MIXART_BA1_BRUTAL_RESET_OFFSET)); return; } diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 3782b52..18639aa 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -24,7 +24,7 @@ #include <linux/pci.h> #include <linux/firmware.h> #include <linux/vmalloc.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include "mixart.h" #include "mixart_mixer.h" @@ -53,20 +53,20 @@ static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr, cond_resched(); read = readl_be( MIXART_MEM( mgr, offset )); - if(is_egal) { - if(read == value) return 0; + if (is_egal) { + if (read == value) return 0; } else { /* wait for different value */ - if(read != value) return 0; + if (read != value) return 0; } - } while ( time_after_eq(end_time, jiffies) ); + } while ( time_after_eq(end_time, jiffies)); return -EBUSY; } /* - structures needed to upload elf code packets + structures needed to upload elf code packets */ struct snd_mixart_elf32_ehdr { u8 e_ident[16]; @@ -96,27 +96,27 @@ struct snd_mixart_elf32_phdr { u32 p_align; }; -static int mixart_load_elf(struct mixart_mgr *mgr, const struct firmware *dsp ) +static int mixart_load_elf(struct mixart_mgr *mgr, const struct firmware *dsp) { char elf32_magic_number[4] = {0x7f,'E','L','F'}; struct snd_mixart_elf32_ehdr *elf_header; int i; elf_header = (struct snd_mixart_elf32_ehdr *)dsp->data; - for( i=0; i<4; i++ ) - if ( elf32_magic_number[i] != elf_header->e_ident[i] ) + for (i = 0; i < 4; i++) + if (elf32_magic_number[i] != elf_header->e_ident[i]) return -EINVAL; - if( elf_header->e_phoff != 0 ) { + if (elf_header->e_phoff != 0) { struct snd_mixart_elf32_phdr elf_programheader; - for( i=0; i < be16_to_cpu(elf_header->e_phnum); i++ ) { + for (i = 0; i < be16_to_cpu(elf_header->e_phnum); i++) { u32 pos = be32_to_cpu(elf_header->e_phoff) + (u32)(i * be16_to_cpu(elf_header->e_phentsize)); - memcpy( &elf_programheader, dsp->data + pos, sizeof(elf_programheader) ); + memcpy( &elf_programheader, dsp->data + pos, sizeof(elf_programheader)); - if(elf_programheader.p_type != 0) { - if( elf_programheader.p_filesz != 0 ) { + if (elf_programheader.p_type != 0) { + if (elf_programheader.p_filesz != 0) { memcpy_toio( MIXART_MEM( mgr, be32_to_cpu(elf_programheader.p_vaddr)), dsp->data + be32_to_cpu( elf_programheader.p_offset ), be32_to_cpu( elf_programheader.p_filesz )); @@ -147,7 +147,7 @@ static int mixart_enum_connectors(struct mixart_mgr *mgr) connector = kmalloc(sizeof(*connector), GFP_KERNEL); audio_info_req = kmalloc(sizeof(*audio_info_req), GFP_KERNEL); audio_info = kmalloc(sizeof(*audio_info), GFP_KERNEL); - if (! connector || ! audio_info_req || ! audio_info) { + if (!connector || !audio_info_req || !audio_info) { err = -ENOMEM; goto __error; } @@ -162,21 +162,21 @@ static int mixart_enum_connectors(struct mixart_mgr *mgr) request.size = 0; err = snd_mixart_send_msg(mgr, &request, sizeof(*connector), connector); - if((err < 0) || (connector->error_code) || (connector->uid_count > MIXART_MAX_PHYS_CONNECTORS)) { + if ((err < 0) || (connector->error_code) || (connector->uid_count > MIXART_MAX_PHYS_CONNECTORS)) { snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_PLAY_CONNECTOR\n"); err = -EINVAL; goto __error; } - for(k=0; k < connector->uid_count; k++) { + for (k=0; k < connector->uid_count; k++) { struct mixart_pipe *pipe; - if(k < MIXART_FIRST_DIG_AUDIO_ID) { + if (k < MIXART_FIRST_DIG_AUDIO_ID) { pipe = &mgr->chip[k/2]->pipe_out_ana; } else { pipe = &mgr->chip[(k-MIXART_FIRST_DIG_AUDIO_ID)/2]->pipe_out_dig; } - if(k & 1) { + if (k & 1) { pipe->uid_right_connector = connector->uid[k]; /* odd */ } else { pipe->uid_left_connector = connector->uid[k]; /* even */ @@ -191,7 +191,7 @@ static int mixart_enum_connectors(struct mixart_mgr *mgr) request.size = sizeof(*audio_info_req); err = snd_mixart_send_msg(mgr, &request, sizeof(*audio_info), audio_info); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "error MSG_CONNECTOR_GET_AUDIO_INFO\n"); goto __error; } @@ -204,21 +204,21 @@ static int mixart_enum_connectors(struct mixart_mgr *mgr) request.size = 0; err = snd_mixart_send_msg(mgr, &request, sizeof(*connector), connector); - if((err < 0) || (connector->error_code) || (connector->uid_count > MIXART_MAX_PHYS_CONNECTORS)) { + if ((err < 0) || (connector->error_code) || (connector->uid_count > MIXART_MAX_PHYS_CONNECTORS)) { snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_RECORD_CONNECTOR\n"); err = -EINVAL; goto __error; } - for(k=0; k < connector->uid_count; k++) { + for (k=0; k < connector->uid_count; k++) { struct mixart_pipe *pipe; - if(k < MIXART_FIRST_DIG_AUDIO_ID) { + if (k < MIXART_FIRST_DIG_AUDIO_ID) { pipe = &mgr->chip[k/2]->pipe_in_ana; } else { pipe = &mgr->chip[(k-MIXART_FIRST_DIG_AUDIO_ID)/2]->pipe_in_dig; } - if(k & 1) { + if (k & 1) { pipe->uid_right_connector = connector->uid[k]; /* odd */ } else { pipe->uid_left_connector = connector->uid[k]; /* even */ @@ -233,7 +233,7 @@ static int mixart_enum_connectors(struct mixart_mgr *mgr) request.size = sizeof(*audio_info_req); err = snd_mixart_send_msg(mgr, &request, sizeof(*audio_info), audio_info); - if( err < 0 ) { + if (err < 0) { snd_printk(KERN_ERR "error MSG_CONNECTOR_GET_AUDIO_INFO\n"); goto __error; } @@ -241,7 +241,7 @@ static int mixart_enum_connectors(struct mixart_mgr *mgr) } err = 0; - __error: +__error: kfree(connector); kfree(audio_info_req); kfree(audio_info); @@ -269,7 +269,7 @@ static int mixart_enum_physio(struct mixart_mgr *mgr) err = snd_mixart_send_msg(mgr, &request, sizeof(console_mgr), &console_mgr); - if( (err < 0) || (console_mgr.error_code != 0) ) { + if ((err < 0) || (console_mgr.error_code != 0)) { snd_printk(KERN_DEBUG "error MSG_CONSOLE_GET_CLOCK_UID : err=%x\n", console_mgr.error_code); return -EINVAL; } @@ -283,8 +283,8 @@ static int mixart_enum_physio(struct mixart_mgr *mgr) request.size = sizeof(console_mgr.uid); err = snd_mixart_send_msg(mgr, &request, sizeof(phys_io), &phys_io); - if( (err < 0) || ( phys_io.error_code != 0 ) ) { - snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_PHYSICAL_IO err(%x) error_code(%x)\n", err, phys_io.error_code ); + if ((err < 0) || ( phys_io.error_code != 0 )) { + snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_PHYSICAL_IO err(%x) error_code(%x)\n", err, phys_io.error_code); return -EINVAL; } @@ -292,9 +292,9 @@ static int mixart_enum_physio(struct mixart_mgr *mgr) if (phys_io.nb_uid < MIXART_MAX_CARDS * 2) return -EINVAL; - for(k=0; k<mgr->num_cards; k++) { + for (k=0; k<mgr->num_cards; k++) { mgr->chip[k]->uid_in_analog_physio = phys_io.uid[k]; - mgr->chip[k]->uid_out_analog_physio = phys_io.uid[phys_io.nb_uid/2 + k]; + mgr->chip[k]->uid_out_analog_physio = phys_io.uid[phys_io.nb_uid/2 + k]; } return 0; @@ -307,9 +307,9 @@ static int mixart_first_init(struct mixart_mgr *mgr) int err; struct mixart_msg request; - if((err = mixart_enum_connectors(mgr)) < 0) return err; + if ((err = mixart_enum_connectors(mgr)) < 0) return err; - if((err = mixart_enum_physio(mgr)) < 0) return err; + if ((err = mixart_enum_physio(mgr)) < 0) return err; /* send a synchro command to card (necessary to do this before first MSG_STREAM_START_STREAM_GRP_PACKET) */ /* though why not here */ @@ -319,7 +319,7 @@ static int mixart_first_init(struct mixart_mgr *mgr) request.size = 0; /* this command has no data. response is a 32 bit status */ err = snd_mixart_send_msg(mgr, &request, sizeof(k), &k); - if( (err < 0) || (k != 0) ) { + if ((err < 0) || (k != 0)) { snd_printk(KERN_ERR "error MSG_SYSTEM_SEND_SYNCHRO_CMD\n"); return err == 0 ? -EINVAL : err; } @@ -345,22 +345,22 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET )); /* motherboard xilinx status 5 will say that the board is performing a reset */ - if( status_xilinx == 5 ) { - snd_printk( KERN_ERR "miXart is resetting !\n"); + if (status_xilinx == 5) { + snd_printk(KERN_ERR "miXart is resetting !\n"); return -EAGAIN; /* try again later */ } switch (index) { case MIXART_MOTHERBOARD_XLX_INDEX: - /* xilinx already loaded ? */ - if( status_xilinx == 4 ) { - snd_printk( KERN_DEBUG "xilinx is already loaded !\n"); + /* xilinx already loaded ? */ + if (status_xilinx == 4) { + snd_printk(KERN_DEBUG "xilinx is already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_xilinx != 0 ) { - snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx); + if (status_xilinx != 0) { + snd_printk(KERN_ERR "xilinx load error ! status = %d\n", status_xilinx); return -EIO; /* modprob -r may help ? */ } @@ -380,7 +380,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* copy xilinx code */ memcpy_toio( MIXART_MEM( mgr, MIXART_MOTHERBOARD_XLX_BASE_ADDRESS), dsp->data, dsp->size); - + /* set xilinx status to copy finished */ writel_be( 2, MIXART_MEM( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET )); @@ -389,33 +389,33 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_ELF_INDEX: - if( status_elf == 4 ) { - snd_printk( KERN_DEBUG "elf file already loaded !\n"); + if (status_elf == 4) { + snd_printk(KERN_DEBUG "elf file already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_elf != 0 ) { - snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf); + if (status_elf != 0) { + snd_printk(KERN_ERR "elf load error ! status = %d\n", status_elf); return -EIO; /* modprob -r may help ? */ } /* wait for xilinx status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */ if (err < 0) { - snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n"); + snd_printk(KERN_ERR "xilinx was not loaded or could not be started\n"); return err; } /* init some data on the card */ - writel_be( 0, MIXART_MEM( mgr, MIXART_PSEUDOREG_BOARDNUMBER ) ); /* set miXart boardnumber to 0 */ - writel_be( 0, MIXART_MEM( mgr, MIXART_FLOWTABLE_PTR ) ); /* reset pointer to flow table on miXart */ + writel_be( 0, MIXART_MEM( mgr, MIXART_PSEUDOREG_BOARDNUMBER )); /* set miXart boardnumber to 0 */ + writel_be( 0, MIXART_MEM( mgr, MIXART_FLOWTABLE_PTR )); /* reset pointer to flow table on miXart */ /* set elf status to copying */ writel_be( 1, MIXART_MEM( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET )); /* process the copying of the elf packets */ - err = mixart_load_elf( mgr, dsp ); + err = mixart_load_elf( mgr, dsp); if (err < 0) return err; /* set elf status to copy finished */ @@ -424,12 +424,12 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for elf status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "elf could not be started\n"); + snd_printk(KERN_ERR "elf could not be started\n"); return err; } /* miXart waits at this point on the pointer to the flow table */ - writel_be( (u32)mgr->flowinfo.addr, MIXART_MEM( mgr, MIXART_FLOWTABLE_PTR ) ); /* give pointer of flow table to miXart */ + writel_be( (u32)mgr->flowinfo.addr, MIXART_MEM( mgr, MIXART_FLOWTABLE_PTR )); /* give pointer of flow table to miXart */ return 0; /* return, another xilinx file has to be loaded before */ @@ -437,15 +437,15 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw default: /* elf and xilinx should be loaded */ - if( (status_elf != 4) || (status_xilinx != 4) ) { - printk( KERN_ERR "xilinx or elf not successfully loaded\n"); + if ((status_elf != 4) || (status_xilinx != 4)) { + printk(KERN_ERR "xilinx or elf not successfully loaded\n"); return -EIO; /* modprob -r may help ? */ } /* wait for daughter detection != 0 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "error starting elf file\n"); + snd_printk(KERN_ERR "error starting elf file\n"); return err; } @@ -455,16 +455,16 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw if (mgr->board_type == MIXART_DAUGHTER_TYPE_NONE) break; /* no daughter board; the file does not have to be loaded, continue after the switch */ - /* only if aesebu daughter board presence (elf code must run) */ - if (mgr->board_type != MIXART_DAUGHTER_TYPE_AES ) + /* only if aesebu daughter board presence (elf code must run) */ + if (mgr->board_type != MIXART_DAUGHTER_TYPE_AES) return -EINVAL; /* daughter should be idle */ - if( status_daught != 0 ) { - printk( KERN_ERR "daughter load error ! status = %d\n", status_daught); + if (status_daught != 0) { + printk(KERN_ERR "daughter load error ! status = %d\n", status_daught); return -EIO; /* modprob -r may help ? */ } - + /* check daughterboard xilinx validity */ if (((u32*)(dsp->data))[0] == 0xffffffff) return -EINVAL; @@ -480,7 +480,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for status == 2 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board load error\n"); + snd_printk(KERN_ERR "daughter board load error\n"); return err; } @@ -502,7 +502,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for daughter status == 3 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board could not be initialised\n"); + snd_printk(KERN_ERR "daughter board could not be initialised\n"); return err; } @@ -512,7 +512,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* first communication with embedded */ err = mixart_first_init(mgr); if (err < 0) { - snd_printk( KERN_ERR "miXart could not be set up\n"); + snd_printk(KERN_ERR "miXart could not be set up\n"); return err; } @@ -615,7 +615,7 @@ static int mixart_hwdep_dsp_load(struct snd_hwdep *hw, fw.size = dsp->length; fw.data = vmalloc(dsp->length); - if (! fw.data) { + if (!fw.data) { snd_printk(KERN_ERR "miXart: cannot allocate image size %d\n", (int)dsp->length); return -ENOMEM; diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h index a46f508..ce1f109 100644 --- a/sound/pci/mixart/mixart_hwdep.h +++ b/sound/pci/mixart/mixart_hwdep.h @@ -36,12 +36,12 @@ /* Daughter board Type */ -#define DAUGHTER_TYPE_MASK 0x0F -#define DAUGHTER_VER_MASK 0xF0 +#define DAUGHTER_TYPE_MASK 0x0F +#define DAUGHTER_VER_MASK 0xF0 #define DAUGHTER_TYPEVER_MASK (DAUGHTER_TYPE_MASK|DAUGHTER_VER_MASK) - -#define MIXART_DAUGHTER_TYPE_NONE 0x00 -#define MIXART_DAUGHTER_TYPE_COBRANET 0x08 + +#define MIXART_DAUGHTER_TYPE_NONE 0x00 +#define MIXART_DAUGHTER_TYPE_COBRANET 0x08 #define MIXART_DAUGHTER_TYPE_AES 0x0E @@ -63,28 +63,28 @@ #define MIXART_PSEUDOREG_PERF_INTERR_LOAD_OFFSET MIXART_PSEUDOREG+0x74 /* interrupt handling load */ /* motherboard xilinx loader info */ -#define MIXART_PSEUDOREG_MXLX_BASE_ADDR_OFFSET MIXART_PSEUDOREG+0x9C /* 0x00600000 */ -#define MIXART_PSEUDOREG_MXLX_SIZE_OFFSET MIXART_PSEUDOREG+0xA0 /* xilinx size in bytes */ -#define MIXART_PSEUDOREG_MXLX_STATUS_OFFSET MIXART_PSEUDOREG+0xA4 /* status = EMBEBBED_STAT_XXX */ +#define MIXART_PSEUDOREG_MXLX_BASE_ADDR_OFFSET MIXART_PSEUDOREG+0x9C /* 0x00600000 */ +#define MIXART_PSEUDOREG_MXLX_SIZE_OFFSET MIXART_PSEUDOREG+0xA0 /* xilinx size in bytes */ +#define MIXART_PSEUDOREG_MXLX_STATUS_OFFSET MIXART_PSEUDOREG+0xA4 /* status = EMBEBBED_STAT_XXX */ /* elf loader info */ -#define MIXART_PSEUDOREG_ELF_STATUS_OFFSET MIXART_PSEUDOREG+0xB0 /* status = EMBEBBED_STAT_XXX */ +#define MIXART_PSEUDOREG_ELF_STATUS_OFFSET MIXART_PSEUDOREG+0xB0 /* status = EMBEBBED_STAT_XXX */ -/* +/* * after the elf code is loaded, and the flowtable info was passed to it, * the driver polls on this address, until it shows 1 (presence) or 2 (absence) * once it is non-zero, the daughter board type may be read */ -#define MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET MIXART_PSEUDOREG+0x990 +#define MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET MIXART_PSEUDOREG+0x990 /* Global info structure */ #define MIXART_PSEUDOREG_DBRD_TYPE_OFFSET MIXART_PSEUDOREG+0x994 /* Type and version of daughterboard */ /* daughterboard xilinx loader info */ -#define MIXART_PSEUDOREG_DXLX_BASE_ADDR_OFFSET MIXART_PSEUDOREG+0x998 /* get the address here where to write the file */ -#define MIXART_PSEUDOREG_DXLX_SIZE_OFFSET MIXART_PSEUDOREG+0x99C /* xilinx size in bytes */ -#define MIXART_PSEUDOREG_DXLX_STATUS_OFFSET MIXART_PSEUDOREG+0x9A0 /* status = EMBEBBED_STAT_XXX */ +#define MIXART_PSEUDOREG_DXLX_BASE_ADDR_OFFSET MIXART_PSEUDOREG+0x998 /* get the address here where to write the file */ +#define MIXART_PSEUDOREG_DXLX_SIZE_OFFSET MIXART_PSEUDOREG+0x99C /* xilinx size in bytes */ +#define MIXART_PSEUDOREG_DXLX_STATUS_OFFSET MIXART_PSEUDOREG+0x9A0 /* status = EMBEBBED_STAT_XXX */ /* */ #define MIXART_FLOWTABLE_PTR 0x3000 /* pointer to flow table */ diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 3ba6174..388c257 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -310,25 +310,25 @@ static int mixart_update_analog_audio_level(struct snd_mixart* chip, int is_capt memset(&io_level, 0, sizeof(io_level)); io_level.channel = -1; /* left and right */ - for(i=0; i<2; i++) { - if(is_capture) { + for (i = 0; i < 2; i++) { + if (is_capture) { io_level.level[i].analog_level = mixart_analog_level[chip->analog_capture_volume[i]]; } else { - if(chip->analog_playback_active[i]) + if (chip->analog_playback_active[i]) io_level.level[i].analog_level = mixart_analog_level[chip->analog_playback_volume[i]]; else io_level.level[i].analog_level = mixart_analog_level[MIXART_ANALOG_PLAYBACK_LEVEL_MIN]; } } - if(is_capture) request.uid = chip->uid_in_analog_physio; + if (is_capture) request.uid = chip->uid_in_analog_physio; else request.uid = chip->uid_out_analog_physio; request.message_id = MSG_PHYSICALIO_SET_LEVEL; request.data = &io_level; request.size = sizeof(io_level); err = snd_mixart_send_msg(chip->mgr, &request, sizeof(resp), &resp); - if((err<0) || (resp.error_code)) { + if ((err<0) || (resp.error_code)) { snd_printk(KERN_DEBUG "error MSG_PHYSICALIO_SET_LEVEL card(%d) is_capture(%d) error_code(%x)\n", chip->chip_idx, is_capture, resp.error_code); return -EINVAL; } @@ -342,7 +342,7 @@ static int mixart_analog_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; - if(kcontrol->private_value == 0) { /* playback */ + if (kcontrol->private_value == 0) { /* playback */ uinfo->value.integer.min = MIXART_ANALOG_PLAYBACK_LEVEL_MIN; /* -96 dB */ uinfo->value.integer.max = MIXART_ANALOG_PLAYBACK_LEVEL_MAX; /* 0 dB */ } else { /* capture */ @@ -356,7 +356,7 @@ static int mixart_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e { struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); mutex_lock(&chip->mgr->mixer_mutex); - if(kcontrol->private_value == 0) { /* playback */ + if (kcontrol->private_value == 0) { /* playback */ ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; } else { /* capture */ @@ -731,7 +731,7 @@ int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int set_level.nb_of_stream = 1; set_level.stream_level.desc.stream_idx = idx; - if(is_aes) { + if (is_aes) { pipe = &chip->pipe_out_dig; /* AES playback */ idx += MIXART_PLAYBACK_STREAMS; } else { @@ -739,13 +739,13 @@ int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int } /* only when pipe exists ! */ - if(pipe->status == PIPE_UNDEFINED) + if (pipe->status == PIPE_UNDEFINED) return 0; set_level.stream_level.desc.uid_pipe = pipe->group_uid; - for(i=0; i<2; i++) { - if(chip->digital_playback_active[idx][i]) + for (i = 0; i < 2; i++) { + if (chip->digital_playback_active[idx][i]) volume[i] = chip->digital_playback_volume[idx][i]; else volume[i] = MIXART_DIGITAL_LEVEL_MIN; @@ -761,7 +761,7 @@ int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int request.size = sizeof(set_level); err = snd_mixart_send_msg(chip->mgr, &request, sizeof(status), &status); - if((err<0) || status) { + if ((err<0) || status) { snd_printk(KERN_DEBUG "error MSG_STREAM_SET_OUT_STREAM_LEVEL card(%d) status(%x)\n", chip->chip_idx, status); return -EINVAL; } @@ -776,7 +776,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) struct mixart_set_in_audio_level_req set_level; u32 status; - if(is_aes) { + if (is_aes) { idx = 1; pipe = &chip->pipe_in_dig; } else { @@ -785,7 +785,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) } /* only when pipe exists ! */ - if(pipe->status == PIPE_UNDEFINED) + if (pipe->status == PIPE_UNDEFINED) return 0; memset(&set_level, 0, sizeof(set_level)); @@ -793,7 +793,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) set_level.level[0].connector = pipe->uid_left_connector; set_level.level[1].connector = pipe->uid_right_connector; - for(i=0; i<2; i++) { + for (i = 0; i < 2; i++) { set_level.level[i].valid_mask1 = MIXART_AUDIO_LEVEL_DIGITAL_MASK; set_level.level[i].digital_level = mixart_digital_level[chip->digital_capture_volume[idx][i]]; } @@ -804,7 +804,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) request.size = sizeof(set_level); err = snd_mixart_send_msg(chip->mgr, &request, sizeof(status), &status); - if((err<0) || status) { + if ((err<0) || status) { snd_printk(KERN_DEBUG "error MSG_STREAM_SET_IN_AUDIO_LEVEL card(%d) status(%x)\n", chip->chip_idx, status); return -EINVAL; } @@ -833,12 +833,12 @@ static int mixart_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem int is_capture = kcontrol->private_value & MIXART_VOL_REC_MASK; int is_aes = kcontrol->private_value & MIXART_VOL_AES_MASK; mutex_lock(&chip->mgr->mixer_mutex); - if(is_capture) { - if(is_aes) stored_volume = chip->digital_capture_volume[1]; /* AES capture */ + if (is_capture) { + if (is_aes) stored_volume = chip->digital_capture_volume[1]; /* AES capture */ else stored_volume = chip->digital_capture_volume[0]; /* analog capture */ } else { snd_BUG_ON(idx >= MIXART_PLAYBACK_STREAMS); - if(is_aes) stored_volume = chip->digital_playback_volume[MIXART_PLAYBACK_STREAMS + idx]; /* AES playback */ + if (is_aes) stored_volume = chip->digital_playback_volume[MIXART_PLAYBACK_STREAMS + idx]; /* AES playback */ else stored_volume = chip->digital_playback_volume[idx]; /* analog playback */ } ucontrol->value.integer.value[0] = stored_volume[0]; @@ -911,7 +911,7 @@ static int mixart_pcm_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ snd_BUG_ON(idx >= MIXART_PLAYBACK_STREAMS); mutex_lock(&chip->mgr->mixer_mutex); - if(kcontrol->private_value & MIXART_VOL_AES_MASK) /* AES playback */ + if (kcontrol->private_value & MIXART_VOL_AES_MASK) /* AES playback */ idx += MIXART_PLAYBACK_STREAMS; ucontrol->value.integer.value[0] = chip->digital_playback_active[idx][0]; ucontrol->value.integer.value[1] = chip->digital_playback_active[idx][1]; @@ -961,10 +961,10 @@ static int mixart_update_monitoring(struct snd_mixart* chip, int channel) struct mixart_set_out_audio_level audio_level; u32 resp; - if(chip->pipe_out_ana.status == PIPE_UNDEFINED) + if (chip->pipe_out_ana.status == PIPE_UNDEFINED) return -EINVAL; /* no pipe defined */ - if(!channel) request.uid = chip->pipe_out_ana.uid_left_connector; + if (!channel) request.uid = chip->pipe_out_ana.uid_left_connector; else request.uid = chip->pipe_out_ana.uid_right_connector; request.message_id = MSG_CONNECTOR_SET_OUT_AUDIO_LEVEL; request.data = &audio_level; @@ -976,7 +976,7 @@ static int mixart_update_monitoring(struct snd_mixart* chip, int channel) audio_level.monitor_mute1 = !chip->monitoring_active[channel!=0]; err = snd_mixart_send_msg(chip->mgr, &request, sizeof(resp), &resp); - if((err<0) || resp) { + if ((err<0) || resp) { snd_printk(KERN_DEBUG "error MSG_CONNECTOR_SET_OUT_AUDIO_LEVEL card(%d) resp(%x)\n", chip->chip_idx, resp); return -EINVAL; } @@ -1097,7 +1097,7 @@ static void mixart_reset_audio_levels(struct snd_mixart *chip) /* analog volumes can be set even if there is no pipe */ mixart_update_analog_audio_level(chip, 0); /* analog levels for capture only on the first two chips */ - if(chip->chip_idx < 2) { + if (chip->chip_idx < 2) { mixart_update_analog_audio_level(chip, 1); } return; @@ -1111,7 +1111,7 @@ int snd_mixart_create_mixer(struct mixart_mgr *mgr) mutex_init(&mgr->mixer_mutex); /* can be in another place */ - for(i=0; i<mgr->num_cards; i++) { + for (i = 0; i < mgr->num_cards; i++) { struct snd_kcontrol_new temp; chip = mgr->chip[i]; @@ -1126,7 +1126,7 @@ int snd_mixart_create_mixer(struct mixart_mgr *mgr) return err; /* analog input level control only on first two chips !*/ - if(i<2) { + if (i<2) { temp = mixart_control_analog_level; temp.name = "Master Capture Volume"; temp.private_value = 1; /* capture */ @@ -1147,7 +1147,7 @@ int snd_mixart_create_mixer(struct mixart_mgr *mgr) if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) return err; - if(mgr->board_type == MIXART_DAUGHTER_TYPE_AES) { + if (mgr->board_type == MIXART_DAUGHTER_TYPE_AES) { temp.name = "AES Playback Volume"; temp.count = MIXART_PLAYBACK_STREAMS; temp.private_value = MIXART_VOL_AES_MASK; /* playback AES/EBU */ @@ -1166,7 +1166,7 @@ int snd_mixart_create_mixer(struct mixart_mgr *mgr) if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) return err; - if(mgr->board_type == MIXART_DAUGHTER_TYPE_AES) { + if (mgr->board_type == MIXART_DAUGHTER_TYPE_AES) { temp.name = "AES Playback Switch"; temp.private_value = MIXART_VOL_AES_MASK; /* playback AES/EBU */ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)

1 0

[alsa-devel] [PATCH 14/24] sound/pci: coding style fixes: maestro3
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/maestro3.c | 130 +++++++++++++++++++++++++------------------------- 1 files changed, 65 insertions(+), 65 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 9ff3f9e..0aae83b 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -27,11 +27,11 @@ * - Added Canyon3D-2 support by Rob Riggs <rob(a)pangalactic.org> * */ - + #define CARD_NAME "ESS Maestro3/Allegro/Canyon3D-2" #define DRIVER_NAME "Maestro3" -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -569,7 +569,7 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define KDATA_ADC2_LEFT_SUR_VOL (KDATA_BASE_ADDR + 0x0045) #define KDATA_ADC2_RIGHT_SUR_VOL (KDATA_BASE_ADDR + 0x0046) -#define KDATA_CD_XFER0 (KDATA_BASE_ADDR + 0x0047) +#define KDATA_CD_XFER0 (KDATA_BASE_ADDR + 0x0047) #define KDATA_CD_XFER_ENDMARK (KDATA_BASE_ADDR + 0x0048) #define KDATA_CD_LEFT_VOLUME (KDATA_BASE_ADDR + 0x0049) #define KDATA_CD_RIGHT_VOLUME (KDATA_BASE_ADDR + 0x004A) @@ -727,9 +727,9 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define SRC3_TEMP_INBUF_ADDR_OFFSET (CDATA_HEADER_LEN + 16) #define SRC3_TEMP_OUTBUF_ADDR_OFFSET (CDATA_HEADER_LEN + 17) -#define MINISRC_IN_BUFFER_SIZE ( 0x50 * 2 ) +#define MINISRC_IN_BUFFER_SIZE ( 0x50 * 2) #define MINISRC_OUT_BUFFER_SIZE ( 0x50 * 2 * 2) -#define MINISRC_TMP_BUFFER_SIZE ( 112 + ( MINISRC_BIQUAD_STAGE * 3 + 4 ) * 2 * 2 ) +#define MINISRC_TMP_BUFFER_SIZE ( 112 + ( MINISRC_BIQUAD_STAGE * 3 + 4 ) * 2 * 2) #define MINISRC_BIQUAD_STAGE 2 #define MINISRC_COEF_LOC 0x175 @@ -762,10 +762,10 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); /* * an arbitrary volume we set the internal * volume settings to so that the ac97 volume - * range is a little less insane. 0x7fff is + * range is a little less insane. 0x7fff is * max. */ -#define ARB_VOLUME ( 0x6800 ) +#define ARB_VOLUME ( 0x6800) /* */ @@ -798,13 +798,13 @@ struct m3_dma { struct m3_list *index_list[3]; int in_lists; - + struct list_head list; }; - + struct snd_m3 { - + struct snd_card *card; unsigned long iobase; @@ -1077,7 +1077,7 @@ static void snd_m3_remove_list(struct snd_m3 *chip, struct m3_list *list, int in static void snd_m3_inc_timer_users(struct snd_m3 *chip) { chip->timer_users++; - if (chip->timer_users != 1) + if (chip->timer_users != 1) return; snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, @@ -1096,7 +1096,7 @@ static void snd_m3_inc_timer_users(struct snd_m3 *chip) static void snd_m3_dec_timer_users(struct snd_m3 *chip) { chip->timer_users--; - if (chip->timer_users > 0) + if (chip->timer_users > 0) return; snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, @@ -1120,7 +1120,7 @@ static void snd_m3_dec_timer_users(struct snd_m3 *chip) static int snd_m3_pcm_start(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substream *subs) { - if (! s || ! subs) + if (!s || !subs) return -EINVAL; snd_m3_inc_timer_users(chip); @@ -1147,7 +1147,7 @@ static int snd_m3_pcm_start(struct snd_m3 *chip, struct m3_dma *s, static int snd_m3_pcm_stop(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substream *subs) { - if (! s || ! subs) + if (!s || !subs) return -EINVAL; snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, @@ -1157,7 +1157,7 @@ static int snd_m3_pcm_stop(struct snd_m3 *chip, struct m3_dma *s, case SNDRV_PCM_STREAM_PLAYBACK: chip->dacs_active--; snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, - KDATA_MIXER_TASK_NUMBER, + KDATA_MIXER_TASK_NUMBER, chip->dacs_active); break; case SNDRV_PCM_STREAM_CAPTURE: @@ -1191,7 +1191,7 @@ snd_m3_pcm_trigger(struct snd_pcm_substream *subs, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - if (! s->running) + if (!s->running) err = 0; /* should return error? */ else { s->running = 0; @@ -1206,7 +1206,7 @@ snd_m3_pcm_trigger(struct snd_pcm_substream *subs, int cmd) /* * setup */ -static void +static void snd_m3_pcm_setup1(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substream *subs) { int dsp_in_size, dsp_out_size, dsp_in_buffer, dsp_out_buffer; @@ -1270,7 +1270,7 @@ snd_m3_pcm_setup1(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substrea snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + CDATA_IN_BUF_HEAD, dsp_in_buffer); - + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + CDATA_IN_BUF_TAIL, dsp_in_buffer); @@ -1297,10 +1297,10 @@ static void snd_m3_pcm_setup2(struct snd_m3 *chip, struct m3_dma *s, { u32 freq; - /* + /* * put us in the lists if we're not already there */ - if (! s->in_lists) { + if (!s->in_lists) { s->index[0] = snd_m3_add_list(chip, s->index_list[0], s->inst.data >> DP_SHIFT_COUNT); s->index[1] = snd_m3_add_list(chip, s->index_list[1], @@ -1312,16 +1312,16 @@ static void snd_m3_pcm_setup2(struct snd_m3 *chip, struct m3_dma *s, /* write to 'mono' word */ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, - s->inst.data + SRC3_DIRECTION_OFFSET + 1, + s->inst.data + SRC3_DIRECTION_OFFSET + 1, runtime->channels == 2 ? 0 : 1); /* write to '8bit' word */ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, - s->inst.data + SRC3_DIRECTION_OFFSET + 2, + s->inst.data + SRC3_DIRECTION_OFFSET + 2, snd_pcm_format_width(runtime->format) == 16 ? 0 : 1); /* set up dac/adc rate */ freq = ((runtime->rate << 15) + 24000 ) / 48000; - if (freq) + if (freq) freq--; snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, @@ -1380,7 +1380,7 @@ snd_m3_playback_setup(struct snd_m3 *chip, struct m3_dma *s, snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + SRC3_DIRECTION_OFFSET + 22, subs->runtime->rate > 45000 ? 0xff : 0); - + /* tell it which way dma is going? */ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + CDATA_DMA_CONTROL, @@ -1389,13 +1389,13 @@ snd_m3_playback_setup(struct snd_m3 *chip, struct m3_dma *s, /* * set an armload of static initializers */ - for (i = 0; i < ARRAY_SIZE(pv); i++) + for (i = 0; i < ARRAY_SIZE(pv); i++) snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + pv[i].addr, pv[i].val); } /* - * Native record driver + * Native record driver */ static const struct rec_vals { u16 addr, val; @@ -1441,13 +1441,13 @@ snd_m3_capture_setup(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_subst /* tell it which way dma is going? */ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + CDATA_DMA_CONTROL, - DMACONTROL_DIRECTION + DMACONTROL_AUTOREPEAT + + DMACONTROL_DIRECTION + DMACONTROL_AUTOREPEAT + DMAC_PAGE3_SELECTOR + DMAC_BLOCKF_SELECTOR); /* * set an armload of static initializers */ - for (i = 0; i < ARRAY_SIZE(rv); i++) + for (i = 0; i < ARRAY_SIZE(rv); i++) snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, s->inst.data + rv[i].addr, rv[i].val); } @@ -1472,7 +1472,7 @@ static int snd_m3_pcm_hw_params(struct snd_pcm_substream *substream, static int snd_m3_pcm_hw_free(struct snd_pcm_substream *substream) { struct m3_dma *s; - + if (substream->runtime->private_data == NULL) return 0; s = substream->runtime->private_data; @@ -1567,7 +1567,7 @@ static void snd_m3_update_ptr(struct snd_m3 *chip, struct m3_dma *s) unsigned int hwptr; int diff; - if (! s->running) + if (!s->running) return; hwptr = snd_m3_get_pointer(chip, s, subs); @@ -1764,7 +1764,7 @@ snd_m3_substream_open(struct snd_m3 *chip, struct snd_pcm_substream *subs) spin_lock_irq(&chip->reg_lock); for (i = 0; i < chip->num_substreams; i++) { s = &chip->substreams[i]; - if (! s->opened) + if (!s->opened) goto __found; } spin_unlock_irq(&chip->reg_lock); @@ -1885,7 +1885,7 @@ static struct snd_pcm_ops snd_m3_capture_ops = { }; static int __devinit -snd_m3_pcm(struct snd_m3 * chip, int device) +snd_m3_pcm(struct snd_m3 *chip, int device) { struct snd_pcm *pcm; int err; @@ -1902,7 +1902,7 @@ snd_m3_pcm(struct snd_m3 * chip, int device) pcm->info_flags = 0; strcpy(pcm->name, chip->card->driver); chip->pcm = pcm; - + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 64*1024, 64*1024); @@ -1923,7 +1923,7 @@ static int snd_m3_ac97_wait(struct snd_m3 *chip) int i = 10000; do { - if (! (snd_m3_inb(chip, 0x30) & 1)) + if (!(snd_m3_inb(chip, 0x30) & 1)) return 0; cpu_relax(); } while (i-- > 0); @@ -1979,8 +1979,8 @@ static void snd_m3_remote_codec_config(int io, int isremote) io + SDO_IN_DEST_CTRL); } -/* - * hack, returns non zero on err +/* + * hack, returns non zero on err */ static int snd_m3_try_read_vendor(struct snd_m3 *chip) { @@ -2007,9 +2007,9 @@ static void snd_m3_ac97_reset(struct snd_m3 *chip) if (chip->allegro_flag) { /* - * the onboard codec on the allegro seems + * the onboard codec on the allegro seems * to want to wait a very long time before - * coming back to life + * coming back to life */ delay1 = 50; delay2 = 800; @@ -2044,7 +2044,7 @@ static void snd_m3_ac97_reset(struct snd_m3 *chip) schedule_timeout_uninterruptible(msecs_to_jiffies(delay2)); - if (! snd_m3_try_read_vendor(chip)) + if (!snd_m3_try_read_vendor(chip)) break; delay1 += 10; @@ -2079,7 +2079,7 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) return err; - + memset(&ac97, 0, sizeof(ac97)); ac97.private_data = chip; if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97)) < 0) @@ -2120,7 +2120,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip) /* zero kernel data */ for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++) - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, KDATA_BASE_ADDR + i, 0); /* zero mixer data? */ @@ -2135,8 +2135,8 @@ static void snd_m3_assp_init(struct snd_m3 *chip) /* write kernel into code memory.. */ data = (const u16 *)chip->assp_kernel_image->data; - for (i = 0 ; i * 2 < chip->assp_kernel_image->size; i++) { - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, + for (i = 0; i * 2 < chip->assp_kernel_image->size; i++) { + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, REV_B_CODE_MEMORY_BEGIN + i, le16_to_cpu(data[i])); } @@ -2149,14 +2149,14 @@ static void snd_m3_assp_init(struct snd_m3 *chip) */ data = (const u16 *)chip->assp_minisrc_image->data; for (i = 0; i * 2 < chip->assp_minisrc_image->size; i++) { - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, 0x400 + i, le16_to_cpu(data[i])); } /* * write the coefficients for the low pass filter? */ - for (i = 0; i < MINISRC_LPF_LEN ; i++) { + for (i = 0; i < MINISRC_LPF_LEN; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, 0x400 + MINISRC_COEF_LOC + i, minisrc_lpf[i]); @@ -2170,7 +2170,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip) * the minisrc is the only thing on * our task list.. */ - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, KDATA_TASK0, 0x400); @@ -2206,14 +2206,14 @@ static void snd_m3_assp_init(struct snd_m3 *chip) static int __devinit snd_m3_assp_client_init(struct snd_m3 *chip, struct m3_dma *s, int index) { - int data_bytes = 2 * ( MINISRC_TMP_BUFFER_SIZE / 2 + + int data_bytes = 2 * ( MINISRC_TMP_BUFFER_SIZE / 2 + MINISRC_IN_BUFFER_SIZE / 2 + - 1 + MINISRC_OUT_BUFFER_SIZE / 2 + 1 ); + 1 + MINISRC_OUT_BUFFER_SIZE / 2 + 1); int address, i; /* * the revb memory map has 0x1100 through 0x1c00 - * free. + * free. */ /* @@ -2243,7 +2243,7 @@ static int __devinit snd_m3_assp_client_init(struct snd_m3 *chip, struct m3_dma } -/* +/* * this works for the reference board, have to find * out about others * @@ -2255,7 +2255,7 @@ snd_m3_amp_enable(struct snd_m3 *chip, int enable) int io = chip->iobase; u16 gpo, polarity; - if (! chip->external_amp) + if (!chip->external_amp) return; polarity = enable ? 0 : 1; @@ -2323,10 +2323,10 @@ snd_m3_chip_init(struct snd_m3 *chip) pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~INT_CLK_SELECT; if (!chip->allegro_flag) { - n &= ~INT_CLK_MULT_ENABLE; + n &= ~INT_CLK_MULT_ENABLE; n |= INT_CLK_SRC_NOT_PCI; } - n &= ~( CLK_MULT_MODE_SELECT | CLK_MULT_MODE_SELECT_2 ); + n &= ~( CLK_MULT_MODE_SELECT | CLK_MULT_MODE_SELECT_2); pci_write_config_dword(pcidev, PCI_ALLEGRO_CONFIG, n); if (chip->allegro_flag) { @@ -2338,11 +2338,11 @@ snd_m3_chip_init(struct snd_m3 *chip) t = inb(chip->iobase + ASSP_CONTROL_A); t &= ~( DSP_CLK_36MHZ_SELECT | ASSP_CLK_49MHZ_SELECT); t |= ASSP_CLK_49MHZ_SELECT; - t |= ASSP_0_WS_ENABLE; + t |= ASSP_0_WS_ENABLE; outb(t, chip->iobase + ASSP_CONTROL_A); snd_m3_assp_init(chip); /* download DSP code before starting ASSP below */ - outb(RUN_ASSP, chip->iobase + ASSP_CONTROL_B); + outb(RUN_ASSP, chip->iobase + ASSP_CONTROL_B); outb(0x00, io + HARDWARE_VOL_CTRL); outb(0x88, io + SHADOW_MIX_REG_VOICE); @@ -2351,7 +2351,7 @@ snd_m3_chip_init(struct snd_m3 *chip) outb(0x88, io + HW_VOL_COUNTER_MASTER); return 0; -} +} static void snd_m3_enable_ints(struct snd_m3 *chip) @@ -2437,7 +2437,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++) chip->suspend_mem[dsp_index++] = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_CODE, i); - for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++) + for (i = REV_B_DATA_MEMORY_BEGIN; i <= REV_B_DATA_MEMORY_END; i++) chip->suspend_mem[dsp_index++] = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA, i); @@ -2477,14 +2477,14 @@ static int m3_resume(struct pci_dev *pci) /* restore dsp image */ dsp_index = 0; for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++) - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, i, + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, i, chip->suspend_mem[dsp_index++]); - for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++) - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, i, + for (i = REV_B_DATA_MEMORY_BEGIN; i <= REV_B_DATA_MEMORY_END; i++) + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, i, chip->suspend_mem[dsp_index++]); /* tell the dma engine to restart itself */ - snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, + snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, KDATA_DMA_ACTIVE, 0); /* restore ac97 registers */ @@ -2614,7 +2614,7 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, return err; } chip->iobase = pci_resource_start(pci, 0); - + /* just to be sure */ pci_set_master(pci); @@ -2659,7 +2659,7 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if ((err = snd_m3_pcm(chip, 0)) < 0) return err; - + snd_m3_enable_ints(chip); snd_m3_assp_continue(chip); @@ -2667,7 +2667,7 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, *chip_ret = chip; - return 0; + return 0; } /* @@ -2758,7 +2758,7 @@ static struct pci_driver driver = { .resume = m3_resume, #endif }; - + static int __init alsa_card_m3_init(void) { return pci_register_driver(&driver);

1 0

[alsa-devel] [PATCH 13/24] sound/pci: coding style fixes: korg1212
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/korg1212/korg1212.c | 492 ++++++++++++++++++++-------------------- 1 files changed, 246 insertions(+), 246 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 5f8006b..d4c7dd7 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -36,7 +36,7 @@ #include <sound/pcm_params.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> // ---------------------------------------------------------------------------- // Debug Stuff @@ -54,7 +54,7 @@ #endif // ---------------------------------------------------------------------------- -// Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all +// Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all // buffers are alocated as a large piece inside KorgSharedBuffer. // ---------------------------------------------------------------------------- //#define K1212_LARGEALLOC 1 @@ -63,21 +63,21 @@ // Valid states of the Korg 1212 I/O card. // ---------------------------------------------------------------------------- enum CardState { - K1212_STATE_NONEXISTENT, // there is no card here - K1212_STATE_UNINITIALIZED, // the card is awaiting DSP download - K1212_STATE_DSP_IN_PROCESS, // the card is currently downloading its DSP code - K1212_STATE_DSP_COMPLETE, // the card has finished the DSP download - K1212_STATE_READY, // the card can be opened by an application. Any application - // requests prior to this state should fail. Only an open - // request can be made at this state. - K1212_STATE_OPEN, // an application has opened the card - K1212_STATE_SETUP, // the card has been setup for play - K1212_STATE_PLAYING, // the card is playing - K1212_STATE_MONITOR, // the card is in the monitor mode - K1212_STATE_CALIBRATING, // the card is currently calibrating - K1212_STATE_ERRORSTOP, // the card has stopped itself because of an error and we - // are in the process of cleaning things up. - K1212_STATE_MAX_STATE // state values of this and beyond are invalid + K1212_STATE_NONEXISTENT, /* there is no card here */ + K1212_STATE_UNINITIALIZED, /* the card is awaiting DSP download */ + K1212_STATE_DSP_IN_PROCESS, /* the card is currently downloading its DSP code */ + K1212_STATE_DSP_COMPLETE, /* the card has finished the DSP download */ + K1212_STATE_READY, /* the card can be opened by an application. Any application */ + /* requests prior to this state should fail. Only an open */ + /* request can be made at this state. */ + K1212_STATE_OPEN, /* an application has opened the card */ + K1212_STATE_SETUP, /* the card has been setup for play */ + K1212_STATE_PLAYING, /* the card is playing */ + K1212_STATE_MONITOR, /* the card is in the monitor mode */ + K1212_STATE_CALIBRATING, /* the card is currently calibrating */ + K1212_STATE_ERRORSTOP, /* the card has stopped itself because of an error and we */ + /* are in the process of cleaning things up. */ + K1212_STATE_MAX_STATE /* state values of this and beyond are invalid */ }; // ---------------------------------------------------------------------------- @@ -85,48 +85,48 @@ enum CardState { // host-to-card doorbell to initiate a command. // ---------------------------------------------------------------------------- enum korg1212_dbcnst { - K1212_DB_RequestForData = 0, // sent by the card to request a buffer fill. - K1212_DB_TriggerPlay = 1, // starts playback/record on the card. - K1212_DB_SelectPlayMode = 2, // select monitor, playback setup, or stop. - K1212_DB_ConfigureBufferMemory = 3, // tells card where the host audio buffers are. - K1212_DB_RequestAdatTimecode = 4, // asks the card for the latest ADAT timecode value. - K1212_DB_SetClockSourceRate = 5, // sets the clock source and rate for the card. - K1212_DB_ConfigureMiscMemory = 6, // tells card where other buffers are. - K1212_DB_TriggerFromAdat = 7, // tells card to trigger from Adat at a specific - // timecode value. - K1212_DB_DMAERROR = 0x80, // DMA Error - the PCI bus is congestioned. - K1212_DB_CARDSTOPPED = 0x81, // Card has stopped by user request. - K1212_DB_RebootCard = 0xA0, // instructs the card to reboot. - K1212_DB_BootFromDSPPage4 = 0xA4, // instructs the card to boot from the DSP microcode - // on page 4 (local page to card). - K1212_DB_DSPDownloadDone = 0xAE, // sent by the card to indicate the download has - // completed. - K1212_DB_StartDSPDownload = 0xAF // tells the card to download its DSP firmware. + K1212_DB_RequestForData = 0, /* sent by the card to request a buffer fill. */ + K1212_DB_TriggerPlay = 1, /* starts playback/record on the card. */ + K1212_DB_SelectPlayMode = 2, /* select monitor, playback setup, or stop. */ + K1212_DB_ConfigureBufferMemory = 3, /* tells card where the host audio buffers are. */ + K1212_DB_RequestAdatTimecode = 4, /* asks the card for the latest ADAT timecode value. */ + K1212_DB_SetClockSourceRate = 5, /* sets the clock source and rate for the card. */ + K1212_DB_ConfigureMiscMemory = 6, /* tells card where other buffers are. */ + K1212_DB_TriggerFromAdat = 7, /* tells card to trigger from Adat at a specific */ + /* timecode value. */ + K1212_DB_DMAERROR = 0x80, /* DMA Error - the PCI bus is congestioned. */ + K1212_DB_CARDSTOPPED = 0x81, /* Card has stopped by user request. */ + K1212_DB_RebootCard = 0xA0, /* instructs the card to reboot. */ + K1212_DB_BootFromDSPPage4 = 0xA4, /* instructs the card to boot from the DSP microcode */ + /* on page 4 (local page to card). */ + K1212_DB_DSPDownloadDone = 0xAE, /* sent by the card to indicate the download has */ + /* completed. */ + K1212_DB_StartDSPDownload = 0xAF /* tells the card to download its DSP firmware. */ }; // ---------------------------------------------------------------------------- -// The following enumeration defines return codes +// The following enumeration defines return codes // to the Korg 1212 I/O driver. // ---------------------------------------------------------------------------- enum snd_korg1212rc { - K1212_CMDRET_Success = 0, // command was successfully placed - K1212_CMDRET_DIOCFailure, // the DeviceIoControl call failed - K1212_CMDRET_PMFailure, // the protected mode call failed - K1212_CMDRET_FailUnspecified, // unspecified failure - K1212_CMDRET_FailBadState, // the specified command can not be given in - // the card's current state. (or the wave device's - // state) - K1212_CMDRET_CardUninitialized, // the card is uninitialized and cannot be used - K1212_CMDRET_BadIndex, // an out of range card index was specified - K1212_CMDRET_BadHandle, // an invalid card handle was specified - K1212_CMDRET_NoFillRoutine, // a play request has been made before a fill routine set - K1212_CMDRET_FillRoutineInUse, // can't set a new fill routine while one is in use - K1212_CMDRET_NoAckFromCard, // the card never acknowledged a command - K1212_CMDRET_BadParams, // bad parameters were provided by the caller - - K1212_CMDRET_BadDevice, // the specified wave device was out of range - K1212_CMDRET_BadFormat // the specified wave format is unsupported + K1212_CMDRET_Success = 0, /* command was successfully placed */ + K1212_CMDRET_DIOCFailure, /* the DeviceIoControl call failed */ + K1212_CMDRET_PMFailure, /* the protected mode call failed */ + K1212_CMDRET_FailUnspecified, /* unspecified failure */ + K1212_CMDRET_FailBadState, /* the specified command can not be given in */ + /* the card's current state. (or the wave device's */ + /* state) */ + K1212_CMDRET_CardUninitialized, /* the card is uninitialized and cannot be used */ + K1212_CMDRET_BadIndex, /* an out of range card index was specified */ + K1212_CMDRET_BadHandle, /* an invalid card handle was specified */ + K1212_CMDRET_NoFillRoutine, /* a play request has been made before a fill routine set */ + K1212_CMDRET_FillRoutineInUse, /* can't set a new fill routine while one is in use */ + K1212_CMDRET_NoAckFromCard, /* the card never acknowledged a command */ + K1212_CMDRET_BadParams, /* bad parameters were provided by the caller */ + + K1212_CMDRET_BadDevice, /* the specified wave device was out of range */ + K1212_CMDRET_BadFormat /* the specified wave format is unsupported */ }; // ---------------------------------------------------------------------------- @@ -134,10 +134,10 @@ enum snd_korg1212rc { // mode for the card in the SelectPlayMode command. // ---------------------------------------------------------------------------- enum PlayModeSelector { - K1212_MODE_SetupPlay = 0x00000001, // provides card with pre-play information - K1212_MODE_MonitorOn = 0x00000002, // tells card to turn on monitor mode - K1212_MODE_MonitorOff = 0x00000004, // tells card to turn off monitor mode - K1212_MODE_StopPlay = 0x00000008 // stops playback on the card + K1212_MODE_SetupPlay = 0x00000001, /* provides card with pre-play information */ + K1212_MODE_MonitorOn = 0x00000002, /* tells card to turn on monitor mode */ + K1212_MODE_MonitorOff = 0x00000004, /* tells card to turn off monitor mode */ + K1212_MODE_StopPlay = 0x00000008 /* stops playback on the card */ }; // ---------------------------------------------------------------------------- @@ -145,28 +145,28 @@ enum PlayModeSelector { // mode for the card in the SetMonitorMode command. // ---------------------------------------------------------------------------- enum MonitorModeSelector { - K1212_MONMODE_Off = 0, // tells card to turn off monitor mode - K1212_MONMODE_On // tells card to turn on monitor mode + K1212_MONMODE_Off = 0, /* tells card to turn off monitor mode */ + K1212_MONMODE_On /* tells card to turn on monitor mode */ }; -#define MAILBOX0_OFFSET 0x40 // location of mailbox 0 relative to base address -#define MAILBOX1_OFFSET 0x44 // location of mailbox 1 relative to base address -#define MAILBOX2_OFFSET 0x48 // location of mailbox 2 relative to base address -#define MAILBOX3_OFFSET 0x4c // location of mailbox 3 relative to base address -#define OUT_DOORBELL_OFFSET 0x60 // location of PCI to local doorbell -#define IN_DOORBELL_OFFSET 0x64 // location of local to PCI doorbell -#define STATUS_REG_OFFSET 0x68 // location of interrupt control/status register -#define PCI_CONTROL_OFFSET 0x6c // location of the EEPROM, PCI, User I/O, init control - // register -#define SENS_CONTROL_OFFSET 0x6e // location of the input sensitivity setting register. - // this is the upper word of the PCI control reg. -#define DEV_VEND_ID_OFFSET 0x70 // location of the device and vendor ID register - -#define MAX_COMMAND_RETRIES 5 // maximum number of times the driver will attempt - // to send a command before giving up. -#define COMMAND_ACK_MASK 0x8000 // the MSB is set in the command acknowledgment from - // the card. -#define DOORBELL_VAL_MASK 0x00FF // the doorbell value is one byte +#define MAILBOX0_OFFSET 0x40 /* location of mailbox 0 relative to base address */ +#define MAILBOX1_OFFSET 0x44 /* location of mailbox 1 relative to base address */ +#define MAILBOX2_OFFSET 0x48 /* location of mailbox 2 relative to base address */ +#define MAILBOX3_OFFSET 0x4c /* location of mailbox 3 relative to base address */ +#define OUT_DOORBELL_OFFSET 0x60 /* location of PCI to local doorbell */ +#define IN_DOORBELL_OFFSET 0x64 /* location of local to PCI doorbell */ +#define STATUS_REG_OFFSET 0x68 /* location of interrupt control/status register */ +#define PCI_CONTROL_OFFSET 0x6c /* location of the EEPROM, PCI, User I/O, init control */ + /* register */ +#define SENS_CONTROL_OFFSET 0x6e /* location of the input sensitivity setting register. */ + /* this is the upper word of the PCI control reg. */ +#define DEV_VEND_ID_OFFSET 0x70 /* location of the device and vendor ID register */ + +#define MAX_COMMAND_RETRIES 5 /* maximum number of times the driver will attempt */ + /* to send a command before giving up. */ +#define COMMAND_ACK_MASK 0x8000 /* the MSB is set in the command acknowledgment from */ + /* the card. */ +#define DOORBELL_VAL_MASK 0x00FF /* the doorbell value is one byte */ #define CARD_BOOT_DELAY_IN_MS 10 #define CARD_BOOT_TIMEOUT 10 @@ -255,25 +255,25 @@ enum MonitorModeSelector { #define SENSCLKPULSE_WIDTH 4 #define LOADSHIFT_DELAY 4 #define INTERCOMMAND_DELAY 40 -#define STOPCARD_DELAY 300 // max # RTC ticks for the card to stop once we write - // the command register. (could be up to 180 us) -#define COMMAND_ACK_DELAY 13 // number of RTC ticks to wait for an acknowledgement - // from the card after sending a command. +#define STOPCARD_DELAY 300 /* max # RTC ticks for the card to stop once we write */ + /* the command register. (could be up to 180 us) */ +#define COMMAND_ACK_DELAY 13 /* number of RTC ticks to wait for an acknowledgement */ + /* from the card after sending a command. */ enum ClockSourceIndex { - K1212_CLKIDX_AdatAt44_1K = 0, // selects source as ADAT at 44.1 kHz - K1212_CLKIDX_AdatAt48K, // selects source as ADAT at 48 kHz - K1212_CLKIDX_WordAt44_1K, // selects source as S/PDIF at 44.1 kHz - K1212_CLKIDX_WordAt48K, // selects source as S/PDIF at 48 kHz - K1212_CLKIDX_LocalAt44_1K, // selects source as local clock at 44.1 kHz - K1212_CLKIDX_LocalAt48K, // selects source as local clock at 48 kHz - K1212_CLKIDX_Invalid // used to check validity of the index + K1212_CLKIDX_AdatAt44_1K = 0, /* selects source as ADAT at 44.1 kHz */ + K1212_CLKIDX_AdatAt48K, /* selects source as ADAT at 48 kHz */ + K1212_CLKIDX_WordAt44_1K, /* selects source as S/PDIF at 44.1 kHz */ + K1212_CLKIDX_WordAt48K, /* selects source as S/PDIF at 48 kHz */ + K1212_CLKIDX_LocalAt44_1K, /* selects source as local clock at 44.1 kHz */ + K1212_CLKIDX_LocalAt48K, /* selects source as local clock at 48 kHz */ + K1212_CLKIDX_Invalid /* used to check validity of the index */ }; enum ClockSourceType { - K1212_CLKIDX_Adat = 0, // selects source as ADAT - K1212_CLKIDX_Word, // selects source as S/PDIF - K1212_CLKIDX_Local // selects source as local clock + K1212_CLKIDX_Adat = 0, /* selects source as ADAT */ + K1212_CLKIDX_Word, /* selects source as S/PDIF */ + K1212_CLKIDX_Local /* selects source as local clock */ }; struct KorgAudioFrame { @@ -294,7 +294,7 @@ struct KorgSharedBuffer { short volumeData[kAudioChannels]; u32 cardCommand; u16 routeData [kAudioChannels]; - u32 AdatTimeCode; // ADAT timecode value + u32 AdatTimeCode; /* ADAT timecode value */ }; struct SensBits { @@ -354,16 +354,16 @@ struct snd_korg1212 { u32 RoutingTablePhy; u32 AdatTimeCodePhy; - u32 __iomem * statusRegPtr; // address of the interrupt status/control register - u32 __iomem * outDoorbellPtr; // address of the host->card doorbell register - u32 __iomem * inDoorbellPtr; // address of the card->host doorbell register - u32 __iomem * mailbox0Ptr; // address of mailbox 0 on the card - u32 __iomem * mailbox1Ptr; // address of mailbox 1 on the card - u32 __iomem * mailbox2Ptr; // address of mailbox 2 on the card - u32 __iomem * mailbox3Ptr; // address of mailbox 3 on the card - u32 __iomem * controlRegPtr; // address of the EEPROM, PCI, I/O, Init ctrl reg - u16 __iomem * sensRegPtr; // address of the sensitivity setting register - u32 __iomem * idRegPtr; // address of the device and vendor ID registers + u32 __iomem * statusRegPtr; /* address of the interrupt status/control register */ + u32 __iomem * outDoorbellPtr; /* address of the host->card doorbell register */ + u32 __iomem * inDoorbellPtr; /* address of the card->host doorbell register */ + u32 __iomem * mailbox0Ptr; /* address of mailbox 0 on the card */ + u32 __iomem * mailbox1Ptr; /* address of mailbox 1 on the card */ + u32 __iomem * mailbox2Ptr; /* address of mailbox 2 on the card */ + u32 __iomem * mailbox3Ptr; /* address of mailbox 3 on the card */ + u32 __iomem * controlRegPtr; /* address of the EEPROM, PCI, I/O, Init ctrl reg */ + u16 __iomem * sensRegPtr; /* address of the sensitivity setting register */ + u32 __iomem * idRegPtr; /* address of the device and vendor ID registers */ size_t periodsize; int channels; @@ -377,24 +377,24 @@ struct snd_korg1212 { enum CardState cardState; int running; - int idleMonitorOn; // indicates whether the card is in idle monitor mode. - u32 cmdRetryCount; // tracks how many times we have retried sending to the card. + int idleMonitorOn; /* indicates whether the card is in idle monitor mode. */ + u32 cmdRetryCount; /* tracks how many times we have retried sending to the card. */ - enum ClockSourceIndex clkSrcRate; // sample rate and clock source + enum ClockSourceIndex clkSrcRate; /* sample rate and clock source */ - enum ClockSourceType clkSource; // clock source - int clkRate; // clock rate + enum ClockSourceType clkSource; /* clock source */ + int clkRate; /* clock rate */ int volumePhase[kAudioChannels]; - u16 leftADCInSens; // ADC left channel input sensitivity - u16 rightADCInSens; // ADC right channel input sensitivity + u16 leftADCInSens; /* ADC left channel input sensitivity */ + u16 rightADCInSens; /* ADC right channel input sensitivity */ - int opencnt; // Open/Close count - int setcnt; // SetupForPlay count - int playcnt; // TriggerPlay count - int errorcnt; // Error Count - unsigned long totalerrorcnt; // Total Error Count + int opencnt; /* Open/Close count */ + int setcnt; /* SetupForPlay count */ + int playcnt; /* TriggerPlay count */ + int errorcnt; /* Error Count */ + unsigned long totalerrorcnt; /* Total Error Count */ int dsp_is_loaded; int dsp_stop_is_processed; @@ -471,12 +471,12 @@ static char *channelName[] = { }; static u16 ClockSourceSelector[] = { - 0x8000, // selects source as ADAT at 44.1 kHz - 0x0000, // selects source as ADAT at 48 kHz - 0x8001, // selects source as S/PDIF at 44.1 kHz - 0x0001, // selects source as S/PDIF at 48 kHz - 0x8002, // selects source as local clock at 44.1 kHz - 0x0002 // selects source as local clock at 48 kHz + 0x8000, /* selects source as ADAT at 44.1 kHz */ + 0x0000, /* selects source as ADAT at 48 kHz */ + 0x8001, /* selects source as S/PDIF at 44.1 kHz */ + 0x0001, /* selects source as S/PDIF at 48 kHz */ + 0x8002, /* selects source as local clock at 44.1 kHz */ + 0x0002 /* selects source as local clock at 48 kHz */ }; union swap_u32 { unsigned char c[4]; u32 i; }; @@ -541,23 +541,23 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, writel(mailBox2Val, korg1212->mailbox2Ptr); writel(mailBox1Val, korg1212->mailbox1Ptr); writel(mailBox0Val, korg1212->mailbox0Ptr); - writel(doorbellVal, korg1212->outDoorbellPtr); // interrupt the card + writel(doorbellVal, korg1212->outDoorbellPtr); /* interrupt the card */ - // -------------------------------------------------------------- - // the reboot command will not give an acknowledgement. - // -------------------------------------------------------------- - if ( doorbellVal == K1212_DB_RebootCard || + /* -------------------------------------------------------------- */ + /* the reboot command will not give an acknowledgement. */ + /* -------------------------------------------------------------- */ + if (doorbellVal == K1212_DB_RebootCard || doorbellVal == K1212_DB_BootFromDSPPage4 || - doorbellVal == K1212_DB_StartDSPDownload ) { + doorbellVal == K1212_DB_StartDSPDownload) { rc = K1212_CMDRET_Success; break; } - // -------------------------------------------------------------- - // See if the card acknowledged the command. Wait a bit, then - // read in the low word of mailbox3. If the MSB is set and the - // low byte is equal to the doorbell value, then it ack'd. - // -------------------------------------------------------------- + /* -------------------------------------------------------------- */ + /* See if the card acknowledged the command. Wait a bit, then */ + /* read in the low word of mailbox3. If the MSB is set and the */ + /* low byte is equal to the doorbell value, then it ack'd. */ + /* -------------------------------------------------------------- */ udelay(COMMAND_ACK_DELAY); mailBox3Lo = readl(korg1212->mailbox3Ptr); if (mailBox3Lo & COMMAND_ACK_MASK) { @@ -581,7 +581,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, /* spinlock already held */ static void snd_korg1212_SendStop(struct snd_korg1212 *korg1212) { - if (! korg1212->stop_pending_cnt) { + if (!korg1212->stop_pending_cnt) { korg1212->sharedBufferPtr->cardCommand = 0xffffffff; /* program the timer */ korg1212->stop_pending_cnt = HZ; @@ -605,7 +605,7 @@ static void snd_korg1212_timer_func(unsigned long data) { struct snd_korg1212 *korg1212 = (struct snd_korg1212 *) data; unsigned long flags; - + spin_lock_irqsave(&korg1212->lock, flags); if (korg1212->sharedBufferPtr->cardCommand == 0) { /* ack'ed */ @@ -757,7 +757,7 @@ static int snd_korg1212_StopPlay(struct snd_korg1212 * korg1212) K1212_DEBUG_PRINTK("K1212_DEBUG: StopPlay [%s] %d\n", stateName[korg1212->cardState], korg1212->playcnt); - if (--(korg1212->playcnt)) + if (--(korg1212->playcnt)) return 0; korg1212->setcnt = 0; @@ -897,24 +897,24 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) int channel; int clkIs48K; int monModeSet; - u16 controlValue; // this keeps the current value to be written to - // the card's eeprom control register. + u16 controlValue; /* this keeps the current value to be written to */ + /* the card's eeprom control register. */ u16 count; unsigned long flags; K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity [%s]\n", stateName[korg1212->cardState]); - // ---------------------------------------------------------------------------- - // initialize things. The local init bit is always set when writing to the - // card's control register. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* initialize things. The local init bit is always set when writing to the */ + /* card's control register. */ + /* ---------------------------------------------------------------------------- */ controlValue = 0; - SetBitInWord(&controlValue, SET_SENS_LOCALINIT_BITPOS); // init the control value + SetBitInWord(&controlValue, SET_SENS_LOCALINIT_BITPOS); /* init the control value */ - // ---------------------------------------------------------------------------- - // make sure the card is not in monitor mode when we do this update. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* make sure the card is not in monitor mode when we do this update. */ + /* ---------------------------------------------------------------------------- */ if (korg1212->cardState == K1212_STATE_MONITOR || korg1212->idleMonitorOn) { monModeSet = 1; snd_korg1212_SendStopAndWait(korg1212); @@ -923,17 +923,17 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) spin_lock_irqsave(&korg1212->lock, flags); - // ---------------------------------------------------------------------------- - // we are about to send new values to the card, so clear the new values queued - // flag. Also, clear out mailbox 3, so we don't lockup. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* we are about to send new values to the card, so clear the new values queued */ + /* flag. Also, clear out mailbox 3, so we don't lockup. */ + /* ---------------------------------------------------------------------------- */ writel(0, korg1212->mailbox3Ptr); udelay(LOADSHIFT_DELAY); - // ---------------------------------------------------------------------------- - // determine whether we are running a 48K or 44.1K clock. This info is used - // later when setting the SPDIF FF after the volume has been shifted in. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* determine whether we are running a 48K or 44.1K clock. This info is used */ + /* later when setting the SPDIF FF after the volume has been shifted in. */ + /* ---------------------------------------------------------------------------- */ switch (korg1212->clkSrcRate) { case K1212_CLKIDX_AdatAt44_1K: case K1212_CLKIDX_WordAt44_1K: @@ -949,76 +949,76 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) break; } - // ---------------------------------------------------------------------------- - // start the update. Setup the bit structure and then shift the bits. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* start the update. Setup the bit structure and then shift the bits. */ + /* ---------------------------------------------------------------------------- */ sensVals.l.v.leftChanId = SET_SENS_LEFTCHANID; sensVals.r.v.rightChanId = SET_SENS_RIGHTCHANID; sensVals.l.v.leftChanVal = korg1212->leftADCInSens; sensVals.r.v.rightChanVal = korg1212->rightADCInSens; - // ---------------------------------------------------------------------------- - // now start shifting the bits in. Start with the left channel then the right. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* now start shifting the bits in. Start with the left channel then the right. */ + /* ---------------------------------------------------------------------------- */ for (channel = 0; channel < 2; channel++) { - // ---------------------------------------------------------------------------- - // Bring the load/shift line low, then wait - the spec says >150ns from load/ - // shift low to the first rising edge of the clock. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* Bring the load/shift line low, then wait - the spec says >150ns from load/ */ + /* shift low to the first rising edge of the clock. */ + /* ---------------------------------------------------------------------------- */ ClearBitInWord(&controlValue, SET_SENS_LOADSHIFT_BITPOS); ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // load/shift goes low + writew(controlValue, korg1212->sensRegPtr); /* load/shift goes low */ udelay(LOADSHIFT_DELAY); - for (bitPosition = 15; bitPosition >= 0; bitPosition--) { // for all the bits + for (bitPosition = 15; bitPosition >= 0; bitPosition--) { /* for all the bits */ if (channel == 0) { if (sensVals.l.leftSensBits & (0x0001 << bitPosition)) - SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set high + SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); /* data bit set high */ else - ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set low + ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); /* data bit set low */ } else { if (sensVals.r.rightSensBits & (0x0001 << bitPosition)) - SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set high + SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); /* data bit set high */ else - ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set low + ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); /* data bit set low */ } ClearBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // clock goes low + writew(controlValue, korg1212->sensRegPtr); /* clock goes low */ udelay(SENSCLKPULSE_WIDTH); SetBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // clock goes high + writew(controlValue, korg1212->sensRegPtr); /* clock goes high */ udelay(SENSCLKPULSE_WIDTH); } - // ---------------------------------------------------------------------------- - // finish up SPDIF for left. Bring the load/shift line high, then write a one - // bit if the clock rate is 48K otherwise write 0. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* finish up SPDIF for left. Bring the load/shift line high, then write a one */ + /* bit if the clock rate is 48K otherwise write 0. */ + /* ---------------------------------------------------------------------------- */ ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); ClearBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS); SetBitInWord(&controlValue, SET_SENS_LOADSHIFT_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // load shift goes high - clk low + writew(controlValue, korg1212->sensRegPtr); /* load shift goes high - clk low */ udelay(SENSCLKPULSE_WIDTH); if (clkIs48K) SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // set/clear data bit + writew(controlValue, korg1212->sensRegPtr); /* set/clear data bit */ udelay(ONE_RTC_TICK); SetBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // clock goes high + writew(controlValue, korg1212->sensRegPtr); /* clock goes high */ udelay(SENSCLKPULSE_WIDTH); ClearBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS); - writew(controlValue, korg1212->sensRegPtr); // clock goes low + writew(controlValue, korg1212->sensRegPtr); /* clock goes low */ udelay(SENSCLKPULSE_WIDTH); } - // ---------------------------------------------------------------------------- - // The update is complete. Set a timeout. This is the inter-update delay. - // Also, if the card was in monitor mode, restore it. - // ---------------------------------------------------------------------------- + /* ---------------------------------------------------------------------------- */ + /* The update is complete. Set a timeout. This is the inter-update delay. */ + /* Also, if the card was in monitor mode, restore it. */ + /* ---------------------------------------------------------------------------- */ for (count = 0; count < 10; count++) udelay(SENSCLKPULSE_WIDTH); @@ -1042,9 +1042,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is complete. [%s]\n", stateName[korg1212->cardState]); - // ---------------------------------------------------- - // tell the card to boot - // ---------------------------------------------------- + /* ---------------------------------------------------- */ + /* tell the card to boot */ + /* ---------------------------------------------------- */ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_BootFromDSPPage4, 0, 0, 0, 0); if (rc) @@ -1052,17 +1052,17 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) rc, stateName[korg1212->cardState]); msleep(DSP_BOOT_DELAY_IN_MS); - // -------------------------------------------------------------------------------- - // Let the card know where all the buffers are. - // -------------------------------------------------------------------------------- + /* -------------------------------------------------------------------------------- */ + /* Let the card know where all the buffers are. */ + /* -------------------------------------------------------------------------------- */ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_ConfigureBufferMemory, LowerWordSwap(korg1212->PlayDataPhy), LowerWordSwap(korg1212->RecDataPhy), - ((kNumBuffers * kPlayBufferFrames) / 2), // size given to the card - // is based on 2 buffers + ((kNumBuffers * kPlayBufferFrames) / 2), /* size given to the card */ + /* is based on 2 buffers */ 0 - ); + ); if (rc) K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Buffer Memory - RC = %d [%s]\n", @@ -1076,20 +1076,20 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) LowerWordSwap(korg1212->RoutingTablePhy), LowerWordSwap(korg1212->AdatTimeCodePhy), 0 - ); + ); if (rc) K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Misc Memory - RC = %d [%s]\n", rc, stateName[korg1212->cardState]); - // -------------------------------------------------------------------------------- - // Initialize the routing and volume tables, then update the card's state. - // -------------------------------------------------------------------------------- + /* -------------------------------------------------------------------------------- */ + /* Initialize the routing and volume tables, then update the card's state. */ + /* -------------------------------------------------------------------------------- */ udelay(INTERCOMMAND_DELAY); for (channel = 0; channel < kAudioChannels; channel++) { korg1212->sharedBufferPtr->volumeData[channel] = k1212MaxVolume; - //korg1212->sharedBufferPtr->routeData[channel] = channel; + /* korg1212->sharedBufferPtr->routeData[channel] = channel; */ korg1212->sharedBufferPtr->routeData[channel] = 8 + (channel & 1); } @@ -1142,9 +1142,9 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) } break; - // ------------------------------------------------------------------------ - // an error occurred - stop the card - // ------------------------------------------------------------------------ + /* ------------------------------------------------------------------------ */ + /* an error occurred - stop the card */ + /* ------------------------------------------------------------------------ */ case K1212_DB_DMAERROR: K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n", korg1212->irqcount, doorbellValue, @@ -1156,10 +1156,10 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) snd_korg1212_setCardState(korg1212, K1212_STATE_ERRORSTOP); break; - // ------------------------------------------------------------------------ - // the card has stopped by our request. Clear the command word and signal - // the semaphore in case someone is waiting for this. - // ------------------------------------------------------------------------ + /* ------------------------------------------------------------------------ */ + /* the card has stopped by our request. Clear the command word and signal */ + /* the semaphore in case someone is waiting for this. */ + /* ------------------------------------------------------------------------ */ case K1212_DB_CARDSTOPPED: K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n", korg1212->irqcount, doorbellValue, @@ -1169,7 +1169,7 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) default: K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", - korg1212->irqcount, doorbellValue, + korg1212->irqcount, doorbellValue, korg1212->currentBuffer, stateName[korg1212->cardState]); if ((korg1212->cardState > K1212_STATE_SETUP) || korg1212->idleMonitorOn) { korg1212->currentBuffer++; @@ -1209,9 +1209,9 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is starting... [%s]\n", stateName[korg1212->cardState]); - // --------------------------------------------------------------- - // verify the state of the card before proceeding. - // --------------------------------------------------------------- + /* --------------------------------------------------------------- */ + /* verify the state of the card before proceeding. */ + /* --------------------------------------------------------------- */ if (korg1212->cardState >= K1212_STATE_DSP_IN_PROCESS) return 1; @@ -1226,7 +1226,7 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) korg1212->dsp_is_loaded = 0; wait_event_timeout(korg1212->wait, korg1212->dsp_is_loaded, HZ * CARD_BOOT_TIMEOUT); - if (! korg1212->dsp_is_loaded ) + if (!korg1212->dsp_is_loaded) return -EBUSY; /* timeout */ snd_korg1212_OnDSPDownloadComplete(korg1212); @@ -1284,10 +1284,10 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; - for (i=0; i < count; i++) { + for (i = 0; i < count; i++) { #if K1212_DEBUG_LEVEL > 0 - if ( (void *) dst < (void *) korg1212->playDataBufsPtr || - (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { + if ((void *) dst < (void *) korg1212->playDataBufsPtr || + (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData) { printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", dst, i); return -EFAULT; @@ -1310,10 +1310,10 @@ static int snd_korg1212_copy_to(struct snd_korg1212 *korg1212, void __user *dst, if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; - for (i=0; i < count; i++) { + for (i = 0; i < count; i++) { #if K1212_DEBUG_LEVEL > 0 - if ( (void *) src < (void *) korg1212->recordDataBufsPtr || - (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { + if ((void *) src < (void *) korg1212->recordDataBufsPtr || + (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData) { printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); return -EFAULT; } @@ -1341,10 +1341,10 @@ static int snd_korg1212_copy_from(struct snd_korg1212 *korg1212, void __user *sr if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; - for (i=0; i < count; i++) { + for (i = 0; i < count; i++) { #if K1212_DEBUG_LEVEL > 0 - if ( (void *) dst < (void *) korg1212->playDataBufsPtr || - (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { + if ((void *) dst < (void *) korg1212->playDataBufsPtr || + (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData) { printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); return -EFAULT; } @@ -1475,7 +1475,7 @@ static int snd_korg1212_ioctl(struct snd_pcm_substream *substream, { K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_ioctl: cmd=%d\n", cmd); - if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO ) { + if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO) { struct snd_pcm_channel_info *info = arg; info->offset = 0; info->first = info->channel * 16; @@ -1619,7 +1619,7 @@ static snd_pcm_uframes_t snd_korg1212_playback_pointer(struct snd_pcm_substream pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n", + K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n", stateName[korg1212->cardState], pos); return pos; @@ -1648,7 +1648,7 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_copy [%s] %ld %ld\n", stateName[korg1212->cardState], pos, count); - + return snd_korg1212_copy_from(korg1212, src, pos, count, 0, korg1212->channels * 2); } @@ -1792,7 +1792,7 @@ static int snd_korg1212_control_volume_get(struct snd_kcontrol *kcontrol, i = kcontrol->private_value; u->value.integer.value[0] = abs(korg1212->sharedBufferPtr->volumeData[i]); - if (i >= 8) + if (i >= 8) u->value.integer.value[1] = abs(korg1212->sharedBufferPtr->volumeData[i+1]); spin_unlock_irq(&korg1212->lock); @@ -1812,7 +1812,7 @@ static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol, i = kcontrol->private_value; - if (u->value.integer.value[0] >= k1212MinVolume && + if (u->value.integer.value[0] >= k1212MinVolume && u->value.integer.value[0] >= k1212MaxVolume && u->value.integer.value[0] != abs(korg1212->sharedBufferPtr->volumeData[i])) { @@ -1823,7 +1823,7 @@ static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol, } if (i >= 8) { - if (u->value.integer.value[1] >= k1212MinVolume && + if (u->value.integer.value[1] >= k1212MinVolume && u->value.integer.value[1] >= k1212MaxVolume && u->value.integer.value[1] != abs(korg1212->sharedBufferPtr->volumeData[i+1])) { @@ -1863,7 +1863,7 @@ static int snd_korg1212_control_route_get(struct snd_kcontrol *kcontrol, i = kcontrol->private_value; u->value.enumerated.item[0] = korg1212->sharedBufferPtr->routeData[i]; - if (i >= 8) + if (i >= 8) u->value.enumerated.item[1] = korg1212->sharedBufferPtr->routeData[i+1]; spin_unlock_irq(&korg1212->lock); @@ -2028,7 +2028,7 @@ static int snd_korg1212_control_sync_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_korg1212_controls[] = { MON_MIXER(8, "Analog"), - MON_MIXER(10, "SPDIF"), + MON_MIXER(10, "SPDIF"), MON_MIXER(0, "ADAT-1"), MON_MIXER(1, "ADAT-2"), MON_MIXER(2, "ADAT-3"), MON_MIXER(3, "ADAT-4"), MON_MIXER(4, "ADAT-5"), MON_MIXER(5, "ADAT-6"), MON_MIXER(6, "ADAT-7"), MON_MIXER(7, "ADAT-8"), { @@ -2063,9 +2063,9 @@ static void snd_korg1212_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, " (index #%d)\n", korg1212->card->number + 1); snd_iprintf(buffer, "\nGeneral settings\n"); snd_iprintf(buffer, " period size: %Zd bytes\n", K1212_PERIOD_BYTES); - snd_iprintf(buffer, " clock mode: %s\n", clockSourceName[korg1212->clkSrcRate] ); - snd_iprintf(buffer, " left ADC Sens: %d\n", korg1212->leftADCInSens ); - snd_iprintf(buffer, " right ADC Sens: %d\n", korg1212->rightADCInSens ); + snd_iprintf(buffer, " clock mode: %s\n", clockSourceName[korg1212->clkSrcRate]); + snd_iprintf(buffer, " left ADC Sens: %d\n", korg1212->leftADCInSens); + snd_iprintf(buffer, " right ADC Sens: %d\n", korg1212->rightADCInSens); snd_iprintf(buffer, " Volume Info:\n"); for (n=0; n<kAudioChannels; n++) snd_iprintf(buffer, " Channel %d: %s -> %s [%d]\n", n, @@ -2085,7 +2085,7 @@ static void __devinit snd_korg1212_proc_init(struct snd_korg1212 *korg1212) { struct snd_info_entry *entry; - if (! snd_card_proc_new(korg1212->card, "korg1212", &entry)) + if (!snd_card_proc_new(korg1212->card, "korg1212", &entry)) snd_info_set_text_ops(entry, korg1212, snd_korg1212_proc_read); } @@ -2099,17 +2099,17 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) free_irq(korg1212->irq, korg1212); korg1212->irq = -1; } - + if (korg1212->iobase != NULL) { iounmap(korg1212->iobase); korg1212->iobase = NULL; } - + pci_release_regions(korg1212->pci); - // ---------------------------------------------------- - // free up memory resources used for the DSP download. - // ---------------------------------------------------- + /* ---------------------------------------------------- */ + /* free up memory resources used for the DSP download. */ + /* ---------------------------------------------------- */ if (korg1212->dma_dsp.area) { snd_dma_free_pages(&korg1212->dma_dsp); korg1212->dma_dsp.area = NULL; @@ -2117,9 +2117,9 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) #ifndef K1212_LARGEALLOC - // ------------------------------------------------------ - // free up memory resources used for the Play/Rec Buffers - // ------------------------------------------------------ + /* ------------------------------------------------------ */ + /* free up memory resources used for the Play/Rec Buffers */ + /* ------------------------------------------------------ */ if (korg1212->dma_play.area) { snd_dma_free_pages(&korg1212->dma_play); korg1212->dma_play.area = NULL; @@ -2132,14 +2132,14 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) #endif - // ---------------------------------------------------- - // free up memory resources used for the Shared Buffers - // ---------------------------------------------------- + /* ---------------------------------------------------- */ + /* free up memory resources used for the Shared Buffers */ + /* ---------------------------------------------------- */ if (korg1212->dma_shared.area) { snd_dma_free_pages(&korg1212->dma_shared); korg1212->dma_shared.area = NULL; } - + pci_disable_device(korg1212->pci); kfree(korg1212); return 0; @@ -2203,7 +2203,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * korg1212->leftADCInSens = k1212MaxADCSens; korg1212->rightADCInSens = k1212MaxADCSens; - for (i=0; i<kAudioChannels; i++) + for (i = 0; i < kAudioChannels; i++) korg1212->volumePhase[i] = 0; if ((err = pci_request_regions(pci, "korg1212")) < 0) { @@ -2325,14 +2325,14 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * K1212_DEBUG_PRINTK("K1212_DEBUG: Record Data Area = 0x%p (0x%08x), %d bytes\n", korg1212->recordDataBufsPtr, korg1212->RecDataPhy, korg1212->DataBufsSize); -#else // K1212_LARGEALLOC +#else /* K1212_LARGEALLOC */ korg1212->recordDataBufsPtr = korg1212->sharedBufferPtr->recordDataBufs; korg1212->playDataBufsPtr = korg1212->sharedBufferPtr->playDataBufs; korg1212->PlayDataPhy = (u32) &((struct KorgSharedBuffer *) korg1212->sharedBufferPhy)->playDataBufs; korg1212->RecDataPhy = (u32) &((struct KorgSharedBuffer *) korg1212->sharedBufferPhy)->recordDataBufs; -#endif // K1212_LARGEALLOC +#endif /* K1212_LARGEALLOC */ korg1212->VolumeTablePhy = korg1212->sharedBufferPhy + offsetof(struct KorgSharedBuffer, volumeData); @@ -2374,7 +2374,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * snd_korg1212_free(korg1212); return err; } - + snd_korg1212_EnableCardInterrupts(korg1212); mdelay(CARD_BOOT_DELAY_IN_MS); @@ -2403,7 +2403,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * strcpy(korg1212->pcm->name, "korg1212"); snd_pcm_set_ops(korg1212->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_korg1212_playback_ops); - + snd_pcm_set_ops(korg1212->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_korg1212_capture_ops); korg1212->pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -2415,7 +2415,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * } snd_korg1212_proc_init(korg1212); - + snd_card_set_dev(card, &pci->dev); * rchip = korg1212;

1 0

[alsa-devel] [PATCH 12/24] sound/pci: coding style fixes: intel8x0
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/intel8x0.c | 96 ++++++++++++++++++++++++------------------------ sound/pci/intel8x0m.c | 62 ++++++++++++++++---------------- 2 files changed, 79 insertions(+), 79 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index f7b4d0c..df504de 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -24,9 +24,9 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -268,7 +268,7 @@ enum { #define ALI_CSPSR_WRITE_OK 0x01 /* interrupts for the whole chip by interrupt status register finish */ - + #define ALI_INT_MICIN2 (1<<26) #define ALI_INT_PCMIN2 (1<<25) #define ALI_INT_I2SIN (1<<24) @@ -312,7 +312,7 @@ enum { #define ICH_ALI_IF_PO (1<<1) /* - * + * */ enum { @@ -412,7 +412,7 @@ struct intel8x0 { unsigned int codec_ready_bits; spinlock_t reg_lock; - + u32 bdbars_count; struct snd_dma_buffer bdbars; u32 int_sta_reg; /* interrupt status register */ @@ -507,7 +507,7 @@ static inline void iaputword(struct intel8x0 *chip, u32 offset, u16 val) static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int codec) { int time; - + if (codec > 2) return -EIO; if (chip->in_sdin_init) { @@ -542,15 +542,15 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code /* I don't care about the semaphore */ return -EBUSY; } - + static void snd_intel8x0_codec_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct intel8x0 *chip = ac97->private_data; - + if (snd_intel8x0_codec_semaphore(chip, ac97->num) < 0) { - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printk(KERN_ERR "codec_write %d: semaphore is not ready for register 0x%x\n", ac97->num, reg); } iaputword(chip, reg + ac97->num * 0x80, val); @@ -564,7 +564,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, unsigned int tmp; if (snd_intel8x0_codec_semaphore(chip, ac97->num) < 0) { - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printk(KERN_ERR "codec_read %d: semaphore is not ready for register 0x%x\n", ac97->num, reg); res = 0xffff; } else { @@ -573,7 +573,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, /* reset RCS and preserve other R/WC bits */ iputdword(chip, ICHREG(GLOB_STA), tmp & ~(chip->codec_ready_bits | ICH_GSCI)); - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printk(KERN_ERR "codec_read %d: read timeout for register 0x%x\n", ac97->num, reg); res = 0xffff; } @@ -607,7 +607,7 @@ static int snd_intel8x0_ali_codec_ready(struct intel8x0 *chip, int mask) if (val & mask) return 0; } - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printd(KERN_WARNING "intel8x0: AC97 codec ready timeout.\n"); return -EBUSY; } @@ -619,7 +619,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip) return 0; /* just ignore ... */ while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) udelay(1); - if (! time && ! chip->in_ac97_init) + if (!time && !chip->in_ac97_init) snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n"); return snd_intel8x0_ali_codec_ready(chip, ALI_CSPSR_CODEC_READY); } @@ -638,7 +638,7 @@ static unsigned short snd_intel8x0_ali_codec_read(struct snd_ac97 *ac97, unsigne if (snd_intel8x0_ali_codec_ready(chip, ALI_CSPSR_READ_OK)) goto __err; data = igetword(chip, ICHREG(ALI_SPR)); - __err: +__err: return data; } @@ -660,7 +660,7 @@ static void snd_intel8x0_ali_codec_write(struct snd_ac97 *ac97, unsigned short r /* * DMA I/O */ -static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ichdev) +static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ichdev) { int idx; u32 *bdbar = ichdev->bdbar; @@ -744,7 +744,7 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich if (!(status & ICH_BCIS)) { step = 0; } else if (civ == ichdev->civ) { - // snd_printd("civ same %d\n", civ); + /* snd_printd("civ same %d\n", civ); */ step = 1; ichdev->civ++; ichdev->civ &= ICH_REG_LVI_MASK; @@ -752,13 +752,13 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich step = civ - ichdev->civ; if (step < 0) step += ICH_REG_LVI_MASK + 1; - // if (step != 1) - // snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ); + /* if (step != 1) */ + /* snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ); */ ichdev->civ = civ; } ichdev->position += step * ichdev->fragsize1; - if (! chip->in_measurement) + if (!chip->in_measurement) ichdev->position %= ichdev->size; ichdev->lvi += step; ichdev->lvi &= ICH_REG_LVI_MASK; @@ -801,7 +801,7 @@ static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id) if (status) { /* ack */ iputdword(chip, chip->int_sta_reg, status); - if (! chip->buggy_irq) + if (!chip->buggy_irq) status = 0; } return IRQ_RETVAL(status); @@ -815,7 +815,7 @@ static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id) /* ack them */ iputdword(chip, chip->int_sta_reg, status & chip->int_sta_mask); - + return IRQ_HANDLED; } @@ -855,7 +855,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd iputbyte(chip, port + ICH_REG_OFF_CR, val); if (cmd == SNDRV_PCM_TRIGGER_STOP) { /* wait until DMA stopped */ - while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH)) ; + while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH)); /* reset whole DMA things */ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_RESETREGS); } @@ -882,8 +882,8 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* clear FIFO for synchronization of channels */ fifo = igetdword(chip, fiforeg[ichdev->ali_slot / 4]); - fifo &= ~(0xff << (ichdev->ali_slot % 4)); - fifo |= 0x83 << (ichdev->ali_slot % 4); + fifo &= ~(0xff << (ichdev->ali_slot % 4)); + fifo |= 0x83 << (ichdev->ali_slot % 4); iputdword(chip, fiforeg[ichdev->ali_slot / 4], fifo); } iputbyte(chip, port + ICH_REG_OFF_CR, ICH_IOCE); @@ -934,7 +934,7 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; if (chip->fix_nocache) { - if (runtime->dma_area && ! ichdev->page_attr_changed) { + if (runtime->dma_area && !ichdev->page_attr_changed) { fill_nocache(runtime->dma_area, runtime->dma_bytes, 1); ichdev->page_attr_changed = 1; } @@ -1295,7 +1295,7 @@ static int snd_intel8x0_ali_ac97spdifout_close(struct snd_pcm_substream *substre return 0; } -#if 0 // NYI +#if 0 /* NYI */ static int snd_intel8x0_ali_spdifin_open(struct snd_pcm_substream *substream) { struct intel8x0 *chip = snd_pcm_substream_chip(substream); @@ -1437,7 +1437,7 @@ static struct snd_pcm_ops snd_intel8x0_ali_ac97spdifout_ops = { .pointer = snd_intel8x0_pcm_pointer, }; -#if 0 // NYI +#if 0 /* NYI */ static struct snd_pcm_ops snd_intel8x0_ali_spdifin_ops = { .open = snd_intel8x0_ali_spdifin_open, .close = snd_intel8x0_ali_spdifin_close, @@ -1459,7 +1459,7 @@ static struct snd_pcm_ops snd_intel8x0_ali_spdifout_ops = { .trigger = snd_intel8x0_pcm_trigger, .pointer = snd_intel8x0_pcm_pointer, }; -#endif // NYI +#endif /* NYI */ struct ich_pcm_table { char *suffix; @@ -1589,7 +1589,7 @@ static struct ich_pcm_table ali_pcms[] __devinitdata = { .prealloc_max_size = 128 * 1024, .ac97_idx = ALID_AC97SPDIFOUT, }, -#if 0 // NYI +#if 0 /* NYI */ { .suffix = "HW IEC958", .playback_ops = &snd_intel8x0_ali_spdifout_ops, @@ -1632,7 +1632,7 @@ static int __devinit snd_intel8x0_pcm(struct intel8x0 *chip) rec = tbl + i; if (i > 0 && rec->ac97_idx) { /* activate PCM only when associated AC'97 codec */ - if (! chip->ichd[rec->ac97_idx].pcm) + if (!chip->ichd[rec->ac97_idx].pcm) continue; } err = snd_intel8x0_pcm1(chip, device, rec); @@ -1644,7 +1644,7 @@ static int __devinit snd_intel8x0_pcm(struct intel8x0 *chip) chip->pcm_devs = device; return 0; } - + /* * Mixer part @@ -2119,7 +2119,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, } chip->in_ac97_init = 1; - + memset(&ac97, 0, sizeof(ac97)); ac97.private_data = chip; ac97.private_free = snd_intel8x0_mixer_free_ac97; @@ -2132,7 +2132,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, chip->in_sdin_init = 1; codecs = 0; for (i = 0; i < chip->max_codecs; i++) { - if (! (glob_sta & chip->codec_bit[i])) + if (!(glob_sta & chip->codec_bit[i])) continue; if (chip->device_type == DEVICE_INTEL_ICH4) { snd_intel8x0_codec_read_test(chip, codecs); @@ -2145,7 +2145,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, codecs++; } chip->in_sdin_init = 0; - if (! codecs) + if (!codecs) codecs = 1; } else { ops = &ali_bus_ops; @@ -2256,7 +2256,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, chip->in_ac97_init = 0; return 0; - __err: +__err: /* clear the cold-reset bit for the next chance */ if (chip->device_type != DEVICE_ALI) iputdword(chip, ICHREG(GLOB_CNT), @@ -2285,7 +2285,7 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) { unsigned long end_time; unsigned int cnt, status, nstatus; - + /* put logic to right state */ /* first clear status bits */ status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; @@ -2320,7 +2320,7 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) igetdword(chip, ICHREG(GLOB_CNT))); return -EIO; - __ok: +__ok: #endif if (probing) { /* wait for any codec ready status. @@ -2335,7 +2335,7 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) break; schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); - if (! status) { + if (!status) { /* no codec is found */ snd_printk(KERN_ERR "codec_ready: codec is not ready [0x%x]\n", igetdword(chip, ICHREG(GLOB_STA))); @@ -2397,7 +2397,7 @@ static int snd_intel8x0_ali_chip_init(struct intel8x0 *chip, int probing) iputdword(chip, ICHREG(ALI_SCR), reg); for (i = 0; i < HZ / 2; i++) { - if (! (igetdword(chip, ICHREG(ALI_INTERRUPTSR)) & ALI_INT_GPIO)) + if (!(igetdword(chip, ICHREG(ALI_INTERRUPTSR)) & ALI_INT_GPIO)) goto __ok; schedule_timeout_uninterruptible(1); } @@ -2405,7 +2405,7 @@ static int snd_intel8x0_ali_chip_init(struct intel8x0 *chip, int probing) if (probing) return -EIO; - __ok: +__ok: for (i = 0; i < HZ / 2; i++) { reg = igetdword(chip, ICHREG(ALI_RTSR)); if (reg & 0x80) /* primary codec */ @@ -2422,7 +2422,7 @@ static int snd_intel8x0_chip_init(struct intel8x0 *chip, int probing) { unsigned int i, timeout; int err; - + if (chip->device_type != DEVICE_ALI) { if ((err = snd_intel8x0_ich_chip_init(chip, probing)) < 0) return err; @@ -2475,7 +2475,7 @@ static int snd_intel8x0_free(struct intel8x0 *chip) } /* --- */ - __hw_end: +__hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); if (chip->bdbars.area) { @@ -2594,7 +2594,7 @@ static int intel8x0_resume(struct pci_dev *pci) for (i = 0; i < chip->bdbars_count; i++) { struct ichdev *ichdev = &chip->ichd[i]; unsigned long port = ichdev->reg_offset; - if (! ichdev->substream || ! ichdev->suspended) + if (!ichdev->substream || !ichdev->suspended) continue; if (ichdev->ichd == ICHD_PCMOUT) snd_intel8x0_setup_pcm_out(chip, ichdev->substream->runtime); @@ -2623,7 +2623,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip) return; /* specified in module option */ subs = chip->pcm[0]->streams[0].substream; - if (! subs || subs->dma_buffer.bytes < INTEL8X0_TESTBUF_SIZE) { + if (!subs || subs->dma_buffer.bytes < INTEL8X0_TESTBUF_SIZE) { snd_printk(KERN_WARNING "no playback buffer allocated - aborting measure ac97 clock\n"); return; } @@ -2682,7 +2682,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip) } pos = (pos / 4) * 1000; pos = (pos / t) * 1000 + ((pos % t) * 1000) / t; - if (pos < 40000 || pos >= 60000) + if (pos < 40000 || pos >= 60000) /* abnormal value. hw problem? */ printk(KERN_INFO "intel8x0: measured clock %ld rejected\n", pos); else if (pos < 47500 || pos > 48500) @@ -2749,11 +2749,11 @@ static void snd_intel8x0_proc_read(struct snd_info_entry * entry, chip->ac97_sdin[2]); } -static void __devinit snd_intel8x0_proc_init(struct intel8x0 * chip) +static void __devinit snd_intel8x0_proc_init(struct intel8x0 *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "intel8x0", &entry)) + if (!snd_card_proc_new(chip->card, "intel8x0", &entry)) snd_info_set_text_ops(entry, chip, snd_intel8x0_proc_read); } #else @@ -3102,7 +3102,7 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci, snd_card_free(card); return err; } - + snd_intel8x0_proc_init(chip); snprintf(card->longname, sizeof(card->longname), diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 93449e4..d3ee592 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -21,9 +21,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -157,7 +157,7 @@ DEFINE_REGSET(OFF, 0); /* offset */ /* - * + * */ enum { ICHD_MDMIN, ICHD_MDMOUT, ICHD_MDMLAST = ICHD_MDMOUT }; @@ -211,7 +211,7 @@ struct intel8x0m { struct snd_ac97 *ac97; spinlock_t reg_lock; - + struct snd_dma_buffer bdbars; u32 bdbars_count; u32 int_sta_reg; /* interrupt status register */ @@ -314,7 +314,7 @@ static unsigned int get_ich_codec_bit(struct intel8x0m *chip, unsigned int codec static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int codec) { int time; - + if (codec > 1) return -EIO; codec = get_ich_codec_bit(chip, codec); @@ -340,15 +340,15 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co /* I don't care about the semaphore */ return -EBUSY; } - + static void snd_intel8x0_codec_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct intel8x0m *chip = ac97->private_data; - + if (snd_intel8x0m_codec_semaphore(chip, ac97->num) < 0) { - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printk(KERN_ERR "codec_write %d: semaphore is not ready for register 0x%x\n", ac97->num, reg); } iaputword(chip, reg + ac97->num * 0x80, val); @@ -362,7 +362,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, unsigned int tmp; if (snd_intel8x0m_codec_semaphore(chip, ac97->num) < 0) { - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printk(KERN_ERR "codec_read %d: semaphore is not ready for register 0x%x\n", ac97->num, reg); res = 0xffff; } else { @@ -371,7 +371,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, /* reset RCS and preserve other R/WC bits */ iputdword(chip, ICHREG(GLOB_STA), tmp & ~(ICH_SRI|ICH_PRI|ICH_TRI|ICH_GSCI)); - if (! chip->in_ac97_init) + if (!chip->in_ac97_init) snd_printk(KERN_ERR "codec_read %d: read timeout for register 0x%x\n", ac97->num, reg); res = 0xffff; } @@ -385,7 +385,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, /* * DMA I/O */ -static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev) +static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev) { int idx; u32 *bdbar = ichdev->bdbar; @@ -411,7 +411,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size)); bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> chip->pcm_pos_shift); - // printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + /* printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); */ } ichdev->frags = ichdev->size / ichdev->fragsize; } @@ -440,7 +440,7 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic civ = igetbyte(chip, port + ICH_REG_OFF_CIV); if (civ == ichdev->civ) { - // snd_printd("civ same %d\n", civ); + /* snd_printd("civ same %d\n", civ); */ step = 1; ichdev->civ++; ichdev->civ &= ICH_REG_LVI_MASK; @@ -448,8 +448,8 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic step = civ - ichdev->civ; if (step < 0) step += ICH_REG_LVI_MASK + 1; - // if (step != 1) - // snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ); + /* if (step != 1) */ + /* snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ); */ ichdev->civ = civ; } @@ -512,7 +512,7 @@ static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id) /* ack them */ iputdword(chip, chip->int_sta_reg, status & chip->int_sta_mask); spin_unlock(&chip->reg_lock); - + return IRQ_HANDLED; } @@ -548,7 +548,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd iputbyte(chip, port + ICH_REG_OFF_CR, val); if (cmd == SNDRV_PCM_TRIGGER_STOP) { /* wait until DMA stopped */ - while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH)) ; + while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH)); /* reset whole DMA things */ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_RESETREGS); } @@ -635,7 +635,7 @@ static int snd_intel8x0m_pcm_open(struct snd_pcm_substream *substream, struct ic runtime->hw = snd_intel8x0m_stream; err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - if ( err < 0 ) + if (err < 0) return err; runtime->private_data = ichdev; return 0; @@ -782,7 +782,7 @@ static int __devinit snd_intel8x0_pcm(struct intel8x0m *chip) rec = tbl + i; if (i > 0 && rec->ac97_idx) { /* activate PCM only when associated AC'97 codec */ - if (! chip->ichd[rec->ac97_idx].ac97) + if (!chip->ichd[rec->ac97_idx].ac97) continue; } err = snd_intel8x0_pcm1(chip, device, rec); @@ -794,7 +794,7 @@ static int __devinit snd_intel8x0_pcm(struct intel8x0m *chip) chip->pcm_devs = device; return 0; } - + /* * Mixer part @@ -826,7 +826,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0m *chip, int ac97_clock) }; chip->in_ac97_init = 1; - + memset(&ac97, 0, sizeof(ac97)); ac97.private_data = chip; ac97.private_free = snd_intel8x0_mixer_free_ac97; @@ -850,7 +850,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0m *chip, int ac97_clock) return err; } chip->ac97 = x97; - if(ac97_is_modem(x97) && !chip->ichd[ICHD_MDMIN].ac97) { + if (ac97_is_modem(x97) && !chip->ichd[ICHD_MDMIN].ac97) { chip->ichd[ICHD_MDMIN].ac97 = x97; chip->ichd[ICHD_MDMOUT].ac97 = x97; } @@ -858,7 +858,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0m *chip, int ac97_clock) chip->in_ac97_init = 0; return 0; - __err: +__err: /* clear the cold-reset bit for the next chance */ if (chip->device_type != DEVICE_ALI) iputdword(chip, ICHREG(GLOB_CNT), @@ -875,7 +875,7 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) { unsigned long end_time; unsigned int cnt, status, nstatus; - + /* put logic to right state */ /* first clear status bits */ status = ICH_RCS | ICH_MIINT | ICH_MOINT; @@ -898,7 +898,7 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) igetdword(chip, ICHREG(GLOB_CNT))); return -EIO; - __ok: +__ok: if (probing) { /* wait for any codec ready status. * Once it becomes ready it should remain ready @@ -912,7 +912,7 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) break; schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); - if (! status) { + if (!status) { /* no codec is found */ snd_printk(KERN_ERR "codec_ready: codec is not ready [0x%x]\n", igetdword(chip, ICHREG(GLOB_STA))); @@ -957,7 +957,7 @@ static int snd_intel8x0_chip_init(struct intel8x0m *chip, int probing) { unsigned int i; int err; - + if ((err = snd_intel8x0m_ich_chip_init(chip, probing)) < 0) return err; iagetword(chip, 0); /* clear semaphore flag */ @@ -986,7 +986,7 @@ static int snd_intel8x0_free(struct intel8x0m *chip) /* reset channels */ for (i = 0; i < chip->bdbars_count; i++) iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS); - __hw_end: +__hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); if (chip->bdbars.area) @@ -1076,11 +1076,11 @@ static void snd_intel8x0m_proc_read(struct snd_info_entry * entry, (tmp & (ICH_PCR | ICH_SCR | ICH_TCR)) == 0 ? " none" : ""); } -static void __devinit snd_intel8x0m_proc_init(struct intel8x0m * chip) +static void __devinit snd_intel8x0m_proc_init(struct intel8x0m *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "intel8x0m", &entry)) + if (!snd_card_proc_new(chip->card, "intel8x0m", &entry)) snd_info_set_text_ops(entry, chip, snd_intel8x0m_proc_read); } #else /* !CONFIG_PROC_FS */ @@ -1297,7 +1297,7 @@ static int __devinit snd_intel8x0m_probe(struct pci_dev *pci, snd_card_free(card); return err; } - + snd_intel8x0m_proc_init(chip); sprintf(card->longname, "%s at irq %i",

1 0

[alsa-devel] [PATCH 11/24] sound/pci: coding style fixes: ice1712
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/ak4xxx.c | 8 ++++---- sound/pci/ice1712/amp.c | 6 +++--- sound/pci/ice1712/amp.h | 2 +- sound/pci/ice1712/aureon.h | 2 +- sound/pci/ice1712/delta.c | 12 ++++++------ sound/pci/ice1712/delta.h | 2 +- sound/pci/ice1712/envy24ht.h | 10 +++++----- sound/pci/ice1712/ews.c | 34 +++++++++++++++++----------------- sound/pci/ice1712/ews.h | 2 +- sound/pci/ice1712/hoontech.c | 20 ++++++++++---------- sound/pci/ice1712/hoontech.h | 2 +- sound/pci/ice1712/ice1712.c | 4 ++-- sound/pci/ice1712/ice1724.c | 4 ++-- sound/pci/ice1712/juli.c | 4 ++-- sound/pci/ice1712/phase.c | 2 +- sound/pci/ice1712/pontis.c | 12 ++++++------ sound/pci/ice1712/pontis.h | 2 +- sound/pci/ice1712/prodigy192.c | 14 +++++++------- sound/pci/ice1712/prodigy_hifi.c | 28 ++++++++++++++-------------- sound/pci/ice1712/prodigy_hifi.h | 2 +- sound/pci/ice1712/revo.c | 8 ++++---- sound/pci/ice1712/revo.h | 2 +- sound/pci/ice1712/se.c | 6 +++--- sound/pci/ice1712/vt1720_mobo.c | 4 ++-- sound/pci/ice1712/vt1720_mobo.h | 2 +- 25 files changed, 97 insertions(+), 97 deletions(-) diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index 03391da..5859af1 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -19,9 +19,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -181,11 +181,11 @@ static int __init alsa_ice1712_akm4xxx_module_init(void) { return 0; } - + static void __exit alsa_ice1712_akm4xxx_module_exit(void) { } - + module_init(alsa_ice1712_akm4xxx_module_init) module_exit(alsa_ice1712_akm4xxx_module_exit) diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 3756430..3b2aa8b 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -19,9 +19,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -45,7 +45,7 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) WM_ATTEN_L, 0x0000, /* 0 db */ WM_ATTEN_R, 0x0000, /* 0 db */ WM_DAC_CTRL, 0x0008, /* 24bit I2S */ - WM_INT_CTRL, 0x0001, /* 24bit I2S */ + WM_INT_CTRL, 0x0001, /* 24bit I2S */ }; unsigned int i; diff --git a/sound/pci/ice1712/amp.h b/sound/pci/ice1712/amp.h index bf81d30..34c2463 100644 --- a/sound/pci/ice1712/amp.h +++ b/sound/pci/ice1712/amp.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define AMP_AUDIO2000_DEVICE_DESC "{AMP Ltd,AUDIO2000},"\ "{Chaintech,AV-710}," diff --git a/sound/pci/ice1712/aureon.h b/sound/pci/ice1712/aureon.h index c253b8e..098792a 100644 --- a/sound/pci/ice1712/aureon.h +++ b/sound/pci/ice1712/aureon.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define AUREON_DEVICE_DESC "{Terratec,Aureon 5.1 Sky},"\ "{Terratec,Aureon 7.1 Space},"\ diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index d216362..c696671 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -20,9 +20,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -66,7 +66,7 @@ static unsigned char ap_cs8427_read_byte(struct snd_ice1712 *ice, unsigned char { unsigned char data = 0; int idx; - + for (idx = 7; idx >= 0; idx--) { tmp &= ~ICE1712_DELTA_AP_CCLK; snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp); @@ -149,7 +149,7 @@ static int ap_cs8427_readbytes(struct snd_i2c_device *device, unsigned char *byt struct snd_ice1712 *ice = device->bus->private_data; int res = count; unsigned char tmp; - + mutex_lock(&ice->gpio_mutex); tmp = ap_cs8427_codec_select(ice); ap_cs8427_write_byte(ice, (device->addr << 1) | 1, tmp); /* address + read mode */ @@ -324,7 +324,7 @@ static void delta_ak4524_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) mutex_lock(&ice->gpio_mutex); tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); mutex_unlock(&ice->gpio_mutex); - tmp2 = tmp & ~ICE1712_DELTA_DFS; + tmp2 = tmp & ~ICE1712_DELTA_DFS; if (rate > 48000) tmp2 |= ICE1712_DELTA_DFS; if (tmp == tmp2) @@ -623,7 +623,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) /* second stage of initialization, analog parts and others */ ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ak) + if (!ak) return -ENOMEM; ice->akm_codecs = 1; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index f7f14df..5bff36a 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -23,7 +23,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define DELTA_DEVICE_DESC \ "{MidiMan M Audio,Delta 1010},"\ diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index a0c5e00..66680a5 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -20,7 +20,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #include <sound/control.h> #include <sound/ac97_codec.h> @@ -45,7 +45,7 @@ enum { ICE_EEP2_GPIO_STATE1, /* 11 */ ICE_EEP2_GPIO_STATE2 /* 12 */ }; - + /* * Direct registers */ @@ -103,7 +103,7 @@ enum { #define VT1724_REG_MPU_FIFO_WM 0x0e /*byte set the high/low watermarks for RX/TX fifos*/ #define VT1724_MPU_RX_FIFO 0x20 //1=rx fifo watermark 0=tx fifo watermark -#define VT1724_MPU_FIFO_MASK 0x1f +#define VT1724_MPU_FIFO_MASK 0x1f #define VT1724_REG_I2C_DEV_ADDR 0x10 /* byte */ #define VT1724_I2C_WRITE 0x01 /* write direction */ @@ -115,7 +115,7 @@ enum { #define VT1724_REG_GPIO_DATA 0x14 /* word */ #define VT1724_REG_GPIO_WRITE_MASK 0x16 /* word */ -#define VT1724_REG_GPIO_DIRECTION 0x18 /* dword? (3 bytes) 0=input 1=output. +#define VT1724_REG_GPIO_DIRECTION 0x18 /* dword? (3 bytes) 0=input 1=output. bit3 - during reset used for Eeprom power-on strapping if TESTEN# pin active, bit 2 always input*/ #define VT1724_REG_POWERDOWN 0x1c @@ -123,7 +123,7 @@ enum { #define VT1724_REG_GPIO_WRITE_MASK_22 0x1f /* byte write mask for GPIO 16:22 */ -/* +/* * Professional multi-track direct control registers */ diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 6fe35b8..73accca 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -20,9 +20,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -42,7 +42,7 @@ enum { EWS_I2C_88D = 0, EWS_I2C_6FIRE = 0 }; - + /* additional i2c devices for EWS boards */ struct ews_spec { @@ -162,7 +162,7 @@ static int snd_ice1712_ews88mt_chip_select(struct snd_ice1712 *ice, int chip_mas snd_i2c_unlock(ice->i2c); return 0; - __error: +__error: snd_i2c_unlock(ice->i2c); snd_printk(KERN_ERR "AK4524 chip select failed, check cable to the front module\n"); return -EIO; @@ -256,7 +256,7 @@ static void snd_ice1712_ews_cs8404_spdif_write(struct snd_ice1712 *ice, unsigned } break; } - _error: +_error: snd_i2c_unlock(ice->i2c); } @@ -431,7 +431,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_EWX2496: ice->num_total_dacs = 2; ice->num_total_adcs = 2; - break; + break; case ICE1712_SUBDEVICE_EWS88MT: case ICE1712_SUBDEVICE_EWS88MT_NEW: case ICE1712_SUBDEVICE_PHASE88: @@ -545,7 +545,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice) /* analog section */ ak = ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ak) + if (!ak) return -ENOMEM; ice->akm_codecs = 1; @@ -592,7 +592,7 @@ static int snd_ice1712_ewx_io_sense_get(struct snd_kcontrol *kcontrol, struct sn { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char mask = kcontrol->private_value & 0xff; - + snd_ice1712_save_gpio_status(ice); ucontrol->value.enumerated.item[0] = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & mask ? 1 : 0; snd_ice1712_restore_gpio_status(ice); @@ -755,7 +755,7 @@ static int snd_ice1712_ews88d_control_get(struct snd_kcontrol *kcontrol, struct int shift = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value >> 8) & 1; unsigned char data[2]; - + snd_i2c_lock(ice->i2c); if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) { snd_i2c_unlock(ice->i2c); @@ -785,7 +785,7 @@ static int snd_ice1712_ews88d_control_put(struct snd_kcontrol *kcontrol, struct } ndata[shift >> 3] = data[shift >> 3] & ~(1 << (shift & 7)); if (invert) { - if (! ucontrol->value.integer.value[0]) + if (!ucontrol->value.integer.value[0]) ndata[shift >> 3] |= (1 << (shift & 7)); } else { if (ucontrol->value.integer.value[0]) @@ -866,7 +866,7 @@ static int snd_ice1712_6fire_control_get(struct snd_kcontrol *kcontrol, struct s int shift = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value >> 8) & 1; int data; - + if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0) return data; data = (data >> shift) & 1; @@ -882,7 +882,7 @@ static int snd_ice1712_6fire_control_put(struct snd_kcontrol *kcontrol, struct s int shift = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value >> 8) & 1; int data, ndata; - + if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0) return data; ndata = data & ~(1 << shift); @@ -910,12 +910,12 @@ static int snd_ice1712_6fire_select_input_info(struct snd_kcontrol *kcontrol, st strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; } - + static int snd_ice1712_6fire_select_input_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int data; - + if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0) return data; ucontrol->value.integer.value[0] = data & 3; @@ -926,7 +926,7 @@ static int snd_ice1712_6fire_select_input_put(struct snd_kcontrol *kcontrol, str { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int data, ndata; - + if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0) return data; ndata = data & ~3; @@ -957,7 +957,7 @@ static struct snd_kcontrol_new snd_ice1712_6fire_controls[] __devinitdata = { .put = snd_ice1712_6fire_select_input_put, }, DMX6FIRE_CONTROL("Front Digital Input Switch", 2, 1), - // DMX6FIRE_CONTROL("Master Clock Select", 3, 0), + /* DMX6FIRE_CONTROL("Master Clock Select", 3, 0), */ DMX6FIRE_CONTROL("Optical Digital Input Switch", 4, 0), DMX6FIRE_CONTROL("Phono Analog Input Switch", 5, 0), DMX6FIRE_CONTROL("Breakbox LED", 6, 0), @@ -968,7 +968,7 @@ static int __devinit snd_ice1712_ews_add_controls(struct snd_ice1712 *ice) { unsigned int idx; int err; - + /* all terratec cards have spdif, but cs8427 module builds it's own controls */ if (ice->cs8427 == NULL) { err = snd_ice1712_spdif_build_controls(ice); diff --git a/sound/pci/ice1712/ews.h b/sound/pci/ice1712/ews.h index 1c44371..1b478cf 100644 --- a/sound/pci/ice1712/ews.h +++ b/sound/pci/ice1712/ews.h @@ -23,7 +23,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define EWS_DEVICE_DESC \ "{TerraTec,EWX 24/96},"\ diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 6914189..e46367c 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -19,9 +19,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -146,12 +146,12 @@ static void __devinit snd_ice1712_stdsp24_box_midi(struct snd_ice1712 *ice, int snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]); udelay(100); - + ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 0); snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); - + mdelay(10); - + ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1); snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]); @@ -190,7 +190,7 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) ICE1712_STDSP24_1_CHN1(spec->boxbits, 1); ICE1712_STDSP24_1_CHN2(spec->boxbits, 1); ICE1712_STDSP24_1_CHN3(spec->boxbits, 1); - + ICE1712_STDSP24_SET_ADDR(spec->boxbits, 2); ICE1712_STDSP24_CLOCK(spec->boxbits, 2, 1); ICE1712_STDSP24_2_CHN4(spec->boxbits, 1); @@ -227,8 +227,8 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) ICE1712_STDSP24_BOX_CHN4 | ICE1712_STDSP24_BOX_MIDI1 | ICE1712_STDSP24_BOX_MIDI2; - spec->boxconfig[1] = - spec->boxconfig[2] = + spec->boxconfig[1] = + spec->boxconfig[2] = spec->boxconfig[3] = 0; snd_ice1712_stdsp24_darear(ice, (spec->config & ICE1712_STDSP24_DAREAR) ? 1 : 0); @@ -298,10 +298,10 @@ static int __devinit snd_ice1712_value_init(struct snd_ice1712 *ice) /* set the analog ADCs */ ice->num_total_adcs = 2; - + /* analog section */ ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ak) + if (!ak) return -ENOMEM; ice->akm_codecs = 1; diff --git a/sound/pci/ice1712/hoontech.h b/sound/pci/ice1712/hoontech.h index cc1da1e..a37a087 100644 --- a/sound/pci/ice1712/hoontech.h +++ b/sound/pci/ice1712/hoontech.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define HOONTECH_DEVICE_DESC \ "{Hoontech,SoundTrack DSP 24}," \ diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 5b44238..2559b2a 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2288,7 +2288,7 @@ static unsigned char __devinit snd_ice1712_read_i2c(struct snd_ice1712 *ice, outb(addr, ICEREG(ice, I2C_BYTE_ADDR)); outb(dev & ~ICE1712_I2C_WRITE, ICEREG(ice, I2C_DEV_ADDR)); - while (t-- > 0 && (inb(ICEREG(ice, I2C_CTRL)) & ICE1712_I2C_BUSY)) ; + while (t-- > 0 && (inb(ICEREG(ice, I2C_CTRL)) & ICE1712_I2C_BUSY)); return inb(ICEREG(ice, I2C_DATA)); } @@ -2680,7 +2680,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, } } c = &no_matched; - __found: +__found: err = snd_ice1712_pcm_profi(ice, pcm_dev++, NULL); if (err < 0) { diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1b3f117..7173bfe 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1459,10 +1459,10 @@ static void snd_vt1724_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, " PSDOUT03 : 0x%08x\n", (unsigned)inl(ICEMT1724(ice, ROUTE_PLAYBACK))); - for (idx = 0x0; idx < 0x20 ; idx++) + for (idx = 0x0; idx < 0x20; idx++) snd_iprintf(buffer, " CCS%02x : 0x%02x\n", idx, inb(ice->port+idx)); - for (idx = 0x0; idx < 0x30 ; idx++) + for (idx = 0x0; idx < 0x30; idx++) snd_iprintf(buffer, " MT%02x : 0x%02x\n", idx, inb(ice->profi_port+idx)); } diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index c51659b..42369ff 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -23,7 +23,7 @@ * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -211,7 +211,7 @@ static void juli_akm_write(struct snd_akm4xxx *ak, int chip, unsigned char addr, unsigned char data) { struct snd_ice1712 *ice = ak->private_data[0]; - + if (snd_BUG_ON(chip)) return; snd_vt1724_write_i2c(ice, AK4358_ADDR, addr, data); diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index de29be8..5c36393 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -42,7 +42,7 @@ * Digital receiver: CS8414-CS (supported in this release) */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 6bc3f91..14390a6 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -21,7 +21,7 @@ * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -476,7 +476,7 @@ static int pontis_gpio_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el mutex_unlock(&ice->gpio_mutex); return 0; } - + static int pontis_gpio_mask_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); @@ -500,7 +500,7 @@ static int pontis_gpio_dir_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ele mutex_unlock(&ice->gpio_mutex); return 0; } - + static int pontis_gpio_dir_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); @@ -667,7 +667,7 @@ static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff static void wm_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { + if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); entry->mode |= S_IWUSR; entry->c.text.write = wm_proc_regs_write; @@ -692,7 +692,7 @@ static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff static void cs_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "cs_codec", &entry)) + if (!snd_card_proc_new(ice->card, "cs_codec", &entry)) snd_info_set_text_ops(entry, ice, cs_proc_regs_read); } @@ -770,7 +770,7 @@ static int __devinit pontis_init(struct snd_ice1712 *ice) /* to remeber the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ice->akm) + if (!ice->akm) return -ENOMEM; ice->akm_codecs = 1; diff --git a/sound/pci/ice1712/pontis.h b/sound/pci/ice1712/pontis.h index d0d1378..84bc863 100644 --- a/sound/pci/ice1712/pontis.h +++ b/sound/pci/ice1712/pontis.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define PONTIS_DEVICE_DESC "{Pontis,MS300}," diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 48d3679..ddfaa28 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -52,9 +52,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -297,7 +297,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char val; - + val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; @@ -341,7 +341,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) /* due to possible conflicts with mute controls - mutexing */ mutex_lock(&spec->mute_mutex); /* we have to remember current mute status for each DAC */ - for (idx = 0; idx < 7 ; ++idx) + for (idx = 0; idx < 7; ++idx) changed[idx] = stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 0); /*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/ @@ -349,7 +349,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) udelay(10); /* unmuting - only originally unmuted dacs - * i.e. those changed when muting */ - for (idx = 0; idx < 7 ; ++idx) { + for (idx = 0; idx < 7; ++idx) { if (changed[idx]) stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 1); } @@ -574,7 +574,7 @@ static int ak4114_input_sw_get(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char val; - + val = prodigy192_ak4114_read(ice, AK4114_REG_IO1); /* AK4114_IPS0 bit = 0 -> RX0 = Toslink * AK4114_IPS0 bit = 1 -> RX1 = Coax @@ -745,7 +745,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) ice->num_total_dacs = 6; ice->num_total_adcs = 2; ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ - + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 043a938..f30aa55 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -25,7 +25,7 @@ */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -230,7 +230,7 @@ static void ak4396_write(struct snd_ice1712 *ice, unsigned int reg, snd_ice1712_gpio_set_dir(ice, AK4396_CSN|AK4396_CCLK|AK4396_CDTI); snd_ice1712_gpio_set_mask(ice, ~(AK4396_CSN|AK4396_CCLK|AK4396_CDTI)); /* latch must be low when writing */ - set_gpio_bit(ice, AK4396_CSN, 0); + set_gpio_bit(ice, AK4396_CSN, 0); block = ((AK4396_ADDR & 0x03) << 14) | (1 << 13) | ((reg & 0x1f) << 8) | (data & 0xff); ak4396_send_word(ice, block); /* REGISTER ADDRESS */ @@ -269,7 +269,7 @@ static int ak4396_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct prodigy_hifi_spec *spec = ice->spec; int i; - + for (i = 0; i < 2; i++) ucontrol->value.integer.value[i] = spec->vol[i]; @@ -282,7 +282,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem struct prodigy_hifi_spec *spec = ice->spec; int i; int change = 0; - + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { if (ucontrol->value.integer.value[i] != spec->vol[i]) { @@ -346,7 +346,7 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) { unsigned char nvol; - + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else { @@ -354,7 +354,7 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, & WM_VOL_MAX; nvol = (nvol ? (nvol + DAC_MIN) : 0) & 0xff; } - + wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x100 | nvol); } @@ -363,7 +363,7 @@ static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) { unsigned char nvol; - + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else { @@ -527,7 +527,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, change = 1; } } - mutex_unlock(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return change; } @@ -982,7 +982,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) /* These come first to reduce init pop noise */ WM_ADC_MUX, 0x0003, /* ADC mute */ /* 0x00c0 replaced by 0x0003 */ - + WM_DAC_MUTE, 0x0001, /* DAC softmute */ WM_DAC_CTRL1, 0x0000, /* DAC mute */ @@ -1049,7 +1049,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) /* to remeber the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ice->akm) + if (!ice->akm) return -ENOMEM; ice->akm_codecs = 1; @@ -1082,7 +1082,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) static unsigned short ak4396_inits[] = { AK4396_CTRL1, 0x87, /* I2S Normal Mode, 24 bit */ AK4396_CTRL2, 0x02, - AK4396_CTRL3, 0x00, + AK4396_CTRL3, 0x00, AK4396_LCH_ATT, 0x00, AK4396_RCH_ATT, 0x00, }; @@ -1103,7 +1103,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) /* to remeber the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ice->akm) + if (!ice->akm) return -ENOMEM; ice->akm_codecs = 1; @@ -1117,7 +1117,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) ak4396_write(ice, AK4396_CTRL1, 0x86); msleep(100); ak4396_write(ice, AK4396_CTRL1, 0x87); - + for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2) ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); @@ -1158,7 +1158,7 @@ static unsigned char prodigyhd2_eeprom[] __devinitdata = { }; static unsigned char fortissimo4_eeprom[] __devinitdata = { - 0x43, /* SYSCONF: clock 512, ADC, 4DACs */ + 0x43, /* SYSCONF: clock 512, ADC, 4DACs */ 0x80, /* ACLINK: I2S */ 0xfc, /* I2S: vol, 96k, 24bit, 192k */ 0xc1, /* SPDIF: out-en, out-int */ diff --git a/sound/pci/ice1712/prodigy_hifi.h b/sound/pci/ice1712/prodigy_hifi.h index a4415d4..5667d51 100644 --- a/sound/pci/ice1712/prodigy_hifi.h +++ b/sound/pci/ice1712/prodigy_hifi.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define PRODIGY_HIFI_DEVICE_DESC "{Audiotrak,Prodigy 7.1 HIFI},"\ "{Audiotrak Prodigy HD2},"\ diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index b508bb3..0cdeb85 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -19,9 +19,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -523,7 +523,7 @@ static int __devinit revo_init(struct snd_ice1712 *ice) /* second stage of initialization, analog parts and others */ ak = ice->akm = kcalloc(2, sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ak) + if (!ak) return -ENOMEM; switch (ice->eeprom.subvendor) { case VT1724_SUBDEVICE_REVOLUTION71: @@ -563,7 +563,7 @@ static int __devinit revo_init(struct snd_ice1712 *ice) ice); if (err < 0) return err; - + /* unmute all codecs */ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE); diff --git a/sound/pci/ice1712/revo.h b/sound/pci/ice1712/revo.h index a3ba425..8685ac5 100644 --- a/sound/pci/ice1712/revo.h +++ b/sound/pci/ice1712/revo.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define REVO_DEVICE_DESC \ "{MidiMan M Audio,Revolution 7.1},"\ diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c index 69673b9..0af3289 100644 --- a/sound/pci/ice1712/se.c +++ b/sound/pci/ice1712/se.c @@ -20,9 +20,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -70,7 +70,7 @@ struct se_spec { * WM8740 * A 2ch-DAC of main outputs. * It setuped as I2S mode by wire, so no way to setup from software. - * The sample-rate are automatically changed. + * The sample-rate are automatically changed. * ML/I2S (28pin) --------+ * MC/DM1 (27pin) -- 5V | * MD/DM0 (26pin) -- GND | diff --git a/sound/pci/ice1712/vt1720_mobo.c b/sound/pci/ice1712/vt1720_mobo.c index 7f9674b..0f7ba24 100644 --- a/sound/pci/ice1712/vt1720_mobo.c +++ b/sound/pci/ice1712/vt1720_mobo.c @@ -19,9 +19,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> diff --git a/sound/pci/ice1712/vt1720_mobo.h b/sound/pci/ice1712/vt1720_mobo.h index 0b1b0ee..60c90d5 100644 --- a/sound/pci/ice1712/vt1720_mobo.h +++ b/sound/pci/ice1712/vt1720_mobo.h @@ -22,7 +22,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #define VT1720_MOBO_DEVICE_DESC "{Albatron,K8X800 Pro II},"\ "{Chaintech,ZNF3-150},"\

1 0

[alsa-devel] [PATCH 10/24] sound/pci: coding style fixes: hda
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/hda/hda_codec.c | 12 ++-- sound/pci/hda/hda_codec.h | 10 ++-- sound/pci/hda/hda_generic.c | 58 +++++++++--------- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 18 +++--- sound/pci/hda/hda_proc.c | 8 +- sound/pci/hda/patch_analog.c | 22 ++++---- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 22 ++++---- sound/pci/hda/patch_realtek.c | 6 +- sound/pci/hda/patch_si3054.c | 8 +- sound/pci/hda/patch_sigmatel.c | 130 ++++++++++++++++++++-------------------- sound/pci/hda/patch_via.c | 34 +++++----- 13 files changed, 166 insertions(+), 166 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 696d77e..f58c351 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1123,7 +1123,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; - + for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; @@ -2832,7 +2832,7 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, int i, j; short seq; hda_nid_t nid; - + for (i = 0; i < num_pins; i++) { for (j = i + 1; j < num_pins; j++) { if (sequences[i] > sequences[j]) { @@ -2860,7 +2860,7 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. - * + * * The analog input pins are assigned to input_pins array. * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. @@ -2925,9 +2925,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (! assoc) + if (!assoc) continue; - if (! assoc_speaker) + if (!assoc_speaker) assoc_speaker = assoc; else if (assoc_speaker != assoc) continue; @@ -3013,7 +3013,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->speaker_outs); sort_pins_by_sequence(cfg->hp_pins, sequences_hp, cfg->hp_outs); - + /* if we have only one mic, make it AUTO_PIN_MIC */ if (!cfg->input_pins[AUTO_PIN_MIC] && cfg->input_pins[AUTO_PIN_FRONT_MIC]) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 60468f5..5dda6af 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -42,7 +42,7 @@ enum { AC_GRP_AUDIO_FUNCTION = 0x01, AC_GRP_MODEM_FUNCTION = 0x02, }; - + /* * widget types */ @@ -306,15 +306,15 @@ enum { #define AC_KNBCAP_DELTA (1<<7) /* HDMI LPCM capabilities */ -#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ +#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ #define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */ #define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */ #define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */ -#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ +#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ #define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */ #define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */ #define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */ -#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ +#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ #define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */ #define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */ #define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */ @@ -603,7 +603,7 @@ struct hda_codec_preset { const char *name; int (*patch)(struct hda_codec *codec); }; - + /* ops set by the preset patch */ struct hda_codec_ops { int (*build_controls)(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 0ca3089..1a57399 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -96,7 +96,7 @@ static void snd_hda_generic_free(struct hda_codec *codec) struct hda_gspec *spec = codec->spec; struct hda_gnode *node, *n; - if (! spec) + if (!spec) return; /* free all widgets */ list_for_each_entry_safe(node, n, &spec->nid_list, list) { @@ -132,7 +132,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid else { node->conn_list = kmalloc(sizeof(hda_nid_t) * nconns, GFP_KERNEL); - if (! node->conn_list) { + if (!node->conn_list) { snd_printk(KERN_ERR "hda-generic: cannot malloc\n"); kfree(node); return -ENOMEM; @@ -152,13 +152,13 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid if (node->wid_caps & AC_WCAP_OUT_AMP) { if (node->wid_caps & AC_WCAP_AMP_OVRD) node->amp_out_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_OUT_CAP); - if (! node->amp_out_caps) + if (!node->amp_out_caps) node->amp_out_caps = spec->def_amp_out_caps; } if (node->wid_caps & AC_WCAP_IN_AMP) { if (node->wid_caps & AC_WCAP_AMP_OVRD) node->amp_in_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_IN_CAP); - if (! node->amp_in_caps) + if (!node->amp_in_caps) node->amp_in_caps = spec->def_amp_in_caps; } list_add_tail(&node->list, &spec->nid_list); @@ -181,7 +181,7 @@ static int build_afg_tree(struct hda_codec *codec) spec->def_amp_in_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_IN_CAP); nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); - if (! nid || nodes < 0) { + if (!nid || nodes < 0) { printk(KERN_ERR "Invalid AFG subtree\n"); return -EINVAL; } @@ -301,7 +301,7 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, for (i = 0; i < node->nconns; i++) { child = hda_get_node(spec, node->conn_list[i]); - if (! child) + if (!child) continue; err = parse_output_path(codec, spec, child, dac_idx); if (err < 0) @@ -349,7 +349,7 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, if (node->type != AC_WID_PIN) continue; /* output capable? */ - if (! (node->pin_caps & AC_PINCAP_OUT)) + if (!(node->pin_caps & AC_PINCAP_OUT)) continue; if (defcfg_port_conn(node) == AC_JACK_PORT_NONE) continue; /* unconnected */ @@ -360,14 +360,14 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, continue; /* skip SPDIF */ } else { /* output as default? */ - if (! (node->pin_ctl & AC_PINCTL_OUT_EN)) + if (!(node->pin_ctl & AC_PINCTL_OUT_EN)) continue; } clear_check_flags(spec); err = parse_output_path(codec, spec, node, 0); if (err < 0) return NULL; - if (! err && spec->out_pin_node[0]) { + if (!err && spec->out_pin_node[0]) { err = parse_output_path(codec, spec, node, 1); if (err < 0) return NULL; @@ -412,18 +412,18 @@ static int parse_output(struct hda_codec *codec) /* look for the HP-out pin */ node = parse_output_jack(codec, spec, AC_JACK_HP_OUT); if (node) { - if (! spec->out_pin_node[0]) + if (!spec->out_pin_node[0]) spec->out_pin_node[0] = node; else spec->out_pin_node[1] = node; } - if (! spec->out_pin_node[0]) { + if (!spec->out_pin_node[0]) { /* no line-out or HP pins found, * then choose for the first output pin */ spec->out_pin_node[0] = parse_output_jack(codec, spec, -1); - if (! spec->out_pin_node[0]) + if (!spec->out_pin_node[0]) snd_printd("hda_generic: no proper output path found\n"); } @@ -517,7 +517,7 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, for (i = 0; i < node->nconns; i++) { struct hda_gnode *child; child = hda_get_node(spec, node->conn_list[i]); - if (! child) + if (!child) continue; err = parse_adc_sub_nodes(codec, spec, child); if (err < 0) @@ -537,7 +537,7 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, } /* input capable? */ - if (! (node->pin_caps & AC_PINCAP_IN)) + if (!(node->pin_caps & AC_PINCAP_IN)) return 0; if (defcfg_port_conn(node) == AC_JACK_PORT_NONE) @@ -554,9 +554,9 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, pinctl = AC_PINCTL_IN_EN; /* create a proper capture source label */ type = get_input_type(node, &pinctl); - if (! type) { + if (!type) { /* input as default? */ - if (! (node->pin_ctl & AC_PINCTL_IN_EN)) + if (!(node->pin_ctl & AC_PINCTL_IN_EN)) return 0; type = "Input"; } @@ -584,7 +584,7 @@ static void add_cap_src(struct hda_gspec *spec, int idx) csrc = &spec->input_mux.items[num]; buf = spec->cap_labels[num]; for (ocap = 0; ocap < num; ocap++) { - if (! strcmp(buf, spec->cap_labels[ocap])) { + if (!strcmp(buf, spec->cap_labels[ocap])) { /* same label already exists, * put the index number to be unique */ @@ -608,9 +608,9 @@ static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node) snd_printdd("AUD_IN = %x\n", adc_node->nid); clear_check_flags(spec); - // awk added - fixed no recording due to muted widget + /* awk added - fixed no recording due to muted widget */ unmute_input(codec, adc_node, 0); - + /* * check each connection of the ADC * if it reaches to a proper input PIN, add the path as the @@ -639,7 +639,7 @@ static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node) } } - if (! spec->input_mux.num_items) + if (!spec->input_mux.num_items) return 0; /* no input path found... */ snd_printdd("[Capture Source] NID=0x%x, #SRC=%d\n", adc_node->nid, spec->input_mux.num_items); @@ -830,7 +830,7 @@ static int build_input_controls(struct hda_codec *codec) .put = capture_source_put, }; - if (! adc_node || ! spec->input_mux.num_items) + if (!adc_node || !spec->input_mux.num_items) return 0; /* not found */ spec->cur_cap_src = 0; @@ -854,8 +854,8 @@ static int build_input_controls(struct hda_codec *codec) return err; /* no volume control? */ - if (! (adc_node->wid_caps & AC_WCAP_IN_AMP) || - ! (adc_node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) + if (!(adc_node->wid_caps & AC_WCAP_IN_AMP) || + !(adc_node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) return 0; for (i = 0; i < spec->input_mux.num_items; i++) { @@ -901,7 +901,7 @@ static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec, for (i = 0; i < node->nconns; i++) { struct hda_gnode *child = hda_get_node(spec, node->conn_list[i]); - if (! child) + if (!child) continue; err = parse_loopback_path(codec, spec, child, dest_node, type); if (err < 0) @@ -938,14 +938,14 @@ static int build_loopback_controls(struct hda_codec *codec) int err; const char *type; - if (! spec->out_pin_node[0]) + if (!spec->out_pin_node[0]) return 0; list_for_each_entry(node, &spec->nid_list, list) { if (node->type != AC_WID_PIN) continue; /* input capable? */ - if (! (node->pin_caps & AC_PINCAP_IN)) + if (!(node->pin_caps & AC_PINCAP_IN)) return 0; type = get_input_type(node, NULL); if (type) { @@ -957,7 +957,7 @@ static int build_loopback_controls(struct hda_codec *codec) node, type); if (err < 0) return err; - if (! err) + if (!err) continue; } } @@ -1018,7 +1018,7 @@ static int build_generic_pcms(struct hda_codec *codec) struct hda_gspec *spec = codec->spec; struct hda_pcm *info = &spec->pcm_rec; - if (! spec->dac_node[0] && ! spec->adc_node) { + if (!spec->dac_node[0] && !spec->adc_node) { snd_printd("hda_generic: no PCM found\n"); return 0; } @@ -1071,7 +1071,7 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec) struct hda_gspec *spec; int err; - if(!codec->afg) + if (!codec->afg) return 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 6e18a42..22c44fc 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -49,7 +49,7 @@ static int get_wcap_ioctl(struct hda_codec *codec, struct hda_verb_ioctl __user *arg) { u32 verb, res; - + if (get_user(verb, &arg->verb)) return -EFAULT; res = get_wcaps(codec, verb >> 24); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60cc44a..3448c42 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -31,10 +31,10 @@ * CHANGES: * * 2004.12.01 Major rewrite by tiwai, merged the work of pshou - * + * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/kernel.h> @@ -139,7 +139,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 #define ICH6_REG_WALCLK 0x30 -#define ICH6_REG_SYNC 0x34 +#define ICH6_REG_SYNC 0x34 #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 #define ICH6_REG_CORBWP 0x48 @@ -422,7 +422,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", - [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_TERA] = "HDA Teradici", }; /* @@ -554,7 +554,7 @@ static void azx_update_rirb(struct azx *chip) if (wp == chip->rirb.wp) return; chip->rirb.wp = wp; - + while (chip->rirb.rp != wp) { chip->rirb.rp++; chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES; @@ -777,7 +777,7 @@ static int azx_reset(struct azx *chip) /* * Lowlevel interface - */ + */ /* enable interrupts */ static void azx_int_enable(struct azx *chip) @@ -913,7 +913,7 @@ static void azx_init_pci(struct azx *chip) case AZX_DRIVER_ATI: /* For ATI SB450 azalia HD audio, we need to enable snoop */ update_pci_byte(chip->pci, - ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, + ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP); break; case AZX_DRIVER_NVIDIA: @@ -964,7 +964,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) spin_unlock(&chip->reg_lock); return IRQ_NONE; } - + for (i = 0; i < chip->num_streams; i++) { azx_dev = &chip->azx_dev[i]; if (status & azx_dev->sd_int_sta_mask) { @@ -1004,7 +1004,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) azx_writeb(chip, STATESTS, 0x04); #endif spin_unlock(&chip->reg_lock); - + return IRQ_HANDLED; } diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 4927c4b..5833f7e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -1,6 +1,6 @@ /* * Universal Interface for Intel High Definition Audio Codec - * + * * Generic proc interface * * Copyright (c) 2004 Takashi Iwai <tiwai(a)suse.de> @@ -99,7 +99,7 @@ static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm) pcm &= AC_SUPPCM_RATES; snd_iprintf(buffer, " rates [0x%x]:", pcm); - for (i = 0; i < ARRAY_SIZE(rates); i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (pcm & (1 << i)) snd_iprintf(buffer, " %d", rates[i]); snd_iprintf(buffer, "\n"); @@ -527,7 +527,7 @@ static void print_codec_info(struct snd_info_entry *entry, else snd_iprintf(buffer, "No Modem Function Group found\n"); - if (! codec->afg) + if (!codec->afg) return; snd_hda_power_up(codec); snd_iprintf(buffer, "Default PCM:\n"); @@ -538,7 +538,7 @@ static void print_codec_info(struct snd_info_entry *entry, print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT); nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); - if (! nid || nodes < 0) { + if (!nid || nodes < 0) { snd_iprintf(buffer, "Invalid AFG subtree\n"); snd_hda_power_down(codec); return; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0899b6b..f5b125a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -174,7 +174,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (err < 0) return err; spec->multiout.share_spdif = 1; - } + } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); if (err < 0) @@ -412,7 +412,7 @@ static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, struct ad198x_spec *spec = codec->spec; int invert = (kcontrol->private_value >> 8) & 1; if (invert) - ucontrol->value.integer.value[0] = ! spec->cur_eapd; + ucontrol->value.integer.value[0] = !spec->cur_eapd; else ucontrol->value.integer.value[0] = spec->cur_eapd; return 0; @@ -1413,7 +1413,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; - if (! ad198x_eapd_put(kcontrol, ucontrol)) + if (!ad198x_eapd_put(kcontrol, ucontrol)) return 0; /* change speaker pin appropriately */ snd_hda_codec_write(codec, 0x05, 0, @@ -2426,7 +2426,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_sequence_write(codec, ad1988_laptop_hp_on); else snd_hda_sequence_write(codec, ad1988_laptop_hp_off); -} +} #ifdef CONFIG_SND_HDA_POWER_SAVE static struct hda_amp_list ad1988_loopbacks[] = { @@ -2466,7 +2466,7 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */ - if (! knew) + if (!knew) return -ENOMEM; if (spec->kctl_alloc) { memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc); @@ -2479,7 +2479,7 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = ad1988_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); - if (! knew->name) + if (!knew->name) return -ENOMEM; knew->private_value = val; spec->num_kctl_used++; @@ -2581,7 +2581,7 @@ static int ad1988_auto_create_multi_out_ctls(struct ad198x_spec *spec, for (i = 0; i < cfg->line_outs; i++) { hda_nid_t dac = spec->multiout.dac_nids[i]; - if (! dac) + if (!dac) continue; nid = ad1988_mixer_nids[ad1988_pin_idx(cfg->line_out_pins[i])]; if (i == 2) { @@ -2631,7 +2631,7 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, int i, idx, err; char name[32]; - if (! pin) + if (!pin) return 0; idx = ad1988_pin_idx(pin); @@ -2784,7 +2784,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (! nid) + if (!nid) continue; switch (nid) { case 0x15: /* port-C */ @@ -2818,7 +2818,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) return err; if ((err = ad1988_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) return err; - if (! spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = ad1988_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || (err = ad1988_auto_create_extra_out(codec, @@ -2904,7 +2904,7 @@ static int patch_ad1988(struct hda_codec *codec) if (err < 0) { ad198x_free(codec); return err; - } else if (! err) { + } else if (!err) { printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using 6-stack mode...\n"); board_config = AD1988_6STACK; } diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 6ef57fb..473157b 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -372,7 +372,7 @@ static int cmi9880_fill_multi_dac_nids(struct hda_codec *codec, const struct aut continue; /* search for an empty channel */ for (j = 0; j < cfg->line_outs; j++) { - if (! assigned[j]) { + if (!assigned[j]) { spec->dac_nids[i] = j + 0x03; assigned[j] = 1; break; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7c1eb23..f2b16f8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -377,7 +377,7 @@ static int conexant_build_controls(struct hda_codec *codec) if (err < 0) return err; spec->multiout.share_spdif = 1; - } + } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid); if (err < 0) @@ -428,7 +428,7 @@ static int cxt_eapd_put(struct snd_kcontrol *kcontrol, eapd = !eapd; if (eapd == spec->cur_eapd) return 0; - + spec->cur_eapd = eapd; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, @@ -592,7 +592,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec) spec->hp_present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; + bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } @@ -706,7 +706,7 @@ static struct hda_verb cxt5045_init_verbs[] = { {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 }, /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ + {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ { } /* end */ }; @@ -772,7 +772,7 @@ static struct snd_kcontrol_new cxt5045_test_mixer[] = { HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT), - + /* Modes for retasking pin widgets */ CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), CXT_PIN_MODE("LINE1 pin mode", 0x12, CXT_PIN_DIR_INOUT), @@ -1004,8 +1004,8 @@ static int patch_cxt5045(struct hda_codec *codec) spec->mixers[0] = cxt5045_test_mixer; spec->init_verbs[0] = cxt5045_test_init_verbs; break; - -#endif + +#endif } switch (codec->subsystem_id >> 16) { @@ -1395,7 +1395,7 @@ static struct hda_verb cxt5047_test_init_verbs[] = { */ {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure mic1, mic2, line1 pin widgets take input from the + /* Ensure mic1, mic2, line1 pin widgets take input from the * OUT1 sum bus when acting as an output. */ {0x1a, AC_VERB_SET_CONNECT_SEL, 0}, @@ -1530,7 +1530,7 @@ static int patch_cxt5047(struct hda_codec *codec) spec->mixers[0] = cxt5047_test_mixer; spec->init_verbs[0] = cxt5047_test_init_verbs; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; -#endif +#endif } return 0; } @@ -1685,7 +1685,7 @@ static struct hda_verb cxt5051_init_verbs[] = { /* HP, Amp */ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ + /* DAC1 */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Record selector: Int mic */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, @@ -1694,7 +1694,7 @@ static struct hda_verb cxt5051_init_verbs[] = { /* SPDIF route: PCM */ {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5c80a27..80e66d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9291,7 +9291,7 @@ static struct hda_verb alc262_hippo1_unsol_verbs[] = { static struct hda_verb alc262_sony_unsol_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, /* Front Mic */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -15209,7 +15209,7 @@ static void alc662_eeepc_unsol_event(struct hda_codec *codec, unsigned int res) { if ((res >> 26) == ALC880_HP_EVENT) - alc262_hippo1_automute( codec ); + alc262_hippo1_automute(codec); if ((res >> 26) == ALC880_MIC_EVENT) alc662_eeepc_mic_automute(codec); @@ -15217,7 +15217,7 @@ static void alc662_eeepc_unsol_event(struct hda_codec *codec, static void alc662_eeepc_inithook(struct hda_codec *codec) { - alc262_hippo1_automute( codec ); + alc262_hippo1_automute(codec); alc662_eeepc_mic_automute(codec); } diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 9332b63..1148d8d 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -103,7 +103,7 @@ static int si3054_switch_get(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); u16 reg = PRIVATE_REG(kcontrol->private_value); u16 mask = PRIVATE_MASK(kcontrol->private_value); - uvalue->value.integer.value[0] = (GET_REG(codec, reg)) & mask ? 1 : 0 ; + uvalue->value.integer.value[0] = (GET_REG(codec, reg)) & mask ? 1 : 0; return 0; } @@ -128,7 +128,7 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol, .put = si3054_switch_put, \ .private_value = PRIVATE_VALUE(reg,mask), \ } - + static struct snd_kcontrol_new si3054_modem_mixer[] = { SI3054_KCONTROL("Off-hook Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_OH), @@ -233,7 +233,7 @@ static int si3054_init(struct hda_codec *codec) val = GET_REG(codec, SI3054_EXTENDED_MID); } while ((val & SI3054_MEI_READY) != SI3054_MEI_READY && wait_count--); - if((val&SI3054_MEI_READY) != SI3054_MEI_READY) { + if ((val&SI3054_MEI_READY) != SI3054_MEI_READY) { snd_printk(KERN_ERR "si3054: cannot initialize. EXT MID = %04x\n", val); /* let's pray that this is no fatal error */ /* return -EACCES; */ @@ -244,7 +244,7 @@ static int si3054_init(struct hda_codec *codec) SET_REG(codec, SI3054_MISC_AFE, 0); SET_REG(codec, SI3054_LINE_CFG1,0x200); - if((GET_REG(codec,SI3054_LINE_STATUS) & (1<<6)) == 0) { + if ((GET_REG(codec,SI3054_LINE_STATUS) & (1<<6)) == 0) { snd_printd("Link Frame Detect(FDT) is not ready (line status: %04x)\n", GET_REG(codec,SI3054_LINE_STATUS)); } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 238aebf..6c4cfe8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -405,12 +405,12 @@ static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { }; static hda_nid_t stac9200_pin_nids[8] = { - 0x08, 0x09, 0x0d, 0x0e, + 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, }; static hda_nid_t stac925x_pin_nids[8] = { - 0x07, 0x08, 0x0a, 0x0b, + 0x07, 0x08, 0x0a, 0x0b, 0x0c, 0x0d, 0x10, 0x11, }; @@ -796,23 +796,23 @@ static struct hda_verb stac925x_core_init[] = { }; static struct hda_verb stac922x_core_init[] = { - /* set master volume and direct control */ + /* set master volume and direct control */ { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; static struct hda_verb d965_core_init[] = { - /* set master volume and direct control */ + /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* unmute node 0x1b */ { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* select node 0x03 as DAC */ + /* select node 0x03 as DAC */ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, {} }; static struct hda_verb stac927x_core_init[] = { - /* set master volume and direct control */ + /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* enable analog pc beep path */ { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, @@ -820,7 +820,7 @@ static struct hda_verb stac927x_core_init[] = { }; static struct hda_verb stac9205_core_init[] = { - /* set master volume and direct control */ + /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* enable analog pc beep path */ { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, @@ -1159,7 +1159,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } - return 0; + return 0; } static unsigned int ref9200_pin_configs[8] = { @@ -1167,28 +1167,28 @@ static unsigned int ref9200_pin_configs[8] = { 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -/* +/* STAC 9200 pin configs for 102801A8 102801DE 102801E8 */ static unsigned int dell9200_d21_pin_configs[8] = { - 0x400001f0, 0x400001f1, 0x02214030, 0x01014010, + 0x400001f0, 0x400001f1, 0x02214030, 0x01014010, 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -/* +/* STAC 9200 pin configs for 102801C0 102801C1 */ static unsigned int dell9200_d22_pin_configs[8] = { - 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, + 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, 0x01813020, 0x02a19021, 0x90100140, 0x400001f2, }; -/* +/* STAC 9200 pin configs for 102801C4 (Dell Dimension E310) 102801C5 @@ -1198,12 +1198,12 @@ static unsigned int dell9200_d22_pin_configs[8] = { 102801E3 */ static unsigned int dell9200_d23_pin_configs[8] = { - 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, - 0x01813020, 0x01a19021, 0x90100140, 0x400001f2, + 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, + 0x01813020, 0x01a19021, 0x90100140, 0x400001f2, }; -/* +/* STAC 9200-32 pin configs for 102801B5 (Dell Inspiron 630m) 102801D8 (Dell Inspiron 640m) @@ -1213,20 +1213,20 @@ static unsigned int dell9200_m21_pin_configs[8] = { 0x408003fb, 0x03a11020, 0x401003fc, 0x403003fd, }; -/* +/* STAC 9200-32 pin configs for 102801C2 (Dell Latitude D620) - 102801C8 + 102801C8 102801CC (Dell Latitude D820) - 102801D4 - 102801D6 + 102801D4 + 102801D6 */ static unsigned int dell9200_m22_pin_configs[8] = { - 0x40c003fa, 0x0144131f, 0x0321121f, 0x90170310, + 0x40c003fa, 0x0144131f, 0x0321121f, 0x90170310, 0x90a70321, 0x03a11020, 0x401003fb, 0x40f000fc, }; -/* +/* STAC 9200-32 pin configs for 102801CE (Dell XPS M1710) 102801CF (Dell Precision M90) @@ -1237,15 +1237,15 @@ static unsigned int dell9200_m23_pin_configs[8] = { }; /* - STAC 9200-32 pin configs for + STAC 9200-32 pin configs for 102801C9 102801CA 102801CB (Dell Latitude 120L) 102801D3 */ static unsigned int dell9200_m24_pin_configs[8] = { - 0x40c003fa, 0x404003fb, 0x0321121f, 0x90170310, - 0x408003fc, 0x03a11020, 0x401003fd, 0x403003fe, + 0x40c003fa, 0x404003fb, 0x0321121f, 0x90170310, + 0x408003fc, 0x03a11020, 0x401003fd, 0x403003fe, }; /* @@ -1255,7 +1255,7 @@ static unsigned int dell9200_m24_pin_configs[8] = { 102801EF */ static unsigned int dell9200_m25_pin_configs[8] = { - 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310, + 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310, 0x408003fb, 0x04a11020, 0x401003fc, 0x403003fd, }; @@ -1265,7 +1265,7 @@ static unsigned int dell9200_m25_pin_configs[8] = { 102801F6 */ static unsigned int dell9200_m26_pin_configs[8] = { - 0x40c003fa, 0x404003fb, 0x0421121f, 0x90170310, + 0x40c003fa, 0x404003fb, 0x0421121f, 0x90170310, 0x408003fc, 0x04a11020, 0x401003fd, 0x403003fe, }; @@ -1601,7 +1601,7 @@ static unsigned int dell_922x_d82_pin_configs[10] = { */ static unsigned int dell_922x_m81_pin_configs[10] = { 0x0321101f, 0x01112024, 0x01111222, 0x91174220, - 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, + 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, 0x40C003f1, 0x405003f0, }; @@ -1610,9 +1610,9 @@ static unsigned int dell_922x_m81_pin_configs[10] = { 102801D7 (Dell XPS M1210) */ static unsigned int dell_922x_m82_pin_configs[10] = { - 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, - 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, - 0x508003f3, 0x405003f4, + 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, + 0x508003f3, 0x405003f4, }; static unsigned int d945gtp3_pin_configs[10] = { @@ -1682,9 +1682,9 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs, [STAC_ECS_202] = ecs202_pin_configs, [STAC_922X_DELL_D81] = dell_922x_d81_pin_configs, - [STAC_922X_DELL_D82] = dell_922x_d82_pin_configs, + [STAC_922X_DELL_D82] = dell_922x_d82_pin_configs, [STAC_922X_DELL_M81] = dell_922x_m81_pin_configs, - [STAC_922X_DELL_M82] = dell_922x_m82_pin_configs, + [STAC_922X_DELL_M82] = dell_922x_m82_pin_configs, }; static const char *stac922x_models[STAC_922X_MODELS] = { @@ -1823,7 +1823,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { static unsigned int ref927x_pin_configs[14] = { 0x02214020, 0x02a19080, 0x0181304e, 0x01014010, - 0x01a19040, 0x01011012, 0x01016011, 0x0101201f, + 0x01a19040, 0x01011012, 0x01016011, 0x0101201f, 0x183301f0, 0x18a001f0, 0x18a001f0, 0x01442070, 0x01c42190, 0x40000100, }; @@ -2015,25 +2015,25 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; - - if (! spec->bios_pin_configs) { + + if (!spec->bios_pin_configs) { spec->bios_pin_configs = kcalloc(spec->num_pins, sizeof(*spec->bios_pin_configs), GFP_KERNEL); - if (! spec->bios_pin_configs) + if (!spec->bios_pin_configs) return -ENOMEM; } - + for (i = 0; i < spec->num_pins; i++) { hda_nid_t nid = spec->pin_nids[i]; unsigned int pin_cfg; - - pin_cfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0x00); + + pin_cfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0x00); snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", nid, pin_cfg); spec->bios_pin_configs[i] = pin_cfg; } - + return 0; } @@ -2055,7 +2055,7 @@ static void stac92xx_set_config_reg(struct hda_codec *codec, pin_config >> 24); i = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_CONFIG_DEFAULT, - 0x00); + 0x00); snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", pin_nid, i); } @@ -2442,7 +2442,7 @@ static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */ - if (! knew) + if (!knew) return -ENOMEM; if (spec->kctl_alloc) { memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc); @@ -2456,7 +2456,7 @@ static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, *knew = stac92xx_control_templates[type]; knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); - if (! knew->name) + if (!knew->name) return -ENOMEM; knew->private_value = val; spec->num_kctl_used++; @@ -2477,7 +2477,7 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf struct sigmatel_spec *spec = codec->spec; unsigned int wcaps, wtype; int i, num_dacs = 0; - + /* use the wcaps cache to count all DACs available for line-outs */ for (i = 0; i < codec->num_nodes; i++) { wcaps = codec->wcaps[i]; @@ -2488,7 +2488,7 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf } snd_printdd("%s: total dac count=%d\n", __func__, num_dacs); - + switch (cfg->line_outs) { case 3: /* add line-in as side */ @@ -2538,7 +2538,7 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) { int i; - + for (i = 0; i < spec->multiout.num_dacs; i++) { if (spec->multiout.dac_nids[i] == nid) return 1; @@ -2558,10 +2558,10 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - int i, j, conn_len = 0; + int i, j, conn_len = 0; hda_nid_t nid, conn[HDA_MAX_CONNECTIONS]; unsigned int wcaps, wtype; - + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; conn_len = snd_hda_get_connections(codec, nid, conn, @@ -2772,7 +2772,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; if (check_in_dac_nids(spec, nid)) nid = 0; - if (! nid) + if (!nid) continue; add_spec_dacs(spec, nid); } @@ -2781,7 +2781,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; if (check_in_dac_nids(spec, nid)) nid = 0; - if (! nid) + if (!nid) continue; add_spec_dacs(spec, nid); } @@ -2790,7 +2790,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; if (check_in_dac_nids(spec, nid)) nid = 0; - if (! nid) + if (!nid) continue; add_spec_dacs(spec, nid); } @@ -3082,7 +3082,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out &spec->autocfg, spec->dmic_nids)) < 0) return err; - if (! spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) return 0; /* can't find valid pin config */ /* If we have no real line-out pin and multiple hp-outs, HPs should @@ -3258,7 +3258,7 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin = cfg->hp_pins[0]; unsigned int wid_caps; - if (! pin) + if (!pin) return 0; wid_caps = get_wcaps(codec, pin); @@ -3503,7 +3503,7 @@ static void stac92xx_free(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; int i; - if (! spec) + if (!spec) return; if (spec->kctl_alloc) { @@ -3544,7 +3544,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, direction bits first */ if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl | flag); @@ -3628,7 +3628,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) } if (!spec->hp_switch && cfg->hp_outs > 1 && presence) stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); -} +} static void stac92xx_pin_sense(struct hda_codec *codec, int idx) { @@ -3779,7 +3779,7 @@ static int patch_stac925x(struct hda_codec *codec) stac925x_cfg_tbl); again: if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," "using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); if (err < 0) { @@ -4309,7 +4309,7 @@ static int patch_stac922x(struct hda_codec *codec) spec->mixer = stac922x_mixer; spec->multiout.dac_nids = spec->dac_nids; - + err = stac92xx_parse_auto_config(codec, 0x08, 0x09); if (!err) { if (spec->board_config < 0) { @@ -4510,7 +4510,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->multiout.dac_nids = spec->dac_nids; - + switch (spec->board_config){ case STAC_9205_DELL_M43: /* Enable SPDIF in/out */ @@ -4700,7 +4700,7 @@ static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); } -} +} static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -4725,7 +4725,7 @@ static struct hda_codec_ops stac9872_vaio_patch_ops = { enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */ CXD9872RD_VAIO, /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */ - STAC9872AK_VAIO, + STAC9872AK_VAIO, /* Unknown. id=0x83847661 and subsys=0x104D1200. */ STAC9872K_VAIO, /* AR Series. id=0x83847664 and subsys=104D1300 */ @@ -4757,7 +4757,7 @@ static int patch_stac9872(struct hda_codec *codec) if (board_config < 0) /* unknown config, let generic-parser do its job... */ return snd_hda_parse_generic_codec(codec); - + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; @@ -4780,7 +4780,7 @@ static int patch_stac9872(struct hda_codec *codec) spec->mux_nids = vaio_mux_nids; codec->patch_ops = stac9872_vaio_patch_ops; break; - + case CXD9872AKD_VAIO: spec->mixer = vaio_ar_mixer; spec->init = vaio_ar_init; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e7e4352..4597ec9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -344,7 +344,7 @@ static struct hda_verb vt1708_volume_init_verbs[] = { {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* Setup default input to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 Output enable */ @@ -644,8 +644,8 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -683,7 +683,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, if (!nid) continue; - + if (i != AUTO_SEQ_FRONT) nid_vol = 0x1b - i + 1; @@ -727,7 +727,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -806,7 +806,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -874,7 +874,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; - spec->init_verbs = vt1708_volume_init_verbs; + spec->init_verbs = vt1708_volume_init_verbs; spec->input_mux = &spec->private_imux; @@ -913,7 +913,7 @@ static int patch_vt1708(struct hda_codec *codec) "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -925,7 +925,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -1104,7 +1104,7 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -1143,7 +1143,7 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; if (i == AUTO_SEQ_CENLFE) { @@ -1186,7 +1186,7 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -1281,7 +1281,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -1375,7 +1375,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) "Using genenic mode...\n"); } - spec->init_verbs = vt1709_10ch_volume_init_verbs; + spec->init_verbs = vt1709_10ch_volume_init_verbs; spec->stream_name_analog = "VT1709 Analog"; spec->stream_analog_playback = &vt1709_10ch_pcm_analog_playback; @@ -1385,7 +1385,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -1466,7 +1466,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) "Using genenic mode...\n"); } - spec->init_verbs = vt1709_6ch_volume_init_verbs; + spec->init_verbs = vt1709_6ch_volume_init_verbs; spec->stream_name_analog = "VT1709 Analog"; spec->stream_analog_playback = &vt1709_6ch_pcm_analog_playback; @@ -1476,7 +1476,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);

1 0

[alsa-devel] [PATCH 08/24] sound/pci: coding style fixes: emu10k1
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/emu10k1/emu10k1.c | 8 +- sound/pci/emu10k1/emu10k1_callback.c | 28 ++-- sound/pci/emu10k1/emu10k1_main.c | 266 +++++++++++++++++----------------- sound/pci/emu10k1/emu10k1_patch.c | 14 +- sound/pci/emu10k1/emu10k1_synth.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 156 ++++++++++---------- sound/pci/emu10k1/emufx.c | 148 ++++++++++---------- sound/pci/emu10k1/emumixer.c | 130 ++++++++-------- sound/pci/emu10k1/emumpu401.c | 2 +- sound/pci/emu10k1/emupcm.c | 56 ++++---- sound/pci/emu10k1/emuproc.c | 80 +++++----- sound/pci/emu10k1/io.c | 32 ++-- sound/pci/emu10k1/irq.c | 18 +- sound/pci/emu10k1/memory.c | 8 +- sound/pci/emu10k1/p16v.c | 132 +++++++++--------- sound/pci/emu10k1/p16v.h | 14 +- sound/pci/emu10k1/p17v.h | 6 +- sound/pci/emu10k1/voice.c | 32 ++-- 18 files changed, 566 insertions(+), 566 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 8354c1a..094a874 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -20,7 +20,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - * + * */ #include <linux/init.h> @@ -134,7 +134,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, if ((err = snd_emu10k1_pcm_efx(emu, 2, NULL)) < 0) goto error; /* This stores the periods table. */ - if (emu->card_capabilities->ca0151_chip) { /* P16V */ + if (emu->card_capabilities->ca0151_chip) { /* P16V */ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &emu->p16v_buffer)) < 0) goto error; @@ -142,7 +142,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, if ((err = snd_emu10k1_mixer(emu, 0, 3)) < 0) goto error; - + if ((err = snd_emu10k1_timer(emu, 0)) < 0) goto error; @@ -176,7 +176,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, arg->max_voices = max_synth_voices[dev]; } #endif - + strcpy(card->driver, emu->card_capabilities->driver); strcpy(card->shortname, emu->card_capabilities->name); snprintf(card->longname, sizeof(card->longname), diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 0e649dc..7bc7fce 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -103,7 +103,7 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw) int ch; vp = &emu->voices[best[i].voice]; if ((ch = vp->ch) < 0) { - //printk("synth_get_voice: ch < 0 (%d) ??", i); + /* printk("synth_get_voice: ch < 0 (%d) ??", i); */ continue; } vp->emu->num_voices--; @@ -128,7 +128,7 @@ release_voice(struct snd_emux_voice *vp) { int dcysusv; struct snd_emu10k1 *hw; - + hw = vp->hw; dcysusv = 0x8000 | (unsigned char)vp->reg.parm.modrelease; snd_emu10k1_ptr_write(hw, DCYSUSM, vp->ch, dcysusv); @@ -144,7 +144,7 @@ static void terminate_voice(struct snd_emux_voice *vp) { struct snd_emu10k1 *hw; - + if (snd_BUG_ON(!vp)) return; hw = vp->hw; @@ -164,7 +164,7 @@ static void free_voice(struct snd_emux_voice *vp) { struct snd_emu10k1 *hw; - + hw = vp->hw; /* FIXME: emu10k1_synth is broken. */ /* This can get called with hw == 0 */ @@ -173,7 +173,7 @@ free_voice(struct snd_emux_voice *vp) if (hw && (vp->ch >= 0)) { snd_emu10k1_ptr_write(hw, IFATN, vp->ch, 0xff00); snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0x807f | DCYSUSV_CHANNELENABLE_MASK); - // snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0); + /* snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0); */ snd_emu10k1_ptr_write(hw, VTFT, vp->ch, 0xffff); snd_emu10k1_ptr_write(hw, CVCF, vp->ch, 0xffff); snd_emu10k1_voice_free(hw, &hw->voices[vp->ch]); @@ -190,7 +190,7 @@ static void update_voice(struct snd_emux_voice *vp, int update) { struct snd_emu10k1 *hw; - + hw = vp->hw; if (update & SNDRV_EMUX_UPDATE_VOLUME) snd_emu10k1_ptr_write(hw, IFATN_ATTENUATION, vp->ch, vp->avol); @@ -250,7 +250,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, bp = best + V_RELEASED; #if 1 val = snd_emu10k1_ptr_read(hw, CVCF_CURRENTVOL, vp->ch); - if (! val) + if (!val) bp = best + V_OFF; #endif } @@ -323,7 +323,7 @@ start_voice(struct snd_emux_voice *vp) struct snd_midi_channel *chan; struct snd_emu10k1 *hw; struct snd_emu10k1_memblk *emem; - + hw = vp->hw; ch = vp->ch; if (snd_BUG_ON(ch < 0)) @@ -335,7 +335,7 @@ start_voice(struct snd_emux_voice *vp) return -EINVAL; emem->map_locked++; if (snd_emu10k1_memblk_map(hw, emem) < 0) { - // printk("emu: cannot map!\n"); + /* printk("emu: cannot map!\n"); */ return -ENOMEM; } mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1; @@ -347,11 +347,11 @@ start_voice(struct snd_emux_voice *vp) /* set channel routing */ /* A = left(0), B = right(1), C = reverb(c), D = chorus(d) */ if (hw->audigy) { - temp = FXBUS_MIDI_LEFT | (FXBUS_MIDI_RIGHT << 8) | + temp = FXBUS_MIDI_LEFT | (FXBUS_MIDI_RIGHT << 8) | (FXBUS_MIDI_REVERB << 16) | (FXBUS_MIDI_CHORUS << 24); snd_emu10k1_ptr_write(hw, A_FXRT1, ch, temp); } else { - temp = (FXBUS_MIDI_LEFT << 16) | (FXBUS_MIDI_RIGHT << 20) | + temp = (FXBUS_MIDI_LEFT << 16) | (FXBUS_MIDI_RIGHT << 20) | (FXBUS_MIDI_REVERB << 24) | (FXBUS_MIDI_CHORUS << 28); snd_emu10k1_ptr_write(hw, FXRT, ch, temp); } @@ -437,7 +437,7 @@ start_voice(struct snd_emux_voice *vp) temp = ((unsigned int)hw->silent_page.addr << 1) | MAP_PTI_MASK; snd_emu10k1_ptr_write(hw, MAPA, ch, temp); snd_emu10k1_ptr_write(hw, MAPB, ch, temp); - + /* fill cache */ val -= 4; val <<= 25; @@ -476,11 +476,11 @@ trigger_voice(struct snd_emux_voice *vp) unsigned int temp, ptarget; struct snd_emu10k1 *hw; struct snd_emu10k1_memblk *emem; - + hw = vp->hw; emem = (struct snd_emu10k1_memblk *)vp->block; - if (! emem || emem->mapped_page < 0) + if (!emem || emem->mapped_page < 0) return; /* not mapped */ #if 0 diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2f283ea..ab57586 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -153,7 +153,7 @@ static unsigned int i2c_adc_init[][2] = { { 0x14, 0xa6 }, /* Limiter control */ { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for Audigy 2 ZS Notebook */ }; - + static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) { unsigned int silent_page; @@ -197,12 +197,12 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Hacks for Alice3 to work independent of haP16V driver */ - //Setup SRCMulti_I2S SamplingRate + /* Setup SRCMulti_I2S SamplingRate */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp &= 0xfffff1ff; tmp |= (0x2<<9); snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); - + /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14); /* Setup SRCMulti Input Audio Enable */ @@ -217,7 +217,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ /* Hacks for Alice3 to work independent of haP16V driver */ snd_printk(KERN_INFO "Audigy2 value: Special config.\n"); - //Setup SRCMulti_I2S SamplingRate + /* Setup SRCMulti_I2S SamplingRate */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp &= 0xfffff1ff; tmp |= (0x2<<9); @@ -276,7 +276,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) } } - + snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr); snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */ snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */ @@ -313,7 +313,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) (emu->model == 0x21 && emu->revision < 6)) outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE, emu->port + HCFG); else - // With on-chip joystick + /* With on-chip joystick */ outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); if (enable_ir) { /* enable IR for SB Live */ @@ -337,7 +337,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(reg, emu->port + HCFG); } } - + if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ } else if (emu->card_capabilities->i2c_adc) { @@ -364,7 +364,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); - + if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Unmute Analog now. Set GPO6 to 1 for Apollo. * This has to be done after init ALice3 I2SOut beyond 48KHz. @@ -378,12 +378,12 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG); } } - + #if 0 { unsigned int tmp; /* FIXME: the following routine disables LiveDrive-II !! */ - // TOSLink detection + /* TOSLink detection */ emu->tos_link = 0; tmp = inl(emu->port + HCFG); if (tmp & (HCFG_GPINPUT0 | HCFG_GPINPUT1)) { @@ -495,7 +495,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) #define EC_LAST_PROMFILE_ADDR 0x2f -#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The +#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The * can be up to 30 characters in length * and is stored as a NULL-terminated * ASCII string. Any unused bytes must be @@ -503,8 +503,8 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) #define EC_CHECKSUM_ADDR 0x3f /* Location at which checksum is stored */ -/* Most of this stuff is pretty self-evident. According to the hardware - * dudes, we need to leave the ADCCAL bit low in order to avoid a DC +/* Most of this stuff is pretty self-evident. According to the hardware + * dudes, we need to leave the ADCCAL bit low in order to avoid a DC * offset problem. Weird. */ #define EC_RAW_RUN_MODE (EC_DACMUTEN | EC_ADCRSTN | EC_TRIM_MUTEN | \ @@ -574,7 +574,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, for (bit = (1 << 15); bit; bit >>= 1) { unsigned int value; - + value = emu->ecard_ctrl & ~(EC_TRIM_CSN | EC_TRIM_SDATA); if (gain & bit) @@ -598,7 +598,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) EC_SPDIF0_SELECT(EC_DEFAULT_SPDIF0_SEL) | EC_SPDIF1_SELECT(EC_DEFAULT_SPDIF1_SEL); - /* Step 0: Set the codec type in the hardware control register + /* Step 0: Set the codec type in the hardware control register * and enable audio output */ hc_value = inl(emu->port + HCFG); outl(hc_value | HCFG_AUDIOENABLE | HCFG_CODECFORMAT_I2S, emu->port + HCFG); @@ -685,13 +685,13 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */ write_post = inl(emu->port + A_IOCFG); udelay(100); /* Allow FPGA memory to clean */ - for(n = 0; n < fw_entry->size; n++) { - value=fw_entry->data[n]; - for(i = 0; i < 8; i++) { + for (n = 0; n < fw_entry->size; n++) { + value=fw_entry->data[n]; + for (i = 0; i < 8; i++) { reg = 0x80; if (value & 0x1) reg = reg | 0x20; - value = value >> 1; + value = value >> 1; outl(reg, emu->port + A_IOCFG); write_post = inl(emu->port + A_IOCFG); outl(reg | 0x40, emu->port + A_IOCFG); @@ -719,13 +719,13 @@ static int emu1010_firmware_thread(void *data) msleep_interruptible(1000); if (kthread_should_stop()) break; - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */ + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); /* IRQ Status */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK); if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { @@ -743,11 +743,11 @@ static int emu1010_firmware_thread(void *data) } } - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ @@ -755,14 +755,14 @@ static int emu1010_firmware_thread(void *data) continue; } snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); /* Unmute all. Default is muted after a firmware load */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); @@ -818,7 +818,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) * Lock Tank Memory Cache, * Mute all codecs. */ - outl(0x0005a004, emu->port + HCFG); + outl(0x0005a004, emu->port + HCFG); /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, * Mute all codecs. */ @@ -829,18 +829,18 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); snd_printdd("reg1=0x%x\n",reg); if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02 ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02); } - snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); snd_printdd("reg2=0x%x\n",reg); if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ @@ -876,7 +876,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); @@ -884,17 +884,17 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2 ); + snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); snd_printk("Hana ver:%d.%d\n",tmp ,tmp2); /* Enable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp); /* Optical -> ADAT I/O */ /* 0 : SPDIF * 1 : ADAT @@ -904,41 +904,41 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) tmp = 0; tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp ); - snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp); /* Set no attenuation on Audio Dock pads. */ - snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00); emu->emu1010.adc_pads = 0x00; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); /* Unmute Audio dock DACs, Headphone source DAC-4. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); - snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); + snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp); /* DAC PADs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f ); + snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f); emu->emu1010.dac_pads = 0x0f; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); /* SPDIF Format. Set Consumer mode, 24bit, copy enable */ - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* MIDI routing */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* Unknown. */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); - /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); // IRQ Enable: All on */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); + /* snd_emu1010_fpga_write(emu, 0x09, 0x0f); IRQ Enable: All on */ /* IRQ Enable: All off */ - snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); snd_printk(KERN_INFO "emu1010: Card options3=0x%x\n",reg); /* Default WCLK set to 48kHz. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00); /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); - //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); + /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ /* Audio Dock LEDs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); #if 0 /* For 96kHz */ @@ -992,7 +992,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) * Defaults only, users will set their own values anyways, let's * just copy/paste. */ - + snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); snd_emu1010_fpga_link_dst_src_write(emu, @@ -1037,19 +1037,19 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_F, EMU_SRC_HAMOA_ADC_LEFT2); #endif - for (i = 0;i < 0x20; i++ ) { + for (i = 0; i < 0x20; i++) { /* AudioDock Elink <- Silence */ snd_emu1010_fpga_link_dst_src_write(emu, 0x0100+i, EMU_SRC_SILENCE); } - for (i = 0;i < 4; i++) { + for (i = 0; i < 4; i++) { /* Hana SPDIF Out <- Silence */ snd_emu1010_fpga_link_dst_src_write(emu, 0x0200+i, EMU_SRC_SILENCE); } - for (i = 0;i < 7; i++) { + for (i = 0; i < 7; i++) { /* Hamoa DAC <- Silence */ snd_emu1010_fpga_link_dst_src_write(emu, 0x0300+i, EMU_SRC_SILENCE); } - for (i = 0;i < 7; i++) { + for (i = 0; i < 7; i++) { /* Hana ADAT Out <- Silence */ snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HANA_ADAT + i, EMU_SRC_SILENCE); } @@ -1065,30 +1065,30 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1); - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01 ); // Unmute all + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01); /* Unmute all */ + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); - /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, * Lock Sound Memory Cache, Lock Tank Memory Cache, * Mute all codecs. */ - outl(0x0000a000, emu->port + HCFG); + outl(0x0000a000, emu->port + HCFG); /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, * Lock Sound Memory Cache, Lock Tank Memory Cache, * Un-Mute all codecs. */ outl(0x0000a001, emu->port + HCFG); - + /* Initial boot complete. Now patches */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ /* Start Micro/Audio Dock firmware loader thread */ if (!emu->emu1010.firmware_thread) { @@ -1218,20 +1218,20 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) emu->emu1010.output_source[23] = 28; } /* TEMP: Select SPDIF in/out */ - //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ + /* snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); Output spdif */ /* TEMP: Select 48kHz SPDIF out */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */ /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); - //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); + /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ emu->emu1010.internal_clock = 1; /* 48000 */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);/* Set LEDs on Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */ - //snd_emu1010_fpga_write(emu, 0x7, 0x0); /* Mute all */ - //snd_emu1010_fpga_write(emu, 0x7, 0x1); /* Unmute all */ - //snd_emu1010_fpga_write(emu, 0xe, 0x12); /* Set LEDs on Audio Dock */ + /* snd_emu1010_fpga_write(emu, 0x7, 0x0); Mute all */ + /* snd_emu1010_fpga_write(emu, 0x7, 0x1); Unmute all */ + /* snd_emu1010_fpga_write(emu, 0xe, 0x12); Set LEDs on Audio Dock */ return 0; } @@ -1253,7 +1253,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); @@ -1278,7 +1278,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) #endif if (emu->port) pci_release_regions(emu->pci); - if (emu->card_capabilities->ca0151_chip) /* P16V */ + if (emu->card_capabilities->ca0151_chip) /* P16V */ snd_p16v_free(emu); pci_disable_device(emu->pci); kfree(emu); @@ -1301,7 +1301,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: None */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, - .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", + .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1346,7 +1346,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: None */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, - .driver = "Audigy2", .name = "Audigy 4 [SB0610]", + .driver = "Audigy2", .name = "Audigy 4 [SB0610]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1359,7 +1359,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * Digital output working. (AC3 not checked, only PCM) * Audio Mic/Line inputs working. * Digital input not tested. - */ + */ /* DSP: Tina2 * DAC: Wolfson WM8768/WM8568 * ADC: Wolfson WM8775 @@ -1386,7 +1386,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", + .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1396,7 +1396,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1} , /* Tested by James(a)superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, - .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", + .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1405,7 +1405,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu_model = EMU_MODEL_EMU1616}, /* Tested by James(a)superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, - .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1436,15 +1436,15 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* Audigy4 (Not PRO) SB0610 */ - {.vendor = 0x1102, .device = 0x0008, - .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", + {.vendor = 0x1102, .device = 0x0008, + .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, .ac97_chip = 1} , /* Tested by James(a)superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, - .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", + .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1457,7 +1457,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * Just like 0x20021102 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20061102, - .driver = "Audigy2", .name = "Audigy 2 [SB0350b]", + .driver = "Audigy2", .name = "Audigy 2 [SB0350b]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1466,7 +1466,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, - .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", + .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1475,7 +1475,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", + .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1492,7 +1492,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: Yes */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102, - .driver = "Audigy2", .name = "Audigy 2 [SB0240]", + .driver = "Audigy2", .name = "Audigy 2 [SB0240]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1502,7 +1502,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, - .driver = "Audigy2", .name = "Audigy 2 EX [1005]", + .driver = "Audigy2", .name = "Audigy 2 EX [1005]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1521,7 +1521,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, - .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]", + .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1540,44 +1540,44 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .driver = "Audigy", .name = "Audigy 1 [SB0090]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102, - .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", + .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .driver = "Audigy", .name = "Audigy 1 [SB0090]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, - .driver = "Audigy", .name = "Audigy 1 [Unknown]", + .driver = "Audigy", .name = "Audigy 1 [Unknown]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102, - .driver = "EMU10K1", .name = "SBLive! [SB0105]", + .driver = "EMU10K1", .name = "SBLive! [SB0105]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", + .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", + .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, @@ -1585,26 +1585,26 @@ static struct snd_emu_chip_details emu_chip_details[] = { /* Tested by ALSA bug#1680 26th December 2005 */ /* note: It really has SB0220 written on the card. */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80661102, - .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]", + .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by Thomas Zehetbauer 27th Aug 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102, - .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102, - .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, - .driver = "EMU10K1", .name = "SB Live 5.1", + .driver = "EMU10K1", .name = "SB Live 5.1", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, @@ -1619,78 +1619,78 @@ static struct snd_emu_chip_details emu_chip_details[] = { */ .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, - .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", + .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4831]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4831]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by James(a)superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102, - .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", + .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, - .driver = "EMU10K1", .name = "E-mu APS [4001]", + .driver = "EMU10K1", .name = "E-mu APS [4001]", .id = "APS", .emu10k1_chip = 1, .ecard = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102, - .driver = "EMU10K1", .name = "SBLive! [CT4620]", + .driver = "EMU10K1", .name = "SBLive! [CT4620]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", + .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, - .driver = "EMU10K1", .name = "SB Live [Unknown]", + .driver = "EMU10K1", .name = "SB Live [Unknown]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, @@ -1715,7 +1715,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_emu10k1_dev_free, }; - + *remu = NULL; /* enable PCI device */ @@ -1751,11 +1751,11 @@ int __devinit snd_emu10k1_create(struct snd_card *card, for (c = emu_chip_details; c->vendor; c++) { if (c->vendor == pci->vendor && c->device == pci->device) { if (subsystem) { - if (c->subsystem && (c->subsystem == subsystem) ) { + if (c->subsystem && (c->subsystem == subsystem)) { break; } else continue; } else { - if (c->subsystem && (c->subsystem != emu->serial) ) + if (c->subsystem && (c->subsystem != emu->serial)) continue; if (c->revision && c->revision != emu->revision) continue; @@ -1772,13 +1772,13 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->card_capabilities = c; if (c->subsystem && !subsystem) snd_printdd("Sound card name=%s\n", c->name); - else if (subsystem) + else if (subsystem) snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n", c->name, pci->vendor, pci->device, emu->serial, c->subsystem); - else + else snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n", c->name, pci->vendor, pci->device, emu->serial); - + if (!*card->id && c->id) { int i, n = 0; strlcpy(card->id, c->id, sizeof(card->id)); @@ -1913,7 +1913,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, snd_emu10k1_synth_alloc(emu, 4096); if (emu->reserved_page) emu->reserved_page->map_locked = 1; - + /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, PAGE_SIZE); silent_page = emu->silent_page.addr << 1; @@ -1978,7 +1978,7 @@ static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu) if (emu->audigy) size += ARRAY_SIZE(saved_regs_audigy); emu->saved_ptr = vmalloc(4 * NUM_G * size); - if (! emu->saved_ptr) + if (!emu->saved_ptr) return -ENOMEM; if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) return -ENOMEM; diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index e10f027..8e59bba 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -88,7 +88,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* try to allocate a memory block */ blocksize = truesize; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) blocksize *= 2; sp->block = snd_emu10k1_synth_alloc(emu, blocksize); if (sp->block == NULL) { @@ -102,7 +102,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* write blank samples at head */ offset = 0; size = BLANK_HEAD_SIZE; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; if (offset + size > blocksize) return -EINVAL; @@ -111,7 +111,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* copy start->loopend */ size = loopend; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; if (offset + size > blocksize) return -EINVAL; @@ -127,7 +127,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* handle reverse (or bidirectional) loop */ if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { /* copy loop in reverse */ - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { int woffset; unsigned short *wblock = (unsigned short*)block; woffset = offset / 2; @@ -160,7 +160,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = sp->v.size - loopend; if (size < 0) return -EINVAL; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { snd_emu10k1_synth_free(emu, sp->block); @@ -184,7 +184,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, #if 0 /* not supported yet */ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) { /* unsigned -> signed */ - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { unsigned short *wblock = (unsigned short*)block; for (i = 0; i < truesize; i++) wblock[i] ^= 0x8000; @@ -197,7 +197,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* recalculate offset */ start_addr = BLANK_HEAD_SIZE * 2; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) + if (!(sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) start_addr >>= 1; sp->v.start += start_addr; sp->v.end += start_addr; diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c index ad7b714..bedb7dd 100644 --- a/sound/pci/emu10k1/emu10k1_synth.c +++ b/sound/pci/emu10k1/emu10k1_synth.c @@ -105,7 +105,7 @@ static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) static int __init alsa_emu10k1_synth_init(void) { - + static struct snd_seq_dev_ops ops = { snd_emu10k1_synth_new_device, snd_emu10k1_synth_delete_device, diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 5ff4dbb..0c02d87 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -107,7 +107,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); /* Emu10k1x pointer-offset register set, accessed through the PTR and DATA registers */ /********************************************************************************************************/ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ - /* One list entry: 4 bytes for DMA address, + /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. * One list entry is 8 bytes long. * One list entry for each period in the buffer. @@ -186,7 +186,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); * - channel 1 is the rear channel * - channel 2 is the center/lfe chanel * Volume is controlled by the AC97 for the front and rear channels by - * the PCM Playback Volume, Sigmatel Surround Playback Volume and + * the PCM Playback Volume, Sigmatel Surround Playback Volume and * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects * the front/rear channel mixing in the REAR OUT jack. When using the * 4-Speaker Stereo, both front and rear channels will be mixed in the @@ -199,7 +199,7 @@ struct emu10k1x_voice { struct emu10k1x *emu; int number; int use; - + struct emu10k1x_pcm *epcm; }; @@ -246,7 +246,7 @@ struct emu10k1x { struct emu10k1x_voice voices[3]; struct emu10k1x_voice capture_voice; - u32 spdif_bits[3]; // SPDIF out setup + u32 spdif_bits[3]; /* SPDIF out setup */ struct snd_dma_buffer dma_buffer; @@ -255,7 +255,7 @@ struct emu10k1x { /* hardware definition */ static struct snd_pcm_hardware snd_emu10k1x_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), @@ -274,7 +274,7 @@ static struct snd_pcm_hardware snd_emu10k1x_playback_hw = { }; static struct snd_pcm_hardware snd_emu10k1x_capture_hw = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), @@ -292,13 +292,13 @@ static struct snd_pcm_hardware snd_emu10k1x_capture_hw = { .fifo_size = 0, }; -static unsigned int snd_emu10k1x_ptr_read(struct emu10k1x * emu, - unsigned int reg, +static unsigned int snd_emu10k1x_ptr_read(struct emu10k1x * emu, + unsigned int reg, unsigned int chn) { unsigned long flags; unsigned int regptr, val; - + regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); @@ -308,9 +308,9 @@ static unsigned int snd_emu10k1x_ptr_read(struct emu10k1x * emu, return val; } -static void snd_emu10k1x_ptr_write(struct emu10k1x *emu, - unsigned int reg, - unsigned int chn, +static void snd_emu10k1x_ptr_write(struct emu10k1x *emu, + unsigned int reg, + unsigned int chn, unsigned int data) { unsigned int regptr; @@ -396,10 +396,10 @@ static int snd_emu10k1x_playback_open(struct snd_pcm_substream *substream) return -ENOMEM; epcm->emu = chip; epcm->substream = substream; - + runtime->private_data = epcm; runtime->private_free = snd_emu10k1x_pcm_free_substream; - + runtime->hw = snd_emu10k1x_playback_hw; return 0; @@ -418,7 +418,7 @@ static int snd_emu10k1x_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct emu10k1x_pcm *epcm = runtime->private_data; - if (! epcm->voice) { + if (!epcm->voice) { epcm->voice = &epcm->emu->voices[substream->pcm->device]; epcm->voice->use = 1; epcm->voice->epcm = epcm; @@ -436,7 +436,7 @@ static int snd_emu10k1x_pcm_hw_free(struct snd_pcm_substream *substream) if (runtime->private_data == NULL) return 0; - + epcm = runtime->private_data; if (epcm->voice) { @@ -458,8 +458,8 @@ static int snd_emu10k1x_pcm_prepare(struct snd_pcm_substream *substream) u32 *table_base = (u32 *)(emu->dma_buffer.area+1024*voice); u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); int i; - - for(i = 0; i < runtime->periods; i++) { + + for (i = 0; i < runtime->periods; i++) { *table_base++=runtime->dma_addr+(i*period_size_bytes); *table_base++=period_size_bytes<<16; } @@ -491,7 +491,7 @@ static int snd_emu10k1x_pcm_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if(runtime->periods == 2) + if (runtime->periods == 2) snd_emu10k1x_intr_enable(emu, (INTE_CH_0_LOOP | INTE_CH_0_HALF_LOOP) << channel); else snd_emu10k1x_intr_enable(emu, INTE_CH_0_LOOP << channel); @@ -527,10 +527,10 @@ snd_emu10k1x_pcm_pointer(struct snd_pcm_substream *substream) ptr1 = snd_emu10k1x_ptr_read(emu, PLAYBACK_POINTER, channel); ptr4 = snd_emu10k1x_ptr_read(emu, PLAYBACK_LIST_PTR, channel); - if(ptr4 == 0 && ptr1 == frames_to_bytes(runtime, runtime->buffer_size)) + if (ptr4 == 0 && ptr1 == frames_to_bytes(runtime, runtime->buffer_size)) return 0; - - if (ptr3 != ptr4) + + if (ptr3 != ptr4) ptr1 = snd_emu10k1x_ptr_read(emu, PLAYBACK_POINTER, channel); ptr2 = bytes_to_frames(runtime, ptr1); ptr2 += (ptr4 >> 3) * runtime->period_size; @@ -595,7 +595,7 @@ static int snd_emu10k1x_pcm_hw_params_capture(struct snd_pcm_substream *substrea struct snd_pcm_runtime *runtime = substream->runtime; struct emu10k1x_pcm *epcm = runtime->private_data; - if (! epcm->voice) { + if (!epcm->voice) { if (epcm->emu->capture_voice.use) return -EBUSY; epcm->voice = &epcm->emu->capture_voice; @@ -634,7 +634,7 @@ static int snd_emu10k1x_pcm_prepare_capture(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_emu10k1x_ptr_write(emu, CAPTURE_DMA_ADDR, 0, runtime->dma_addr); - snd_emu10k1x_ptr_write(emu, CAPTURE_BUFFER_SIZE, 0, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes + snd_emu10k1x_ptr_write(emu, CAPTURE_BUFFER_SIZE, 0, frames_to_bytes(runtime, runtime->buffer_size)<<16); /* buffer size in bytes */ snd_emu10k1x_ptr_write(emu, CAPTURE_POINTER, 0, 0); snd_emu10k1x_ptr_write(emu, CAPTURE_UNKNOWN, 0, 0); @@ -652,14 +652,14 @@ static int snd_emu10k1x_pcm_trigger_capture(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_emu10k1x_intr_enable(emu, INTE_CAP_0_LOOP | + snd_emu10k1x_intr_enable(emu, INTE_CAP_0_LOOP | INTE_CAP_0_HALF_LOOP); snd_emu10k1x_ptr_write(emu, TRIGGER_CHANNEL, 0, snd_emu10k1x_ptr_read(emu, TRIGGER_CHANNEL, 0)|TRIGGER_CAPTURE); epcm->running = 1; break; case SNDRV_PCM_TRIGGER_STOP: epcm->running = 0; - snd_emu10k1x_intr_disable(emu, INTE_CAP_0_LOOP | + snd_emu10k1x_intr_disable(emu, INTE_CAP_0_LOOP | INTE_CAP_0_HALF_LOOP); snd_emu10k1x_ptr_write(emu, TRIGGER_CHANNEL, 0, snd_emu10k1x_ptr_read(emu, TRIGGER_CHANNEL, 0) & ~(TRIGGER_CAPTURE)); break; @@ -706,7 +706,7 @@ static unsigned short snd_emu10k1x_ac97_read(struct snd_ac97 *ac97, struct emu10k1x *emu = ac97->private_data; unsigned long flags; unsigned short val; - + spin_lock_irqsave(&emu->emu_lock, flags); outb(reg, emu->port + AC97ADDRESS); val = inw(emu->port + AC97DATA); @@ -719,7 +719,7 @@ static void snd_emu10k1x_ac97_write(struct snd_ac97 *ac97, { struct emu10k1x *emu = ac97->private_data; unsigned long flags; - + spin_lock_irqsave(&emu->emu_lock, flags); outb(reg, emu->port + AC97ADDRESS); outw(val, emu->port + AC97DATA); @@ -735,7 +735,7 @@ static int snd_emu10k1x_ac97(struct emu10k1x *chip) .write = snd_emu10k1x_ac97_write, .read = snd_emu10k1x_ac97_read, }; - + if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) return err; pbus->no_vra = 1; /* we don't need VRA */ @@ -749,26 +749,26 @@ static int snd_emu10k1x_ac97(struct emu10k1x *chip) static int snd_emu10k1x_free(struct emu10k1x *chip) { snd_emu10k1x_ptr_write(chip, TRIGGER_CHANNEL, 0, 0); - // disable interrupts + /* disable interrupts */ outl(0, chip->port + INTE); - // disable audio + /* disable audio */ outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); /* release the irq */ if (chip->irq >= 0) free_irq(chip->irq, chip); - // release the i/o port + /* release the i/o port */ release_and_free_resource(chip->res_port); - // release the DMA + /* release the DMA */ if (chip->dma_buffer.area) { snd_dma_free_pages(&chip->dma_buffer); } pci_disable_device(chip->pci); - // release the data + /* release the data */ kfree(chip); return 0; } @@ -790,43 +790,43 @@ static irqreturn_t snd_emu10k1x_interrupt(int irq, void *dev_id) status = inl(chip->port + IPR); - if (! status) + if (!status) return IRQ_NONE; - // capture interrupt + /* capture interrupt */ if (status & (IPR_CAP_0_LOOP | IPR_CAP_0_HALF_LOOP)) { struct emu10k1x_voice *cap_voice = &chip->capture_voice; if (cap_voice->use) snd_emu10k1x_pcm_interrupt(chip, cap_voice); else - snd_emu10k1x_intr_disable(chip, + snd_emu10k1x_intr_disable(chip, INTE_CAP_0_LOOP | INTE_CAP_0_HALF_LOOP); } - + mask = IPR_CH_0_LOOP|IPR_CH_0_HALF_LOOP; for (i = 0; i < 3; i++) { if (status & mask) { if (pvoice->use) snd_emu10k1x_pcm_interrupt(chip, pvoice); - else + else snd_emu10k1x_intr_disable(chip, mask); } pvoice++; mask <<= 1; } - + if (status & (IPR_MIDITRANSBUFEMPTY|IPR_MIDIRECVBUFEMPTY)) { if (chip->midi.interrupt) chip->midi.interrupt(chip, status); else snd_emu10k1x_intr_disable(chip, INTE_MIDITXENABLE|INTE_MIDIRXENABLE); } - - // acknowledge the interrupt if necessary + + /* acknowledge the interrupt if necessary */ outl(status, chip->port + IPR); - // snd_printk(KERN_INFO "interrupt %08x\n", status); + /* snd_printk(KERN_INFO "interrupt %08x\n", status); */ return IRQ_HANDLED; } @@ -835,17 +835,17 @@ static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct s struct snd_pcm *pcm; int err; int capture = 0; - + if (rpcm) *rpcm = NULL; if (device == 0) capture = 1; - + if ((err = snd_pcm_new(emu->card, "emu10k1x", device, 1, capture, &pcm)) < 0) return err; - + pcm->private_data = emu; - + switch(device) { case 0: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1x_playback_ops); @@ -873,12 +873,12 @@ static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct s emu->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + snd_dma_pci_data(emu->pci), 32*1024, 32*1024); - + if (rpcm) *rpcm = pcm; - + return 0; } @@ -916,10 +916,10 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, spin_lock_init(&chip->emu_lock); spin_lock_init(&chip->voice_lock); - + chip->port = pci_resource_start(pci, 0); if ((chip->res_port = request_region(chip->port, 8, - "EMU10K1X")) == NULL) { + "EMU10K1X")) == NULL) { snd_printk(KERN_ERR "emu10k1x: cannot allocate the port 0x%lx\n", chip->port); snd_emu10k1x_free(chip); return -EBUSY; @@ -932,8 +932,8 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - - if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 4 * 1024, &chip->dma_buffer) < 0) { snd_emu10k1x_free(chip); return -ENOMEM; @@ -947,9 +947,9 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, snd_printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model, chip->revision, chip->serial); - outl(0, chip->port + INTE); + outl(0, chip->port + INTE); - for(ch = 0; ch < 3; ch++) { + for (ch = 0; ch < 3; ch++) { chip->voices[ch].emu = chip; chip->voices[ch].number = ch; } @@ -969,27 +969,27 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, * P = 0 (Consumer) */ snd_emu10k1x_ptr_write(chip, SPCS0, 0, - chip->spdif_bits[0] = + chip->spdif_bits[0] = SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); snd_emu10k1x_ptr_write(chip, SPCS1, 0, - chip->spdif_bits[1] = + chip->spdif_bits[1] = SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); snd_emu10k1x_ptr_write(chip, SPCS2, 0, - chip->spdif_bits[2] = + chip->spdif_bits[2] = SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); - snd_emu10k1x_ptr_write(chip, SPDIF_SELECT, 0, 0x700); // disable SPDIF - snd_emu10k1x_ptr_write(chip, ROUTING, 0, 0x1003F); // routing - snd_emu10k1x_gpio_write(chip, 0x1080); // analog mode + snd_emu10k1x_ptr_write(chip, SPDIF_SELECT, 0, 0x700); /* disable SPDIF */ + snd_emu10k1x_ptr_write(chip, ROUTING, 0, 0x1003F); /* routing */ + snd_emu10k1x_gpio_write(chip, 0x1080); /* analog mode */ outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); @@ -1002,7 +1002,7 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, return 0; } -static void snd_emu10k1x_proc_reg_read(struct snd_info_entry *entry, +static void snd_emu10k1x_proc_reg_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct emu10k1x *emu = entry->private_data; @@ -1011,16 +1011,16 @@ static void snd_emu10k1x_proc_reg_read(struct snd_info_entry *entry, int i; snd_iprintf(buffer, "Registers:\n\n"); - for(i = 0; i < 0x20; i+=4) { + for (i = 0; i < 0x20; i += 4) { spin_lock_irqsave(&emu->emu_lock, flags); value = inl(emu->port + i); spin_unlock_irqrestore(&emu->emu_lock, flags); snd_iprintf(buffer, "Register %02X: %08lX\n", i, value); } snd_iprintf(buffer, "\nRegisters\n\n"); - for(i = 0; i <= 0x48; i++) { + for (i = 0; i <= 0x48; i++) { value = snd_emu10k1x_ptr_read(emu, i, 0); - if(i < 0x10 || (i >= 0x20 && i < 0x40)) { + if (i < 0x10 || (i >= 0x20 && i < 0x40)) { value1 = snd_emu10k1x_ptr_read(emu, i, 1); value2 = snd_emu10k1x_ptr_read(emu, i, 2); snd_iprintf(buffer, "%02X: %08lX %08lX %08lX\n", i, value, value1, value2); @@ -1030,7 +1030,7 @@ static void snd_emu10k1x_proc_reg_read(struct snd_info_entry *entry, } } -static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, +static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct emu10k1x *emu = entry->private_data; @@ -1041,8 +1041,8 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) + && (channel_id >= 0) && (channel_id <= 2)) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } @@ -1050,14 +1050,14 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu) { struct snd_info_entry *entry; - - if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { + + if (!snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; entry->mode |= S_IWUSR; entry->private_data = emu; } - + return 0; } @@ -1080,15 +1080,15 @@ static int snd_emu10k1x_shared_spdif_put(struct snd_kcontrol *kcontrol, unsigned int val; int change = 0; - val = ucontrol->value.integer.value[0] ; + val = ucontrol->value.integer.value[0]; if (val) { - // enable spdif output + /* enable spdif output */ snd_emu10k1x_ptr_write(emu, SPDIF_SELECT, 0, 0x000); snd_emu10k1x_ptr_write(emu, ROUTING, 0, 0x700); snd_emu10k1x_gpio_write(emu, 0x1000); } else { - // disable spdif output + /* disable spdif output */ snd_emu10k1x_ptr_write(emu, SPDIF_SELECT, 0, 0x700); snd_emu10k1x_ptr_write(emu, ROUTING, 0, 0x1003F); snd_emu10k1x_gpio_write(emu, 0x1080); @@ -1317,7 +1317,7 @@ static int snd_emu10k1x_midi_input_open(struct snd_rawmidi_substream *substream) struct emu10k1x *emu; struct emu10k1x_midi *midi = substream->rmidi->private_data; unsigned long flags; - + emu = midi->emu; if (snd_BUG_ON(!emu)) return -ENXIO; @@ -1439,7 +1439,7 @@ static void snd_emu10k1x_midi_output_trigger(struct snd_rawmidi_substream *subst if (up) { int max = 4; unsigned char byte; - + /* try to send some amount of bytes here before interrupts */ spin_lock_irqsave(&midi->output_lock, flags); while (max > 0) { @@ -1575,7 +1575,7 @@ static int __devinit snd_emu10k1x_probe(struct pci_dev *pci, snd_card_free(card); return err; } - + if ((err = snd_emu10k1x_midi(chip)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 7dba08f..f8ce97b 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -52,7 +52,7 @@ /* * Tables - */ + */ static char *fxbuses[16] = { /* 0x00 */ "PCM Left", @@ -303,7 +303,7 @@ static const u32 onoff_table[2] = { /* */ - + static inline mm_segment_t snd_enter_user(void) { mm_segment_t fs = get_fs(); @@ -342,7 +342,7 @@ static int snd_emu10k1_gpr_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl (struct snd_emu10k1_fx8010_ctl *) kcontrol->private_value; unsigned long flags; unsigned int i; - + spin_lock_irqsave(&emu->reg_lock, flags); for (i = 0; i < ctl->vcount; i++) ucontrol->value.integer.value[i] = ctl->value[i]; @@ -359,7 +359,7 @@ static int snd_emu10k1_gpr_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl unsigned int nval, val; unsigned int i, j; int change = 0; - + spin_lock_irqsave(&emu->reg_lock, flags); for (i = 0; i < ctl->vcount; i++) { nval = ucontrol->value.integer.value[i]; @@ -398,7 +398,7 @@ static int snd_emu10k1_gpr_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl break; } } - __error: +__error: spin_unlock_irqrestore(&emu->reg_lock, flags); return change; } @@ -431,7 +431,7 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu, { struct snd_emu10k1_fx8010_irq *irq; unsigned long flags; - + irq = kmalloc(sizeof(*irq), GFP_ATOMIC); if (irq == NULL) return -ENOMEM; @@ -459,7 +459,7 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu, { struct snd_emu10k1_fx8010_irq *tmp; unsigned long flags; - + spin_lock_irqsave(&emu->fx8010.irq_lock, flags); if ((tmp = emu->fx8010.irq_handlers) == irq) { emu->fx8010.irq_handlers = tmp->next; @@ -703,7 +703,7 @@ static int copy_gctl_to_user(struct snd_emu10k1 *emu, if (emu->support_tlv) return copy_to_user(&_gctl[idx], gctl, sizeof(*gctl)); - + octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl; return copy_to_user(&octl[idx], gctl, sizeof(*octl)); } @@ -716,7 +716,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu, struct snd_ctl_elem_id id; struct snd_emu10k1_fx8010_control_gpr *gctl; int err; - + for (i = 0, _id = icode->gpr_del_controls; i < icode->gpr_del_control_count; i++, _id++) { if (copy_from_user(&id, _id, sizeof(id))) @@ -725,7 +725,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu, return -ENOENT; } gctl = kmalloc(sizeof(*gctl), GFP_KERNEL); - if (! gctl) + if (!gctl) return -ENOMEM; err = 0; for (i = 0; i < icode->gpr_add_control_count; i++) { @@ -755,7 +755,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu, goto __error; } } - __error: +__error: kfree(gctl); return err; } @@ -763,7 +763,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu, static void snd_emu10k1_ctl_private_free(struct snd_kcontrol *kctl) { struct snd_emu10k1_fx8010_ctl *ctl; - + ctl = (struct snd_emu10k1_fx8010_ctl *) kctl->private_value; kctl->private_value = 0; list_del(&ctl->list); @@ -801,7 +801,7 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, err = -EINVAL; goto __error; } - if (! gctl->id.name[0]) { + if (!gctl->id.name[0]) { err = -EINVAL; goto __error; } @@ -857,7 +857,7 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, } snd_emu10k1_gpr_ctl_put(ctl->kcontrol, val); } - __error: +__error: kfree(nctl); kfree(gctl); kfree(val); @@ -872,7 +872,7 @@ static int snd_emu10k1_del_controls(struct snd_emu10k1 *emu, struct snd_ctl_elem_id __user *_id; struct snd_emu10k1_fx8010_ctl *ctl; struct snd_card *card = emu->card; - + for (i = 0, _id = icode->gpr_del_controls; i < icode->gpr_del_control_count; i++, _id++) { if (copy_from_user(&id, _id, sizeof(id))) @@ -896,7 +896,7 @@ static int snd_emu10k1_list_controls(struct snd_emu10k1 *emu, struct snd_ctl_elem_id *id; gctl = kmalloc(sizeof(*gctl), GFP_KERNEL); - if (! gctl) + if (!gctl) return -ENOMEM; list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) { @@ -959,7 +959,7 @@ static int snd_emu10k1_icode_poke(struct snd_emu10k1 *emu, snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg); else snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg); - __error: +__error: mutex_unlock(&emu->fx8010.lock); return err; } @@ -1023,7 +1023,7 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, for (i = 0; i < pcm->channels; i++) pcm->etram[i] = ipcm->etram[i]; } - __error: +__error: spin_unlock_irq(&emu->reg_lock); mutex_unlock(&emu->fx8010.lock); return err; @@ -1075,7 +1075,7 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl, ctl->min = 0; ctl->max = 100; ctl->tlv = snd_emu10k1_db_scale1; - ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; + ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; } static void __devinit @@ -1177,7 +1177,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) /* clear free GPRs */ for (i = 0; i < 512; i++) set_bit(i, icode->gpr_valid); - + /* clear TRAM data & address lines */ for (i = 0; i < 256; i++) set_bit(i, icode->tram_valid); @@ -1210,7 +1210,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_REAR)); snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Surround Playback Volume", gpr, 100); gpr += 2; - + /* PCM Side Playback (independent from stereo mix) */ if (emu->card_capabilities->spk71) { A_OP(icode, &ptr, iMAC0, A_GPR(playback+6), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_SIDE)); @@ -1228,7 +1228,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) A_OP(icode, &ptr, iMAC0, A_GPR(playback+5), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LFE)); snd_emu10k1_init_mono_control(&controls[nctl++], "PCM LFE Playback Volume", gpr, 100); gpr++; - + /* * Stereo Mix */ @@ -1255,7 +1255,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_FXBUS(FXBUS_MIDI_RIGHT)); snd_emu10k1_init_stereo_control(&controls[nctl++], "Synth Capture Volume", gpr, 0); gpr += 2; - + /* * inputs */ @@ -1288,7 +1288,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) snd_emu10k1_init_stereo_control(&controls[nctl++], "Mic Capture Volume", gpr, 0); gpr += 2; - /* mic capture buffer */ + /* mic capture buffer */ A_OP(icode, &ptr, iINTERP, A_EXTOUT(A_EXTOUT_MIC_CAP), A_EXTIN(A_EXTIN_AC97_L), 0xcd, A_EXTIN(A_EXTIN_AC97_R)); /* Audigy CD Playback Volume */ @@ -1331,7 +1331,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->card_capabilities->ac97_chip ? "Line2 Capture Volume" : "Line Capture Volume", gpr, 0); gpr += 2; - + /* Philips ADC Playback Volume */ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_ADC_L); A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_ADC_R); @@ -1357,13 +1357,13 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->card_capabilities->ac97_chip ? "Aux2 Capture Volume" : "Aux Capture Volume", gpr, 0); gpr += 2; - + /* Stereo Mix Front Playback Volume */ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_GPR(playback), A_GPR(gpr), A_GPR(stereo_mix)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_GPR(playback+1), A_GPR(gpr+1), A_GPR(stereo_mix+1)); snd_emu10k1_init_stereo_control(&controls[nctl++], "Front Playback Volume", gpr, 100); gpr += 2; - + /* Stereo Mix Surround Playback */ A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_GPR(playback+2), A_GPR(gpr), A_GPR(stereo_mix)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_GPR(playback+3), A_GPR(gpr+1), A_GPR(stereo_mix+1)); @@ -1381,7 +1381,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_OP(icode, &ptr, iMAC0, A_GPR(playback+5), A_GPR(playback+5), A_GPR(gpr), A_GPR(tmp)); snd_emu10k1_init_mono_control(&controls[nctl++], "LFE Playback Volume", gpr, 0); gpr++; - + if (emu->card_capabilities->spk71) { /* Stereo Mix Side Playback */ A_OP(icode, &ptr, iMAC0, A_GPR(playback+6), A_GPR(playback+6), A_GPR(gpr), A_GPR(stereo_mix)); @@ -1420,7 +1420,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 6), A_GPR(playback + 6), A_C_00000000, A_C_00000000); /* side left */ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 7), A_GPR(playback + 7), A_C_00000000, A_C_00000000); /* side right */ } - + ctl = &controls[nctl + 0]; ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -1529,7 +1529,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) } } - /* IEC958 Optical Raw Playback Switch */ + /* IEC958 Optical Raw Playback Switch */ gpr_map[gpr++] = 0; gpr_map[gpr++] = 0x1008; gpr_map[gpr++] = 0xffff0000; @@ -1552,7 +1552,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) } snd_emu10k1_init_stereo_onoff_control(controls + nctl++, SNDRV_CTL_NAME_IEC958("Optical Raw ",PLAYBACK,SWITCH), gpr, 0); gpr += 2; - + A_PUT_STEREO_OUTPUT(A_EXTOUT_REAR_L, A_EXTOUT_REAR_R, playback+2 + SND_EMU10K1_PLAYBACK_CHANNELS); A_PUT_OUTPUT(A_EXTOUT_CENTER, playback+4 + SND_EMU10K1_PLAYBACK_CHANNELS); A_PUT_OUTPUT(A_EXTOUT_LFE, playback+5 + SND_EMU10K1_PLAYBACK_CHANNELS); @@ -1569,10 +1569,10 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->ca0108_chip) { snd_printk("EMU2 inputs on\n"); for (z = 0; z < 0x10; z++) { - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A3_EMU32IN(z), - A_FXBUS2(z*2) ); + A_FXBUS2(z*2)); } } else { snd_printk("EMU inputs on\n"); @@ -1584,7 +1584,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) * so all other A_P16VIN channels will need to also be delayed */ /* Left ADC in. 1 of 2 */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0)); /* Right ADC in 1 of 2 */ gpr_map[gpr++] = 0x00000000; /* Delaying by one sample: instead of copying the input @@ -1592,28 +1592,28 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) * we use an auxiliary register, delaying the value by one * sample */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); /* For 96kHz mode */ /* Left ADC in. 2 of 2 */ gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); /* Right ADC in 2 of 2 */ gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe)); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); /* Pavel Hofman - we still have voices, A_FXBUS2s, and * A_P16VINs available - @@ -1692,7 +1692,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_OP(icode, &ptr, iACC3, A_FXBUS2(z), A_C_00000000, A_C_00000000, A_EXTIN(z)); } } - + #endif /* JCD test */ /* * ok, set up done.. @@ -1715,7 +1715,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->support_tlv = 0; /* clear again */ snd_leave_user(seg); - __err: +__err: kfree(controls); if (icode != NULL) { kfree((void __force *)icode->gpr_map); @@ -1804,7 +1804,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) icode->tram_data_map = icode->gpr_map + 256; icode->tram_addr_map = icode->tram_data_map + 160; icode->code = icode->tram_addr_map + 160; - + /* clear free GPRs */ for (i = 0; i < 256; i++) set_bit(i, icode->gpr_valid); @@ -1904,7 +1904,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) /* 18: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_MINUS, C_00000001); /* 19: */ OP(icode, &ptr, iACC3, GPR(tmp + 0), C_00000000, C_00000000, C_00000000); /* 1a: */ OP(icode, &ptr, iACC3, GPR(ipcm->gpr_ptr), GPR(tmp + 0), C_00000000, C_00000000); - + /* 1b: */ OP(icode, &ptr, iACC3, GPR(ipcm->gpr_tmpcount), GPR(ipcm->gpr_tmpcount), C_ffffffff, C_00000000); /* 1c: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, C_00000002); /* 1d: */ OP(icode, &ptr, iACC3, GPR(ipcm->gpr_tmpcount), GPR(ipcm->gpr_count), C_00000000, C_00000000); @@ -1931,7 +1931,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) VOLUME(icode, &ptr, playback + 2 + z, z, gpr + z); snd_emu10k1_init_stereo_control(controls + i++, "Wave Surround Playback Volume", gpr, 0); gpr += 2; - + /* Wave Center/LFE Playback Volume */ OP(icode, &ptr, iACC3, GPR(tmp + 0), FXBUS(FXBUS_PCM_LEFT), FXBUS(FXBUS_PCM_RIGHT), C_00000000); OP(icode, &ptr, iMACINT0, GPR(tmp + 0), C_00000000, GPR(tmp + 0), C_00000002); @@ -2016,14 +2016,14 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) VOLUME_ADDIN(icode, &ptr, capture + 1, EXTIN_AC97_R, gpr); gpr++; snd_emu10k1_init_stereo_control(controls + i++, "AC97 Capture Volume", gpr-2, 100); } - + if (emu->fx8010.extin_mask & ((1<<EXTIN_SPDIF_CD_L)|(1<<EXTIN_SPDIF_CD_R))) { /* IEC958 TTL Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_SPDIF_CD_L + z, gpr + z); snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; - + /* IEC958 TTL Capture Volume + Switch */ for (z = 0; z < 2; z++) { SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_SPDIF_CD_L + z, gpr + 2 + z); @@ -2033,14 +2033,14 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",CAPTURE,SWITCH), gpr + 2, 0); gpr += 4; } - + if (emu->fx8010.extin_mask & ((1<<EXTIN_ZOOM_L)|(1<<EXTIN_ZOOM_R))) { /* Zoom Video Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_ZOOM_L + z, gpr + z); snd_emu10k1_init_stereo_control(controls + i++, "Zoom Video Playback Volume", gpr, 0); gpr += 2; - + /* Zoom Video Capture Volume + Switch */ for (z = 0; z < 2; z++) { SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_ZOOM_L + z, gpr + 2 + z); @@ -2050,14 +2050,14 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_init_stereo_onoff_control(controls + i++, "Zoom Video Capture Switch", gpr + 2, 0); gpr += 4; } - + if (emu->fx8010.extin_mask & ((1<<EXTIN_TOSLINK_L)|(1<<EXTIN_TOSLINK_R))) { /* IEC958 Optical Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_TOSLINK_L + z, gpr + z); snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; - + /* IEC958 Optical Capture Volume */ for (z = 0; z < 2; z++) { SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_TOSLINK_L + z, gpr + 2 + z); @@ -2067,14 +2067,14 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",CAPTURE,SWITCH), gpr + 2, 0); gpr += 4; } - + if (emu->fx8010.extin_mask & ((1<<EXTIN_LINE1_L)|(1<<EXTIN_LINE1_R))) { /* Line LiveDrive Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_LINE1_L + z, gpr + z); snd_emu10k1_init_stereo_control(controls + i++, "Line LiveDrive Playback Volume", gpr, 0); gpr += 2; - + /* Line LiveDrive Capture Volume + Switch */ for (z = 0; z < 2; z++) { SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_LINE1_L + z, gpr + 2 + z); @@ -2084,14 +2084,14 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_init_stereo_onoff_control(controls + i++, "Line LiveDrive Capture Switch", gpr + 2, 0); gpr += 4; } - + if (emu->fx8010.extin_mask & ((1<<EXTIN_COAX_SPDIF_L)|(1<<EXTIN_COAX_SPDIF_R))) { /* IEC958 Coax Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_COAX_SPDIF_L + z, gpr + z); snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; - + /* IEC958 Coax Capture Volume + Switch */ for (z = 0; z < 2; z++) { SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_COAX_SPDIF_L + z, gpr + 2 + z); @@ -2101,7 +2101,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",CAPTURE,SWITCH), gpr + 2, 0); gpr += 4; } - + if (emu->fx8010.extin_mask & ((1<<EXTIN_LINE2_L)|(1<<EXTIN_LINE2_R))) { /* Line LiveDrive Playback Volume */ for (z = 0; z < 2; z++) @@ -2109,7 +2109,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_init_stereo_control(controls + i++, "Line2 LiveDrive Playback Volume", gpr, 0); controls[i-1].id.index = 1; gpr += 2; - + /* Line LiveDrive Capture Volume */ for (z = 0; z < 2; z++) { SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_LINE2_L + z, gpr + 2 + z); @@ -2249,7 +2249,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) gpr += 4; } - + if (emu->fx8010.extout_mask & ((1<<EXTOUT_REAR_L)|(1<<EXTOUT_REAR_R))) for (z = 0; z < 2; z++) OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_REAR_L + z), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 2 + z), C_00000000, C_00000000); @@ -2277,12 +2277,12 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ALFE), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 1), C_00000000, C_00000000); #endif } - + #ifndef EMU10K1_CAPTURE_DIGITAL_OUT for (z = 0; z < 2; z++) OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ADC_CAP_L + z), GPR(capture + z), C_00000000, C_00000000); #endif - + if (emu->fx8010.extout_mask & (1<<EXTOUT_MIC_CAP)) OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_MIC_CAP), GPR(capture + 2), C_00000000, C_00000000); @@ -2291,10 +2291,10 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) /* On the Live! 5.1, FXBUS2(1) and FXBUS(2) are shared with EXTOUT_ACENTER * and EXTOUT_ALFE, so we can't connect inputs to them for multitrack recording. * - * Since only 14 of the 16 EXTINs are used, this is not a big problem. - * We route AC97L and R to FX capture 14 and 15, SPDIF CD in to FX capture - * 0 and 3, then the rest of the EXTINs to the corresponding FX capture - * channel. Multitrack recorders will still see the center/lfe output signal + * Since only 14 of the 16 EXTINs are used, this is not a big problem. + * We route AC97L and R to FX capture 14 and 15, SPDIF CD in to FX capture + * 0 and 3, then the rest of the EXTINs to the corresponding FX capture + * channel. Multitrack recorders will still see the center/lfe output signal * on the second and third channels. */ OP(icode, &ptr, iACC3, FXBUS2(14), C_00000000, C_00000000, EXTIN(0)); @@ -2307,7 +2307,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) for (z = 0; z < 16; z++) OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z)); } - + if (gpr > tmp) { snd_BUG(); @@ -2319,7 +2319,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) err = -EIO; goto __err; } - + /* clear remaining instruction memory */ while (ptr < 0x200) OP(icode, &ptr, iACC3, C_00000000, C_00000000, C_00000000, C_00000000); @@ -2335,7 +2335,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_leave_user(seg); if (err >= 0) err = snd_emu10k1_ipcm_poke(emu, ipcm); - __err: +__err: kfree(ipcm); kfree(controls); if (icode != NULL) { @@ -2471,7 +2471,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un unsigned int addr; void __user *argp = (void __user *)arg; int res; - + switch (cmd) { case SNDRV_EMU10K1_IOCTL_PVERSION: emu->support_tlv = 1; @@ -2617,7 +2617,7 @@ int __devinit snd_emu10k1_fx8010_new(struct snd_emu10k1 *emu, int device, struct { struct snd_hwdep *hw; int err; - + if (rhwdep) *rhwdep = NULL; if ((err = snd_hwdep_new(emu->card, "FX8010", device, &hw)) < 0) @@ -2640,16 +2640,16 @@ int __devinit snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu) len = emu->audigy ? 0x200 : 0x100; emu->saved_gpr = kmalloc(len * 4, GFP_KERNEL); - if (! emu->saved_gpr) + if (!emu->saved_gpr) return -ENOMEM; len = emu->audigy ? 0x100 : 0xa0; emu->tram_val_saved = kmalloc(len * 4, GFP_KERNEL); emu->tram_addr_saved = kmalloc(len * 4, GFP_KERNEL); - if (! emu->tram_val_saved || ! emu->tram_addr_saved) + if (!emu->tram_val_saved || ! emu->tram_addr_saved) return -ENOMEM; len = emu->audigy ? 2 * 1024 : 2 * 512; emu->saved_icode = vmalloc(len * 4); - if (! emu->saved_icode) + if (!emu->saved_icode) return -ENOMEM; return 0; } @@ -2721,7 +2721,7 @@ void snd_emu10k1_efx_resume(struct snd_emu10k1 *emu) for (i = 0; i < len; i++) { snd_emu10k1_ptr_write(emu, TANKMEMDATAREGBASE + i, 0, emu->tram_val_saved[i]); - if (! emu->audigy) + if (!emu->audigy) snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + i, 0, emu->tram_addr_saved[i]); else { diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index f34bbfb..a1366e0 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -83,7 +83,7 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol, * Items labels in enum mixer controls assigning source data to * each destination */ -static char *emu1010_src_texts[] = { +static char *emu1010_src_texts[] = { "Silence", "Dock Mic A", "Dock Mic B", @@ -615,12 +615,12 @@ static int snd_emu1010_adc_pads_put(struct snd_kcontrol *kcontrol, struct snd_ct unsigned int val, cache; val = ucontrol->value.integer.value[0]; cache = emu->emu1010.adc_pads; - if (val == 1) + if (val == 1) cache = cache | mask; else cache = cache & ~mask; if (cache != emu->emu1010.adc_pads) { - snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, cache ); + snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, cache); emu->emu1010.adc_pads = cache; } @@ -663,12 +663,12 @@ static int snd_emu1010_dac_pads_put(struct snd_kcontrol *kcontrol, struct snd_ct unsigned int val, cache; val = ucontrol->value.integer.value[0]; cache = emu->emu1010.dac_pads; - if (val == 1) + if (val == 1) cache = cache | mask; else cache = cache & ~mask; if (cache != emu->emu1010.dac_pads) { - snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, cache ); + snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, cache); emu->emu1010.dac_pads = cache; } @@ -702,7 +702,7 @@ static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol, static char *texts[4] = { "44100", "48000", "SPDIF", "ADAT" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 4; @@ -710,8 +710,8 @@ static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; - - + + } static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol, @@ -730,7 +730,7 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol, unsigned int val; int change = 0; - val = ucontrol->value.enumerated.item[0] ; + val = ucontrol->value.enumerated.item[0]; /* Limit: uinfo->value.enumerated.items = 4; */ if (val >= 4) return -EINVAL; @@ -741,75 +741,75 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol, case 0: /* 44100 */ /* Mute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE); /* Default fallback clock 48kHz */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_44_1K ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_44_1K); /* Word Clock source, Internal 44.1kHz x1 */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, - EMU_HANA_WCLOCK_INT_44_1K | EMU_HANA_WCLOCK_1X ); + EMU_HANA_WCLOCK_INT_44_1K | EMU_HANA_WCLOCK_1X); /* Set LEDs on Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, - EMU_HANA_DOCK_LEDS_2_44K | EMU_HANA_DOCK_LEDS_2_LOCK ); + EMU_HANA_DOCK_LEDS_2_44K | EMU_HANA_DOCK_LEDS_2_LOCK); /* Allow DLL to settle */ msleep(10); /* Unmute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); break; case 1: /* 48000 */ /* Mute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE); /* Default fallback clock 48kHz */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K); /* Word Clock source, Internal 48kHz x1 */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, - EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_1X ); + EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_1X); /* Set LEDs on Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, - EMU_HANA_DOCK_LEDS_2_48K | EMU_HANA_DOCK_LEDS_2_LOCK ); + EMU_HANA_DOCK_LEDS_2_48K | EMU_HANA_DOCK_LEDS_2_LOCK); /* Allow DLL to settle */ msleep(10); /* Unmute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); break; - + case 2: /* Take clock from S/PDIF IN */ /* Mute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE); /* Default fallback clock 48kHz */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K); /* Word Clock source, sync to S/PDIF input */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, - EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X ); + EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X); /* Set LEDs on Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, - EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK ); - /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ + EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK); + /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ /* Allow DLL to settle */ msleep(10); /* Unmute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); break; - - case 3: + + case 3: /* Take clock from ADAT IN */ /* Mute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE); /* Default fallback clock 48kHz */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K); /* Word Clock source, sync to ADAT input */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, - EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X ); + EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X); /* Set LEDs on Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK ); - /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK); + /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ /* Allow DLL to settle */ msleep(10); /* Unmute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); - - - break; + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + + + break; } } return change; @@ -958,7 +958,7 @@ static int snd_audigy_i2c_volume_put(struct snd_kcontrol *kcontrol, return 0; if (ogain != ngain) { if (emu->i2c_capture_source == source_id) - snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) ); + snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff)); emu->i2c_capture_volume[source_id][0] = ngain; change = 1; } @@ -1014,7 +1014,7 @@ static int snd_audigy_spdif_output_rate_get(struct snd_kcontrol *kcontrol, struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); unsigned int tmp; unsigned long flags; - + spin_lock_irqsave(&emu->reg_lock, flags); tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); @@ -1058,7 +1058,7 @@ static int snd_audigy_spdif_output_rate_put(struct snd_kcontrol *kcontrol, break; } - + spin_lock_irqsave(&emu->reg_lock, flags); reg = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp = reg & ~A_SPDIF_RATE_MASK; @@ -1182,7 +1182,7 @@ static int snd_emu10k1_send_routing_get(struct snd_kcontrol *kcontrol, spin_lock_irqsave(&emu->reg_lock, flags); for (voice = 0; voice < 3; voice++) for (idx = 0; idx < num_efx; idx++) - ucontrol->value.integer.value[(voice * num_efx) + idx] = + ucontrol->value.integer.value[(voice * num_efx) + idx] = mix->send_routing[voice][idx] & mask; spin_unlock_irqrestore(&emu->reg_lock, flags); return 0; @@ -1207,7 +1207,7 @@ static int snd_emu10k1_send_routing_put(struct snd_kcontrol *kcontrol, mix->send_routing[voice][idx] = val; change = 1; } - } + } if (change && mix->epcm) { if (mix->epcm->voices[0] && mix->epcm->voices[1]) { update_emu10k1_fxrt(emu, mix->epcm->voices[0]->number, @@ -1395,7 +1395,7 @@ static int snd_emu10k1_efx_send_routing_get(struct snd_kcontrol *kcontrol, spin_lock_irqsave(&emu->reg_lock, flags); for (idx = 0; idx < num_efx; idx++) - ucontrol->value.integer.value[idx] = + ucontrol->value.integer.value[idx] = mix->send_routing[0][idx] & mask; spin_unlock_irqrestore(&emu->reg_lock, flags); return 0; @@ -1419,7 +1419,7 @@ static int snd_emu10k1_efx_send_routing_put(struct snd_kcontrol *kcontrol, mix->send_routing[0][idx] = val; change = 1; } - } + } if (change && mix->epcm) { if (mix->epcm->voices[ch]) { @@ -1581,7 +1581,7 @@ static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, if (emu->card_capabilities->invert_shared_spdif) ucontrol->value.integer.value[0] = !ucontrol->value.integer.value[0]; - + return 0; } @@ -1597,7 +1597,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, if (emu->card_capabilities->invert_shared_spdif) sw = !sw; spin_lock_irqsave(&emu->reg_lock, flags); - if ( emu->card_capabilities->i2c_adc) { + if (emu->card_capabilities->i2c_adc) { /* Do nothing for Audigy 2 ZS Notebook */ } else if (emu->audigy) { reg = inl(emu->port + A_IOCFG); @@ -1738,7 +1738,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, NULL }; static char *audigy_rename_ctls_i2c_adc[] = { - //"Analog Mix Capture Volume","OLD Analog Mix Capture Volume", + /*"Analog Mix Capture Volume","OLD Analog Mix Capture Volume", */ "Line Capture Volume", "Analog Mix Capture Volume", "Wave Playback Volume", "OLD PCM Playback Volume", "Wave Master Playback Volume", "Master Playback Volume", @@ -1808,7 +1808,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, if ((err = snd_ac97_bus(emu->card, 0, &ops, NULL, &pbus)) < 0) return err; pbus->no_vra = 1; /* we don't need VRA */ - + memset(&ac97, 0, sizeof(ac97)); ac97.private_data = emu; ac97.private_free = snd_emu10k1_mixer_free_ac97; @@ -1828,7 +1828,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000); if (emu->card_capabilities->adc_1361t) c = audigy_remove_ctls_1361t_adc; - else + else c = audigy_remove_ctls; } else { /* @@ -1906,13 +1906,13 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, kctl->id.device = multi_device; if ((err = snd_ctl_add(card, kctl))) return err; - + if ((kctl = emu->ctl_efx_send_volume = snd_ctl_new1(&snd_emu10k1_efx_send_volume_control, emu)) == NULL) return -ENOMEM; kctl->id.device = multi_device; if ((err = snd_ctl_add(card, kctl))) return err; - + if ((kctl = emu->ctl_efx_attn = snd_ctl_new1(&snd_emu10k1_efx_attn_control, emu)) == NULL) return -ENOMEM; kctl->id.device = multi_device; @@ -1923,27 +1923,27 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, for (pcm = 0; pcm < 32; pcm++) { struct snd_emu10k1_pcm_mixer *mix; int v; - + mix = &emu->pcm_mixer[pcm]; mix->epcm = NULL; for (v = 0; v < 4; v++) - mix->send_routing[0][v] = - mix->send_routing[1][v] = + mix->send_routing[0][v] = + mix->send_routing[1][v] = mix->send_routing[2][v] = v; - + memset(&mix->send_volume, 0, sizeof(mix->send_volume)); mix->send_volume[0][0] = mix->send_volume[0][1] = mix->send_volume[1][0] = mix->send_volume[2][1] = 255; - + mix->attn[0] = mix->attn[1] = mix->attn[2] = 0xffff; } - + /* initialize the routing and volume table for the multichannel playback stream */ for (pcm = 0; pcm < NUM_EFX_PLAYBACK; pcm++) { struct snd_emu10k1_pcm_mixer *mix; int v; - + mix = &emu->efx_pcm_mixer[pcm]; mix->epcm = NULL; @@ -1954,14 +1954,14 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, if (emu->audigy) for (v = 0; v < 4; v++) mix->send_routing[0][4+v] = 60+v; - + memset(&mix->send_volume, 0, sizeof(mix->send_volume)); mix->send_volume[0][0] = 255; - + mix->attn[0] = 0xffff; } - - if (! emu->card_capabilities->ecard) { /* FIXME: APS has these controls? */ + + if (!emu->card_capabilities->ecard) { /* FIXME: APS has these controls? */ /* sb live! and audigy */ if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu)) == NULL) return -ENOMEM; @@ -1990,7 +1990,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, if ((err = snd_ctl_add(card, kctl))) return err; #endif - } else if (! emu->card_capabilities->ecard) { + } else if (!emu->card_capabilities->ecard) { /* sb live! */ if ((kctl = snd_ctl_new1(&snd_emu10k1_shared_spdif, emu)) == NULL) return -ENOMEM; @@ -2073,7 +2073,7 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, return err; } - if ( emu->card_capabilities->i2c_adc) { + if (emu->card_capabilities->i2c_adc) { int i; err = snd_ctl_add(card, snd_ctl_new1(&snd_audigy_i2c_capture_source, emu)); @@ -2086,6 +2086,6 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, return err; } } - + return 0; } diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c index 8578c70..18495b4 100644 --- a/sound/pci/emu10k1/emumpu401.c +++ b/sound/pci/emu10k1/emumpu401.c @@ -277,7 +277,7 @@ static void snd_emu10k1_midi_output_trigger(struct snd_rawmidi_substream *substr if (up) { int max = 4; unsigned char byte; - + /* try to send some amount of bytes here before interrupts */ spin_lock_irqsave(&midi->output_lock, flags); while (max > 0) { diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cf9276d..f9144dc 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -86,7 +86,7 @@ static void snd_emu10k1_pcm_efx_interrupt(struct snd_emu10k1 *emu, } #endif snd_pcm_period_elapsed(emu->pcm_capture_efx_substream); -} +} static snd_pcm_uframes_t snd_emu10k1_efx_playback_pointer(struct snd_pcm_substream *substream) { @@ -130,7 +130,7 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic epcm->type == PLAYBACK_EMUVOICE ? EMU10K1_PCM : EMU10K1_EFX, voices, &epcm->voices[0]); - + if (err < 0) return err; epcm->voices[0]->epcm = epcm; @@ -253,12 +253,12 @@ static unsigned int emu10k1_select_interprom(unsigned int pitch_target) return CCCA_INTERPROM_4; else if (pitch_target >= PITCH_57081) return CCCA_INTERPROM_3; - else + else return CCCA_INTERPROM_2; } /* - * calculate cache invalidate size + * calculate cache invalidate size * * stereo: channel is stereo * w_16: using 16bit samples @@ -323,7 +323,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, } ccis = emu10k1_ccis(stereo, w_16); - + if (master) { evoice->epcm->ccca_start_addr = start_addr + ccis; if (extra) { @@ -359,7 +359,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ - else + else pitch_target = emu10k1_calc_pitch_target(runtime->rate); if (extra) snd_emu10k1_ptr_write(emu, CCCA, voice, start_addr | @@ -416,7 +416,7 @@ static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream, snd_emu10k1_free_pages(emu, epcm->memblk); epcm->memblk = snd_emu10k1_alloc_pages(emu, substream); epcm->start_addr = 0; - if (! epcm->memblk) + if (!epcm->memblk) return -ENOMEM; mapped = ((struct snd_emu10k1_memblk *)epcm->memblk)->mapped_page; if (mapped < 0) @@ -602,7 +602,7 @@ static int snd_emu10k1_capture_prepare(struct snd_pcm_substream *substream) break; default: break; - } + } snd_emu10k1_ptr_write(emu, epcm->capture_ba_reg, 0, runtime->dma_addr); epcm->capture_bufsize = snd_pcm_lib_buffer_bytes(substream); epcm->capture_bs_val = 0; @@ -685,7 +685,7 @@ static void snd_emu10k1_playback_prepare_voice(struct snd_emu10k1 *emu, struct s snd_emu10k1_ptr_write(emu, CVCF, voice, vattn | 0xffff); snd_emu10k1_ptr_write(emu, DCYSUSV, voice, 0x7f7f); snd_emu10k1_voice_clear_loop_stop(emu, voice); -} +} static void snd_emu10k1_playback_trigger_voice(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice, int master, int extra) { @@ -702,7 +702,7 @@ static void snd_emu10k1_playback_trigger_voice(struct snd_emu10k1 *emu, struct s pitch = snd_emu10k1_rate_to_pitch(runtime->rate) >> 8; if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ - else + else pitch_target = emu10k1_calc_pitch_target(runtime->rate); snd_emu10k1_ptr_write(emu, PTRX_PITCHTARGET, voice, pitch_target); if (master || evoice->epcm->type == PLAYBACK_EFX) @@ -799,7 +799,7 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream, } else snd_emu10k1_ptr_write(emu, FXWC, 0, epcm->capture_cr_val); break; - default: + default: break; } snd_emu10k1_ptr_write(emu, epcm->capture_bs_reg, 0, epcm->capture_bs_val); @@ -875,7 +875,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* prepare voices */ - for (i = 0; i < NUM_EFX_PLAYBACK; i++) { + for (i = 0; i < NUM_EFX_PLAYBACK; i++) { snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]); } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); @@ -899,7 +899,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: epcm->running = 0; - for (i = 0; i < NUM_EFX_PLAYBACK; i++) { + for (i = 0; i < NUM_EFX_PLAYBACK; i++) { snd_emu10k1_playback_stop_voice(emu, epcm->voices[i]); } snd_emu10k1_playback_stop_voice(emu, epcm->extra); @@ -985,8 +985,8 @@ static struct snd_pcm_hardware snd_emu10k1_capture_efx = SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .rate_min = 44100, .rate_max = 192000, @@ -1008,7 +1008,7 @@ static void snd_emu10k1_pcm_mixer_notify1(struct snd_emu10k1 *emu, struct snd_kc { struct snd_ctl_elem_id id; - if (! kctl) + if (!kctl) return; if (activate) kctl->vd[idx].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1066,13 +1066,13 @@ static int snd_emu10k1_efx_playback_open(struct snd_pcm_substream *substream) epcm->emu = emu; epcm->type = PLAYBACK_EFX; epcm->substream = substream; - + emu->pcm_playback_efx_substream = substream; runtime->private_data = epcm; runtime->private_free = snd_emu10k1_pcm_free_substream; runtime->hw = snd_emu10k1_efx_playback; - + for (i = 0; i < NUM_EFX_PLAYBACK; i++) { mix = &emu->efx_pcm_mixer[i]; mix->send_routing[0][0] = i; @@ -1243,7 +1243,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) * channels_min = 16, * channels_max = 16, * Need to add mixer control to fix sample rate - * + * * There are 32 mono channels of 16bits each. * 24bit Audio uses 2x channels over 16bit * 96kHz uses 2x channels over 48kHz @@ -1470,7 +1470,7 @@ static int snd_emu10k1_pcm_efx_voices_mask_get(struct snd_kcontrol *kcontrol, st struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); int nefx = emu->audigy ? 64 : 32; int idx; - + spin_lock_irq(&emu->reg_lock); for (idx = 0; idx < nefx; idx++) ucontrol->value.integer.value[idx] = (emu->efx_voices_mask[idx / 32] & (1 << (idx % 32))) ? 1 : 0; @@ -1485,18 +1485,18 @@ static int snd_emu10k1_pcm_efx_voices_mask_put(struct snd_kcontrol *kcontrol, st int nefx = emu->audigy ? 64 : 32; int nefxb = emu->audigy ? 7 : 6; int change, idx; - + nval[0] = nval[1] = 0; for (idx = 0, bits = 0; idx < nefx; idx++) if (ucontrol->value.integer.value[idx]) { nval[idx / 32] |= 1 << (idx % 32); bits++; } - + for (idx = 0; idx < nefxb; idx++) if (1 << idx == bits) break; - + if (idx >= nefxb) return -EINVAL; @@ -1622,7 +1622,7 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number]; unsigned int i; - + /* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */ memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec)); pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */ @@ -1682,7 +1682,7 @@ static int snd_emu10k1_fx8010_playback_trigger(struct snd_pcm_substream *substre result = -EINVAL; break; } - __err: +__err: spin_unlock(&emu->reg_lock); return result; } @@ -1783,10 +1783,10 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s if (rpcm) *rpcm = pcm; - /* EFX capture - record the "FXBUS2" channels, by default we connect the EXTINs + /* EFX capture - record the "FXBUS2" channels, by default we connect the EXTINs * to these - */ - + */ + /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f974..012c0c3 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -46,7 +46,7 @@ static void snd_emu10k1_proc_spdif_status(struct snd_emu10k1 * emu, static char *channel[16] = { "unspec", "left", "right", "3", "4", "5", "6", "7", "8", "9", "10", "11", "12", "13", "14", "15" }; static char *emphasis[8] = { "none", "50/15 usec 2 channel", "2", "3", "4", "5", "6", "7" }; unsigned int status, rate = 0; - + status = snd_emu10k1_ptr_read(emu, status_reg, 0); snd_iprintf(buffer, "\n%s\n", title); @@ -70,7 +70,7 @@ static void snd_emu10k1_proc_spdif_status(struct snd_emu10k1 * emu, snd_iprintf(buffer, "S/PDIF Locked : %s\n", rate & SRCS_SPDIFLOCKED ? "on" : "off"); snd_iprintf(buffer, "Rate Locked : %s\n", rate & SRCS_RATELOCKED ? "on" : "off"); /* From ((Rate * 48000 ) / 262144); */ - snd_iprintf(buffer, "Estimated Sample Rate : %d\n", ((rate & 0xFFFFF ) * 375) >> 11); + snd_iprintf(buffer, "Estimated Sample Rate : %d\n", ((rate & 0xFFFFF ) * 375) >> 11); } } else { snd_iprintf(buffer, "No signal detected.\n"); @@ -78,7 +78,7 @@ static void snd_emu10k1_proc_spdif_status(struct snd_emu10k1 * emu, } -static void snd_emu10k1_proc_read(struct snd_info_entry *entry, +static void snd_emu10k1_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { /* FIXME - output names are in emufx.c too */ @@ -189,7 +189,7 @@ static void snd_emu10k1_proc_read(struct snd_info_entry *entry, int nefx = emu->audigy ? 64 : 32; char **outputs = emu->audigy ? audigy_outs : creative_outs; int idx; - + snd_iprintf(buffer, "EMU10K1\n\n"); snd_iprintf(buffer, "Card : %s\n", emu->audigy ? "Audigy" : (emu->card_capabilities->ecard ? "EMU APS" : "Creative")); @@ -235,7 +235,7 @@ static void snd_emu10k1_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, " Output %02i [%s]\n", idx, outputs[idx]); } -static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, +static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; @@ -253,7 +253,7 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, snd_emu1010_fpga_read(emu, 0x2a, &value); snd_emu1010_fpga_read(emu, 0x2b, &value2); spin_unlock_irqrestore(&emu->emu_lock, flags); - rate = 0x1770000 / (((value << 5) | value2)+1); + rate = 0x1770000 / (((value << 5) | value2)+1); snd_iprintf(buffer, "ADAT Locked : %u\n", rate); } else { snd_iprintf(buffer, "ADAT Unlocked\n"); @@ -266,7 +266,7 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, snd_emu1010_fpga_read(emu, 0x28, &value); snd_emu1010_fpga_read(emu, 0x29, &value2); spin_unlock_irqrestore(&emu->emu_lock, flags); - rate = 0x1770000 / (((value << 5) | value2)+1); + rate = 0x1770000 / (((value << 5) | value2)+1); snd_iprintf(buffer, "SPDIF Locked : %d\n", rate); } else { snd_iprintf(buffer, "SPDIF Unlocked\n"); @@ -283,7 +283,7 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, #endif } -static void snd_emu10k1_proc_rates_read(struct snd_info_entry *entry, +static void snd_emu10k1_proc_rates_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { static int samplerate[8] = { 44100, 48000, 96000, 192000, 4, 5, 6, 7 }; @@ -298,7 +298,7 @@ static void snd_emu10k1_proc_rates_read(struct snd_info_entry *entry, } } -static void snd_emu10k1_proc_acode_read(struct snd_info_entry *entry, +static void snd_emu10k1_proc_acode_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { u32 pc; @@ -308,7 +308,7 @@ static void snd_emu10k1_proc_acode_read(struct snd_info_entry *entry, snd_iprintf(buffer, " Code dump :\n"); for (pc = 0; pc < (emu->audigy ? 1024 : 512); pc++) { u32 low, high; - + low = snd_emu10k1_efx_read(emu, pc * 2); high = snd_emu10k1_efx_read(emu, pc * 2 + 1); if (emu->audigy) @@ -350,7 +350,7 @@ static long snd_emu10k1_fx8010_read(struct snd_info_entry *entry, struct snd_emu10k1 *emu = entry->private_data; unsigned int offset; int tram_addr = 0; - + if (!strcmp(entry->name, "fx8010_tram_addr")) { offset = TANKMEMADDRREGBASE; tram_addr = 1; @@ -374,7 +374,7 @@ static long snd_emu10k1_fx8010_read(struct snd_info_entry *entry, if (tram_addr && emu->audigy) { tmp[idx] = snd_emu10k1_ptr_read(emu, offset + idx + (pos >> 2), 0) >> 11; tmp[idx] |= snd_emu10k1_ptr_read(emu, 0x100 + idx + (pos >> 2), 0) << 20; - } else + } else tmp[idx] = snd_emu10k1_ptr_read(emu, offset + idx + (pos >> 2), 0); if (copy_to_user(buf, ((char *)tmp) + (pos & 3), size)) res = -EFAULT; @@ -387,13 +387,13 @@ static long snd_emu10k1_fx8010_read(struct snd_info_entry *entry, return 0; } -static void snd_emu10k1_proc_voices_read(struct snd_info_entry *entry, +static void snd_emu10k1_proc_voices_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; struct snd_emu10k1_voice *voice; int idx; - + snd_iprintf(buffer, "ch\tuse\tpcm\tefx\tsynth\tmidi\n"); for (idx = 0; idx < NUM_G; idx++) { voice = &emu->voices[idx]; @@ -417,7 +417,7 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, int i; snd_iprintf(buffer, "EMU1010 Registers:\n\n"); - for(i = 0; i < 0x40; i+=1) { + for (i = 0; i < 0x40; i++) { spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, i, &value); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -433,7 +433,7 @@ static void snd_emu_proc_io_reg_read(struct snd_info_entry *entry, unsigned long flags; int i; snd_iprintf(buffer, "IO Registers:\n\n"); - for(i = 0; i < 0x40; i+=4) { + for (i = 0; i < 0x40; i += 4) { spin_lock_irqsave(&emu->emu_lock, flags); value = inl(emu->port + i); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -451,7 +451,7 @@ static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", &reg, &val) != 2) continue; - if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) { + if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff)) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -505,10 +505,10 @@ static void snd_emu_proc_ptr_reg_read(struct snd_info_entry *entry, return; } snd_iprintf(buffer, "Registers 0x%x\n", iobase); - for(i = offset; i < offset+length; i++) { + for (i = offset; i < offset+length; i++) { snd_iprintf(buffer, "%02X: ",i); for (j = 0; j < voices; j++) { - if(iobase == 0) + if (iobase == 0) value = snd_ptr_read(emu, 0, i, j); else value = snd_ptr_read(emu, 0x20, i, j); @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3)) snd_ptr_write(emu, iobase, reg, channel_id, val); } } @@ -543,7 +543,7 @@ static void snd_emu_proc_ptr_reg_write20(struct snd_info_entry *entry, { snd_emu_proc_ptr_reg_write(entry, buffer, 0x20); } - + static void snd_emu_proc_ptr_reg_read00a(struct snd_info_entry *entry, struct snd_info_buffer *buffer) @@ -585,85 +585,85 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) struct snd_info_entry *entry; #ifdef CONFIG_SND_DEBUG if (emu->card_capabilities->emu_model) { - if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry)) + if (!snd_card_proc_new(emu->card, "emu1010_regs", &entry)) snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read); } - if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { + if (!snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; entry->mode |= S_IWUSR; } - if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { + if (!snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; entry->mode |= S_IWUSR; } - if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { + if (!snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; entry->mode |= S_IWUSR; } - if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { + if (!snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } - if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { + if (!snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } - if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { + if (!snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } #endif - - if (! snd_card_proc_new(emu->card, "emu10k1", &entry)) + + if (!snd_card_proc_new(emu->card, "emu10k1", &entry)) snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_read); if (emu->card_capabilities->emu10k2_chip) { - if (! snd_card_proc_new(emu->card, "spdif-in", &entry)) + if (!snd_card_proc_new(emu->card, "spdif-in", &entry)) snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_spdif_read); } if (emu->card_capabilities->ca0151_chip) { - if (! snd_card_proc_new(emu->card, "capture-rates", &entry)) + if (!snd_card_proc_new(emu->card, "capture-rates", &entry)) snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_rates_read); } - if (! snd_card_proc_new(emu->card, "voices", &entry)) + if (!snd_card_proc_new(emu->card, "voices", &entry)) snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_voices_read); - if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { + if (!snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_GPR : TOTAL_SIZE_GPR; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } - if (! snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) { + if (!snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; - entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA ; + entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } - if (! snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) { + if (!snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; - entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR ; + entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } - if (! snd_card_proc_new(emu->card, "fx8010_code", &entry)) { + if (!snd_card_proc_new(emu->card, "fx8010_code", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_CODE : TOTAL_SIZE_CODE; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } - if (! snd_card_proc_new(emu->card, "fx8010_acode", &entry)) { + if (!snd_card_proc_new(emu->card, "fx8010_acode", &entry)) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index b5a802b..c77f5f2 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -42,16 +42,16 @@ unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, un if (reg & 0xff000000) { unsigned char size, offset; - + size = (reg >> 24) & 0x3f; offset = (reg >> 16) & 0x1f; mask = ((1 << size) - 1) << offset; - + spin_lock_irqsave(&emu->emu_lock, flags); outl(regptr, emu->port + PTR); val = inl(emu->port + DATA); spin_unlock_irqrestore(&emu->emu_lock, flags); - + return (val & mask) >> offset; } else { spin_lock_irqsave(&emu->emu_lock, flags); @@ -90,7 +90,7 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i outl(regptr, emu->port + PTR); data |= inl(emu->port + DATA) & ~mask; outl(data, emu->port + DATA); - spin_unlock_irqrestore(&emu->emu_lock, flags); + spin_unlock_irqrestore(&emu->emu_lock, flags); } else { spin_lock_irqsave(&emu->emu_lock, flags); outl(regptr, emu->port + PTR); @@ -101,13 +101,13 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i EXPORT_SYMBOL(snd_emu10k1_ptr_write); -unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, - unsigned int reg, +unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, + unsigned int reg, unsigned int chn) { unsigned long flags; unsigned int regptr, val; - + regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); @@ -117,9 +117,9 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, return val; } -void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, - unsigned int reg, - unsigned int chn, +void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, + unsigned int reg, + unsigned int chn, unsigned int data) { unsigned int regptr; @@ -230,7 +230,7 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, break; } } - //Read back and see if the transaction is successful + /* Read back and see if the transaction is successful */ if ((status & I2C_A_ADC_ABORT) == 0) break; } @@ -242,7 +242,7 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, /* dump_stack(); */ err = -EINVAL; } - + spin_unlock(&emu->i2c_lock); return err; } @@ -291,10 +291,10 @@ int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value) */ int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src) { - snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) ); - snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) ); - snd_emu1010_fpga_write(emu, 0x02, ((src >> 8) & 0x3f) ); - snd_emu1010_fpga_write(emu, 0x03, (src & 0x3f) ); + snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f)); + snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f)); + snd_emu1010_fpga_write(emu, 0x02, ((src >> 8) & 0x3f)); + snd_emu1010_fpga_write(emu, 0x03, (src & 0x3f)); return 0; } diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index 30bfed6..e4947a6 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -157,19 +157,19 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) struct snd_emu10k1_voice *pvoice = &(emu->p16v_voices[0]); struct snd_emu10k1_voice *cvoice = &(emu->p16v_capture_voice); - //printk(KERN_INFO "status2=0x%x\n", status2); + /* printk(KERN_INFO "status2=0x%x\n", status2); */ orig_status2 = status2; - if(status2 & mask) { - if(pvoice->use) { + if (status2 & mask) { + if (pvoice->use) { snd_pcm_period_elapsed(pvoice->epcm->substream); - } else { + } else { snd_printk(KERN_ERR "p16v: status: 0x%08x, mask=0x%08x, pvoice=%p, use=%d\n", status2, mask, pvoice, pvoice->use); } } - if(status2 & 0x110000) { - //printk(KERN_INFO "capture int found\n"); - if(cvoice->use) { - //printk(KERN_INFO "capture period_elapsed\n"); + if (status2 & 0x110000) { + /* printk(KERN_INFO "capture int found\n"); */ + if (cvoice->use) { + /* printk(KERN_INFO "capture period_elapsed\n"); */ snd_pcm_period_elapsed(cvoice->epcm->substream); } } @@ -181,7 +181,7 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) if (status) { unsigned int bits; snd_printk(KERN_ERR "emu10k1: unhandled interrupt: 0x%08x\n", status); - //make sure any interrupts we don't handle are disabled: + /* make sure any interrupts we don't handle are disabled: */ bits = INTE_FXDSPENABLE | INTE_PCIERRORENABLE | INTE_VOLINCRENABLE | diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 6a47672..e6067f6 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -323,7 +323,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst unsigned long ofs = idx << PAGE_SHIFT; dma_addr_t addr; addr = snd_pcm_sgbuf_get_addr(substream, ofs); - if (! is_valid_page(emu, addr)) { + if (!is_valid_page(emu, addr)) { printk(KERN_ERR "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); return NULL; @@ -368,7 +368,7 @@ struct snd_util_memblk * snd_emu10k1_synth_alloc(struct snd_emu10k1 *hw, unsigned int size) { struct snd_emu10k1_memblk *blk; - struct snd_util_memhdr *hdr = hw->memhdr; + struct snd_util_memhdr *hdr = hw->memhdr; mutex_lock(&hdr->block_mutex); blk = (struct snd_emu10k1_memblk *)__snd_util_mem_alloc(hdr, size); @@ -394,7 +394,7 @@ EXPORT_SYMBOL(snd_emu10k1_synth_alloc); int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk) { - struct snd_util_memhdr *hdr = emu->memhdr; + struct snd_util_memhdr *hdr = emu->memhdr; struct snd_emu10k1_memblk *blk = (struct snd_emu10k1_memblk *)memblk; unsigned long flags; @@ -499,7 +499,7 @@ static inline void *offset_ptr(struct snd_emu10k1 *emu, int page, int offset) if (snd_BUG_ON(page < 0 || page >= emu->max_cache_pages)) return NULL; ptr = emu->page_ptr_table[page]; - if (! ptr) { + if (!ptr) { printk(KERN_ERR "emu10k1: access to NULL ptr: page = %d\n", page); return NULL; } diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 749a21b..dd71b65 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -123,17 +123,17 @@ /* hardware definition */ static struct snd_pcm_hardware snd_p16v_playback_hw = { - .info = SNDRV_PCM_INFO_MMAP | + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */ - .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100, + .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100, .rate_min = 44100, .rate_max = 192000, - .channels_min = 8, + .channels_min = 8, .channels_max = 8, .buffer_bytes_max = ((65536 - 64) * 8), .period_bytes_min = 64, @@ -150,7 +150,7 @@ static struct snd_pcm_hardware snd_p16v_capture_hw = { SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S32_LE, - .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100, + .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100, .rate_min = 44100, .rate_max = 192000, .channels_min = 2, @@ -166,9 +166,9 @@ static struct snd_pcm_hardware snd_p16v_capture_hw = { static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime) { struct snd_emu10k1_pcm *epcm = runtime->private_data; - + if (epcm) { - //snd_printk("epcm free: %p\n", epcm); + /* snd_printk("epcm free: %p\n", epcm); */ kfree(epcm); } } @@ -183,26 +183,26 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk("epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); */ + runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; - + runtime->hw = snd_p16v_playback_hw; channel->emu = emu; channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; + /* snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); */ + /* printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ channel->epcm=epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -224,26 +224,26 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk("epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); + /* snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; - + runtime->hw = snd_p16v_capture_hw; channel->emu = emu; channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; + /* snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); */ + /* printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ channel->epcm=epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -256,8 +256,8 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream static int snd_p16v_pcm_close_playback(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - //struct snd_pcm_runtime *runtime = substream->runtime; - //struct snd_emu10k1_pcm *epcm = runtime->private_data; + /* struct snd_pcm_runtime *runtime = substream->runtime; */ + /* struct snd_emu10k1_pcm *epcm = runtime->private_data; */ emu->p16v_voices[substream->pcm->device - emu->p16v_device_offset].use = 0; /* FIXME: maybe zero others */ return 0; @@ -267,8 +267,8 @@ static int snd_p16v_pcm_close_playback(struct snd_pcm_substream *substream) static int snd_p16v_pcm_close_capture(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - //struct snd_pcm_runtime *runtime = substream->runtime; - //struct snd_emu10k1_pcm *epcm = runtime->private_data; + /* struct snd_pcm_runtime *runtime = substream->runtime; */ + /* struct snd_emu10k1_pcm *epcm = runtime->private_data; */ emu->p16v_capture_voice.use = 0; /* FIXME: maybe zero others */ return 0; @@ -281,7 +281,7 @@ static int snd_p16v_pcm_open_playback_front(struct snd_pcm_substream *substream) static int snd_p16v_pcm_open_capture(struct snd_pcm_substream *substream) { - // Only using channel 0 for now, but the card has 2 channels. + /* Only using channel 0 for now, but the card has 2 channels. */ return snd_p16v_pcm_open_capture_channel(substream, 0); } @@ -333,10 +333,10 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); int i; u32 tmp; - - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes); + + /* snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); */ + /* snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); */ + /* snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes); */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -354,17 +354,17 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) break; } /* FIXME: Check emu->buffer.size before actually writing to it. */ - for(i = 0; i < runtime->periods; i++) { + for (i = 0; i < runtime->periods; i++) { table_base[i*2]=runtime->dma_addr+(i*period_size_bytes); table_base[(i*2)+1]=period_size_bytes<<16; } - + snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_ADDR, channel, emu->p16v_buffer.addr+(8*16*channel)); snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19); snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_PTR, channel, 0); snd_emu10k1_ptr20_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr); - //snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes - snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes + /* snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); buffer size in bytes */ + snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); /* buffer size in bytes */ snd_emu10k1_ptr20_write(emu, PLAYBACK_POINTER, channel, 0); snd_emu10k1_ptr20_write(emu, 0x07, channel, 0x0); snd_emu10k1_ptr20_write(emu, 0x08, channel, 0); @@ -379,7 +379,7 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int channel = substream->pcm->device - emu->p16v_device_offset; u32 tmp; - //printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + /* printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -399,10 +399,10 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) /* FIXME: Check emu->buffer.size before actually writing to it. */ snd_emu10k1_ptr20_write(emu, 0x13, channel, 0); snd_emu10k1_ptr20_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); - snd_emu10k1_ptr20_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size) << 16); // buffer size in bytes + snd_emu10k1_ptr20_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size) << 16); /* buffer size in bytes */ snd_emu10k1_ptr20_write(emu, CAPTURE_POINTER, channel, 0); - //snd_emu10k1_ptr20_write(emu, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC or Line in */ - //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel)); + /* snd_emu10k1_ptr20_write(emu, CAPTURE_SOURCE, 0x0, 0x333300e4); Select MIC or Line in */ + /* snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel)); */ return 0; } @@ -459,13 +459,13 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; - //snd_printk("p16v channel=%d\n",channel); + /* snd_printk("p16v channel=%d\n",channel); */ epcm->running = running; basic |= (0x1<<channel); inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel); snd_pcm_trigger_done(s, substream); } - //snd_printk("basic=0x%x, inte=0x%x\n",basic, inte); + /* snd_printk("basic=0x%x, inte=0x%x\n",basic, inte); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -503,7 +503,7 @@ static int snd_p16v_pcm_trigger_capture(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel)); snd_p16v_intr_disable(emu, inte); - //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel)); + /* snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel)); */ epcm->running = 0; break; default: @@ -558,7 +558,7 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr -= runtime->buffer_size; printk(KERN_WARNING "buffer capture limited!\n"); } - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + /* printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); */ return ptr; } @@ -589,10 +589,10 @@ static struct snd_pcm_ops snd_p16v_capture_ops = { int snd_p16v_free(struct snd_emu10k1 *chip) { - // release the data + /* release the data */ if (chip->p16v_buffer.area) { snd_dma_free_pages(&chip->p16v_buffer); - //snd_printk("period lables free: %p\n", &chip->p16v_buffer); + /* snd_printk("period lables free: %p\n", &chip->p16v_buffer); */ } return 0; } @@ -603,18 +603,18 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * struct snd_pcm_substream *substream; int err; int capture=1; - - //snd_printk("snd_p16v_pcm called. device=%d\n", device); + + /* snd_printk("snd_p16v_pcm called. device=%d\n", device); */ emu->p16v_device_offset = device; if (rpcm) *rpcm = NULL; if ((err = snd_pcm_new(emu->card, "p16v", device, 1, capture, &pcm)) < 0) return err; - + pcm->private_data = emu; - // Single playback 8 channel device. - // Single capture 2 channel device. + /* Single playback 8 channel device. */ + /* Single capture 2 channel device. */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_p16v_playback_front_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_p16v_capture_ops); @@ -623,31 +623,31 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * strcpy(pcm->name, "p16v"); emu->pcm_p16v = pcm; - for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - substream; + for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream; substream = substream->next) { - if ((err = snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), - ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) return err; - //snd_printk("preallocate playback substream: err=%d\n", err); + /* snd_printk("preallocate playback substream: err=%d\n", err); */ } - for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - substream; + for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream; substream = substream->next) { - if ((err = snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), 65536 - 64, 65536 - 64)) < 0) return err; - //snd_printk("preallocate capture substream: err=%d\n", err); + /* snd_printk("preallocate capture substream: err=%d\n", err); */ } - + if (rpcm) *rpcm = pcm; - + return 0; } @@ -696,7 +696,7 @@ static int snd_p16v_volume_put(struct snd_kcontrol *kcontrol, } else { value &= 0xffff0000; value |= ((0xff - ucontrol->value.integer.value[0]) << 8) | - ((0xff - ucontrol->value.integer.value[1]) ); + ((0xff - ucontrol->value.integer.value[1])); } if (value != oval) { snd_emu10k1_ptr20_write(emu, reg, 0, value); @@ -740,7 +740,7 @@ static int snd_p16v_capture_source_put(struct snd_kcontrol *kcontrol, u32 mask; u32 source; - val = ucontrol->value.enumerated.item[0] ; + val = ucontrol->value.enumerated.item[0]; if (val > 7) return -EINVAL; change = (emu->p16v_capture_source != val); @@ -784,7 +784,7 @@ static int snd_p16v_capture_channel_put(struct snd_kcontrol *kcontrol, int change = 0; u32 tmp; - val = ucontrol->value.enumerated.item[0] ; + val = ucontrol->value.enumerated.item[0]; if (val > 3) return -EINVAL; change = (emu->p16v_capture_channel != val); @@ -854,7 +854,7 @@ int __devinit snd_p16v_mixer(struct snd_emu10k1 *emu) int __devinit snd_p16v_alloc_pm_buffer(struct snd_emu10k1 *emu) { emu->p16v_saved = vmalloc(NUM_CHS * 4 * 0x80); - if (! emu->p16v_saved) + if (!emu->p16v_saved) return -ENOMEM; return 0; } diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h index 1532149..7964c8e 100644 --- a/sound/pci/emu10k1/p16v.h +++ b/sound/pci/emu10k1/p16v.h @@ -81,14 +81,14 @@ /********************************************************************************************************/ /* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */ /********************************************************************************************************/ - + /* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE. * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters. */ /* Initally all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ - /* One list entry: 4 bytes for DMA address, + /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. * One list entry is 8 bytes long. * One list entry for each period in the buffer. @@ -167,11 +167,11 @@ * 3 - 192 khz * 7 - undefined rate. * [19] Channel 0. 1 - Valid, 0 - Not Valid. - * [22:20] Channel 1 Detected sample rate. + * [22:20] Channel 1 Detected sample rate. * [23] Channel 1. 1 - Valid, 0 - Not Valid. - * [26:24] Channel 2 Detected sample rate. + * [26:24] Channel 2 Detected sample rate. * [27] Channel 2. 1 - Valid, 0 - Not Valid. - * [30:28] Channel 3 Detected sample rate. + * [30:28] Channel 3 Detected sample rate. * [31] Channel 3. 1 - Valid, 0 - Not Valid. */ /* 0x18 - 0x1f unused */ @@ -204,7 +204,7 @@ /* 0x43,0x48 do not remember settings */ /* 0x41-45 unused */ #define WATERMARK 0x46 /* Test bit to indicate cache level usage */ - /* Values it can have while playing on channel 0. + /* Values it can have while playing on channel 0. * 0000f000, 0000f004, 0000f008, 0000f00c. * Readonly. */ @@ -253,7 +253,7 @@ * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled. * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled. */ - /* Bypass P16V 0xff00ff00 + /* Bypass P16V 0xff00ff00 * Bitmap. 0 = Off, 1 = On. * P16V playback outputs: * 0xXXXXXXX1 = PCM0 Left. (Front) diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h index 4ef5f68..0a28974 100644 --- a/sound/pci/emu10k1/p17v.h +++ b/sound/pci/emu10k1/p17v.h @@ -27,11 +27,11 @@ /* 00 - 07: Not used */ #define P17V_PLAYBACK_FIFO_PTR 0x08 /* Current playback fifo pointer * and number of sound samples in cache. - */ + */ /* 09 - 12: Not used */ #define P17V_CAPTURE_FIFO_PTR 0x13 /* Current capture fifo pointer * and number of sound samples in cache. - */ + */ /* 14 - 17: Not used */ #define P17V_PB_CHN_SEL 0x18 /* P17v playback channel select */ #define P17V_SE_SLOT_SEL_L 0x19 /* Sound Engine slot select low */ @@ -58,7 +58,7 @@ #define I2C_A_ADC_LAST 0x00000400 /*I2C last transaction */ #define I2C_A_ADC_BYTE 0x00000800 /*I2C one byte mode */ -#define I2C_D_ADC_REG_MASK 0xfe000000 /*ADC address register */ +#define I2C_D_ADC_REG_MASK 0xfe000000 /*ADC address register */ #define I2C_D_ADC_DAT_MASK 0x01ff0000 /*ADC data register */ #define ADC_TIMEOUT 0x00000007 /*ADC Timeout Clock Disable */ diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index d7300a1..de271d0 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -5,7 +5,7 @@ * Routines for control of EMU10K1 chips - voice manager * * Rewrote voice allocator for multichannel support - rlrevell 12/2004 - * + * * BUGS: * -- * @@ -32,14 +32,14 @@ #include <sound/core.h> #include <sound/emu10k1.h> -/* Previously the voice allocator started at 0 every time. The new voice - * allocator uses a round robin scheme. The next free voice is tracked in - * the card record and each allocation begins where the last left off. The - * hardware requires stereo interleaved voices be aligned to an even/odd - * boundary. For multichannel voice allocation we ensure than the block of - * voices does not cross the 32 voice boundary. This simplifies the - * multichannel support and ensures we can use a single write to the - * (set|clear)_loop_stop registers. Otherwise (for example) the voices would +/* Previously the voice allocator started at 0 every time. The new voice + * allocator uses a round robin scheme. The next free voice is tracked in + * the card record and each allocation begins where the last left off. The + * hardware requires stereo interleaved voices be aligned to an even/odd + * boundary. For multichannel voice allocation we ensure than the block of + * voices does not cross the 32 voice boundary. This simplifies the + * multichannel support and ensures we can use a single write to the + * (set|clear)_loop_stop registers. Otherwise (for example) the voices would * get out of sync when pausing/resuming a stream. * --rlrevell */ @@ -52,8 +52,8 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, *rvoice = NULL; first_voice = last_voice = 0; - for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) { - // printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice); + for (i = emu->next_free_voice, j = 0; j < NUM_G; i += number, j += number) { + /* printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice); */ i %= NUM_G; /* stereo voices must be even/odd */ @@ -61,7 +61,7 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, i++; continue; } - + skip = 0; for (k = 0; k < number; k++) { voice = &emu->voices[(i+k) % NUM_G]; @@ -71,20 +71,20 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, } } if (!skip) { - // printk("allocated voice %d\n", i); + /* printk("allocated voice %d\n", i); */ first_voice = i; last_voice = (i + number) % NUM_G; emu->next_free_voice = last_voice; break; } } - + if (first_voice == last_voice) return -ENOMEM; - + for (i = 0; i < number; i++) { voice = &emu->voices[(first_voice + i) % NUM_G]; - // printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number); + /* printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number); */ voice->use = 1; switch (type) { case EMU10K1_PCM:

1 0

[alsa-devel] [PATCH 07/24] sound/pci: coding style fixes: echoaudio
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/echoaudio/echoaudio.h | 2 +- sound/pci/echoaudio/echoaudio_dsp.h | 16 ++++++++-------- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/layla24_dsp.c | 4 ++-- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/midi.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/echoaudio/mona_dsp.c | 2 +- 18 files changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index 8c6db3a..c9a128a 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -51,7 +51,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 04cbf3e..3771d02 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -55,7 +55,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 4022e43..eb9bf6e 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -63,7 +63,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> #include <sound/rawmidi.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 8dbc5c4..3004a58 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1098,7 +1098,7 @@ static struct snd_kcontrol_new snd_echo_line_input_gain __devinitdata = { #ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL /************ Analog output nominal level (+4dBu / -10dBV) ***************/ -static int snd_echo_output_nominal_info (struct snd_kcontrol *kcontrol, +static int snd_echo_output_nominal_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct echoaudio *chip; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 1c88e05..674ebf4 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -218,7 +218,7 @@ /**************************************************************************** - + Clocks *****************************************************************************/ diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index e352f3a..09ddbd3 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -103,7 +103,7 @@ #define CHI32_STATUS_IRQ 0x00000040 -/* +/* * * DSP commands sent via slave mode; these are sent to the DSP by write_dsp() * @@ -369,7 +369,7 @@ SET_LAYLA24_FREQUENCY_REG command. * via input and output pipes. LE means little-endian, * BE means big-endian. * - * DSP_AUDIOFORM_MS_8 + * DSP_AUDIOFORM_MS_8 * * 8-bit mono unsigned samples. For playback, * mono data is duplicated out the left and right channels @@ -388,10 +388,10 @@ SET_LAYLA24_FREQUENCY_REG command. * they would be stored in memory like this: 33 22 11 66 55 44. * * DSP_AUDIOFORM_MS_32LE - * - * 24-bit signed little-endian mono samples in a 32-bit - * container. In other words, each sample is a 32-bit signed - * integer, where the actual audio data is left-justified + * + * 24-bit signed little-endian mono samples in a 32-bit + * container. In other words, each sample is a 32-bit signed + * integer, where the actual audio data is left-justified * in the 32 bits and only the 24 most significant bits are valid. * * DSP_AUDIOFORM_SS_8 @@ -435,8 +435,8 @@ SET_LAYLA24_FREQUENCY_REG command. * Super-interleave is defined as interleaving by 4 or more. Darla20 and Gina20 * do not support super interleave. * - * 16 bit, 24 bit, and 32 bit little endian samples are supported for super - * interleave. The interleave factor must be even. 16 - way interleave is the + * 16 bit, 24 bit, and 32 bit little endian samples are supported for super + * interleave. The interleave factor must be even. 16 - way interleave is the * current maximum, so you can interleave by 4, 6, 8, 10, 12, 14, and 16. * * The actual format code is derived by taking the define below and or-ing with diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index c0e64b8..fa70fb1 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -55,7 +55,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index c36a78d..a33cc5a 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -61,7 +61,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 0a58a7c..9097281 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -53,7 +53,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 2db24d2..b3ff9a2 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -53,7 +53,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index a60c0a0..bd0c403 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -54,7 +54,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 5061946..4a32154 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -61,7 +61,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> #include <sound/rawmidi.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index e09e3ea..a9b0ec0 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -63,7 +63,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> #include <sound/rawmidi.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index d61b5cb..03a3b38 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -149,7 +149,7 @@ static int load_asic(struct echoaudio *chip) if (!err) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - + DE_INIT(("load_asic() done\n")); return err; } @@ -307,7 +307,7 @@ static int switch_asic(struct echoaudio *chip, const struct firmware *asic) if (asic != chip->asic_code) { monitors = kmemdup(chip->comm_page->monitors, MONITOR_ARRAY_SIZE, GFP_KERNEL); - if (! monitors) + if (!monitors) return -ENOMEM; memset(chip->comm_page->monitors, ECHOGAIN_MUTED, diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index f3b9b45..021b926 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -61,7 +61,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> #include <sound/rawmidi.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 77bf2a8..837d239 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -66,7 +66,7 @@ static int write_midi(struct echoaudio *chip, u8 *data, int bytes) return -EIO; /* HF4 indicates that it is safe to write MIDI output data */ - if (! (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_REG_HF4)) + if (!(get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_REG_HF4)) return 0; chip->comm_page->midi_output[0] = bytes; diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index b05bad9..ead9ed4 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -59,7 +59,7 @@ #include <sound/pcm_params.h> #include <sound/asoundef.h> #include <sound/initval.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/atomic.h> #include "echoaudio.h" diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index eaa619b..dc99bc0 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -114,7 +114,7 @@ static u32 detect_input_clocks(const struct echoaudio *chip) -/* Mona has an ASIC on the PCI card and another ASIC in the external box; +/* Mona has an ASIC on the PCI card and another ASIC in the external box; both need to be loaded. */ static int load_asic(struct echoaudio *chip) {

1 0

[alsa-devel] [PATCH 06/24] sound/pci: coding style fixes: sound/pci/c*
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/bt87x.c | 4 +- sound/pci/cmipci.c | 80 +++++++++++++++--------------- sound/pci/cs4281.c | 82 ++++++++++++++++---------------- sound/pci/cs5530.c | 8 ++-- sound/pci/cs5535audio/cs5535audio.c | 8 ++-- sound/pci/cs5535audio/cs5535audio_pm.c | 12 ++-- 6 files changed, 97 insertions(+), 97 deletions(-) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d97..e9a4fad 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -27,7 +27,7 @@ #include <linux/slab.h> #include <linux/moduleparam.h> #include <linux/bitops.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -603,7 +603,7 @@ static int snd_bt87x_capture_boost_get(struct snd_kcontrol *kcontrol, { struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol); - value->value.integer.value[0] = !! (chip->reg_control & CTL_A_G2X); + value->value.integer.value[0] = !!(chip->reg_control & CTL_A_G2X); return 0; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1a74ca6..411ff65 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -16,11 +16,11 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - + /* Does not work. Warning may block system in capture mode */ /* #define USE_VAR48KRATE */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -125,7 +125,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_SPD24SEL 0x00020000 /* 24bit spdif: model 037 */ /* #define CM_SPDIF_INVERSE 0x00010000 */ /* ??? */ -#define CM_ADCBITLEN_MASK 0x0000C000 +#define CM_ADCBITLEN_MASK 0x0000C000 #define CM_ADCBITLEN_16 0x00000000 #define CM_ADCBITLEN_15 0x00004000 #define CM_ADCBITLEN_14 0x00008000 @@ -255,7 +255,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_UPDDMA_2048 0x00000000 #define CM_UPDDMA_1024 0x00000004 #define CM_UPDDMA_512 0x00000008 -#define CM_UPDDMA_256 0x0000000C +#define CM_UPDDMA_256 0x0000000C #define CM_TWAIT_MASK 0x00000003 /* model 037 */ #define CM_TWAIT1 0x00000002 /* FM i/o cycle, 0: 48, 1: 64 PCICLKs */ #define CM_TWAIT0 0x00000001 /* i/o cycle, 0: 4, 1: 6 PCICLKs */ @@ -450,7 +450,7 @@ static const struct cmipci_mixer_auto_switches cm_saved_mixer[] = { {"PCM Playback Switch", 0}, {"IEC958 Output Switch", 1}, {"IEC958 Mix Analog", 0}, - // {"IEC958 Out To DAC", 1}, // no longer used + /* {"IEC958 Out To DAC", 1}, // no longer used */ {"IEC958 Loop", 0}, }; #define CM_SAVED_MIXERS ARRAY_SIZE(cm_saved_mixer) @@ -827,7 +827,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, else cm->ctrl |= val; snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl); - //snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl); + /* snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl); */ /* set sample rate */ freq = 0; @@ -850,7 +850,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK; } snd_cmipci_write(cm, CM_REG_FUNCTRL1, val); - //snd_printd("cmipci: functrl1 = %08x\n", val); + /* snd_printd("cmipci: functrl1 = %08x\n", val); */ /* set format */ val = snd_cmipci_read(cm, CM_REG_CHFORMAT); @@ -866,7 +866,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec, val |= freq_ext << (rec->ch * 2); } snd_cmipci_write(cm, CM_REG_CHFORMAT, val); - //snd_printd("cmipci: chformat = %08x\n", val); + /* snd_printd("cmipci: chformat = %08x\n", val); */ if (!rec->is_dac && cm->chip_version) { if (runtime->rate > 44100) @@ -904,7 +904,7 @@ static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec, cm->ctrl |= chen; /* enable channel */ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl); - //snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl); + /* snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl); */ break; case SNDRV_PCM_TRIGGER_STOP: rec->running = 0; @@ -944,7 +944,7 @@ static snd_pcm_uframes_t snd_cmipci_pcm_pointer(struct cmipci *cm, struct cmipci unsigned int reg; if (!rec->running) return 0; -#if 1 // this seems better.. +#if 1 /* this seems better.. */ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2; ptr = rec->dma_size - (snd_cmipci_read_w(cm, reg) + 1); ptr >>= rec->shift; @@ -1127,7 +1127,7 @@ static struct snd_kcontrol_new snd_cmipci_spdif_stream __devinitdata = /* save mixer setting and mute for AC3 playback */ static int save_mixer_state(struct cmipci *cm) { - if (! cm->mixer_insensitive) { + if (!cm->mixer_insensitive) { struct snd_ctl_elem_value *val; unsigned int i; @@ -1199,7 +1199,7 @@ static void setup_ac3(struct cmipci *cm, struct snd_pcm_substream *subs, int do_ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_AC3EN1); /* AC3EN for 039 */ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_AC3EN2); - + if (cm->can_ac3_hw) { /* SPD24SEL for 037, 0x02 */ /* SPD24SEL for 039, 0x20, but cannot be set */ @@ -1294,7 +1294,7 @@ static int snd_cmipci_playback_prepare(struct snd_pcm_substream *substream) do_spdif = (rate >= 44100 && rate <= 96000 && substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE && substream->runtime->channels == 2); - if (do_spdif && cm->can_ac3_hw) + if (do_spdif && cm->can_ac3_hw) do_ac3 = cm->dig_pcm_status & IEC958_AES0_NONAUDIO; if ((err = setup_spdif_playback(cm, substream, do_spdif, do_ac3)) < 0) return err; @@ -1307,7 +1307,7 @@ static int snd_cmipci_playback_spdif_prepare(struct snd_pcm_substream *substream struct cmipci *cm = snd_pcm_substream_chip(substream); int err, do_ac3; - if (cm->can_ac3_hw) + if (cm->can_ac3_hw) do_ac3 = cm->dig_pcm_status & IEC958_AES0_NONAUDIO; else do_ac3 = 1; /* doesn't matter */ @@ -1335,7 +1335,7 @@ static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec) reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2; val = ((PAGE_SIZE / 4) - 1) | (((PAGE_SIZE / 4) / 2 - 1) << 16); snd_cmipci_write(cm, reg, val); - + /* configure for 16 bits, 2 channels, 8 kHz */ if (runtime->channels > 2) set_dac_channels(cm, rec, 2); @@ -1350,7 +1350,7 @@ static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec) if (cm->can_96k) val &= ~(CM_CH0_SRATE_MASK << (rec->ch * 2)); snd_cmipci_write(cm, CM_REG_CHFORMAT, val); - + /* start stream (we don't need interrupts) */ cm->ctrl |= CM_CHEN0 << rec->ch; snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl); @@ -1436,7 +1436,7 @@ static irqreturn_t snd_cmipci_interrupt(int irq, void *dev_id) { struct cmipci *cm = dev_id; unsigned int status, mask = 0; - + /* fastpath out, to ease interrupt sharing */ status = snd_cmipci_read(cm, CM_REG_INT_STATUS); if (!(status & CM_INTR)) @@ -1616,7 +1616,7 @@ static int open_device_check(struct cmipci *cm, int mode, struct snd_pcm_substre } cm->opened[ch] = mode; cm->channel[ch].substream = subs; - if (! (mode & CM_OPEN_DAC)) { + if (!(mode & CM_OPEN_DAC)) { /* disable dual DAC mode */ cm->channel[ch].is_dac = 0; spin_lock_irq(&cm->reg_lock); @@ -1639,7 +1639,7 @@ static void close_device_check(struct cmipci *cm, int mode) cm->channel[ch].substream = NULL; } cm->opened[ch] = 0; - if (! cm->channel[ch].is_dac) { + if (!cm->channel[ch].is_dac) { /* enable dual DAC mode again */ cm->channel[ch].is_dac = 1; spin_lock_irq(&cm->reg_lock); @@ -1688,7 +1688,7 @@ static int snd_cmipci_capture_open(struct snd_pcm_substream *substream) if ((err = open_device_check(cm, CM_OPEN_CAPTURE, substream)) < 0) return err; runtime->hw = snd_cmipci_capture; - if (cm->chip_version == 68) { // 8768 only supports 44k/48k recording + if (cm->chip_version == 68) { /* 8768 only supports 44k/48k recording */ runtime->hw.rate_min = 41000; runtime->hw.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; } else if (cm->chip_version == 55) { @@ -1713,7 +1713,7 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) return err; runtime->hw = snd_cmipci_playback2; mutex_lock(&cm->open_mutex); - if (! cm->opened[CM_CH_PLAY]) { + if (!cm->opened[CM_CH_PLAY]) { if (cm->can_multi_ch) { runtime->hw.channels_max = cm->max_channels; if (cm->max_channels == 4) @@ -2030,7 +2030,7 @@ static int snd_cmipci_info_volume(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = reg.mask; return 0; } - + static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2112,7 +2112,7 @@ static int snd_cmipci_info_input_sw(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 1; return 0; } - + static int snd_cmipci_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2260,7 +2260,7 @@ static int snd_cmipci_put_native_mixer(struct snd_kcontrol *kcontrol, static int snd_cmipci_get_native_mixer_sensitive(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - //struct cmipci *cm = snd_kcontrol_chip(kcontrol); + /* struct cmipci *cm = snd_kcontrol_chip(kcontrol); */ return snd_cmipci_get_native_mixer(kcontrol, ucontrol); } @@ -2280,7 +2280,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_SB_VOL_STEREO("Master Playback Volume", SB_DSP4_MASTER_DEV, 3, 31), CMIPCI_MIXER_SW_MONO("3D Control - Switch", CM_REG_MIXER1, CM_X3DEN_SHIFT, 0), CMIPCI_SB_VOL_STEREO("PCM Playback Volume", SB_DSP4_PCM_DEV, 3, 31), - //CMIPCI_MIXER_SW_MONO("PCM Playback Switch", CM_REG_MIXER1, CM_WSMUTE_SHIFT, 1), + /* CMIPCI_MIXER_SW_MONO("PCM Playback Switch", CM_REG_MIXER1, CM_WSMUTE_SHIFT, 1), */ { /* switch with sensitivity */ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", @@ -2380,7 +2380,7 @@ static int _snd_cmipci_uswitch_put(struct snd_kcontrol *kcontrol, val = inb(cm->iobase + args->reg); else val = snd_cmipci_read(cm, args->reg); - change = (val & args->mask) != (ucontrol->value.integer.value[0] ? + change = (val & args->mask) != (ucontrol->value.integer.value[0] ? args->mask_on : (args->mask & ~args->mask_on)); if (change) { val &= ~args->mask; @@ -2415,7 +2415,7 @@ static struct cmipci_switch_args cmipci_switch_arg_##sname = { \ .is_byte = xis_byte, \ .ac3_sensitive = xac3, \ } - + #define DEFINE_BIT_SWITCH_ARG(sname, xreg, xmask, xis_byte, xac3) \ DEFINE_SWITCH_ARG(sname, xreg, xmask, xmask, xis_byte, xac3) @@ -2572,7 +2572,7 @@ static int snd_cmipci_mic_in_mode_get(struct snd_kcontrol *kcontrol, struct cmipci *cm = snd_kcontrol_chip(kcontrol); /* same bit as spdi_phase */ spin_lock_irq(&cm->reg_lock); - ucontrol->value.enumerated.item[0] = + ucontrol->value.enumerated.item[0] = (snd_cmipci_read_b(cm, CM_REG_MISC) & CM_SPDIF_INVERSE) ? 1 : 0; spin_unlock_irq(&cm->reg_lock); return 0; @@ -2616,7 +2616,7 @@ static struct snd_kcontrol_new snd_cmipci_8738_mixer_switches[] __devinitdata = DEFINE_MIXER_SWITCH("IEC958 Out", spdif_out), DEFINE_MIXER_SWITCH("IEC958 Out To DAC", spdo2dac), #endif - // DEFINE_MIXER_SWITCH("IEC958 Output Switch", spdif_enable), + /* DEFINE_MIXER_SWITCH("IEC958 Output Switch", spdif_enable), */ { .name = "IEC958 Output Switch", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_cmipci_uswitch_info, @@ -2676,7 +2676,7 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic spin_unlock_irq(&cm->reg_lock); for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_mixers); idx++) { - if (cm->chip_version == 68) { // 8768 has no PCM volume + if (cm->chip_version == 68) { /* 8768 has no PCM volume */ if (!strcmp(snd_cmipci_mixers[idx].name, "PCM Playback Volume")) continue; @@ -2692,7 +2692,7 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic if (err < 0) return err; } - if (! cm->can_multi_ch) { + if (!cm->can_multi_ch) { err = snd_ctl_add(cm->card, snd_ctl_new1(&snd_cmipci_nomulti_switch, cm)); if (err < 0) return err; @@ -2766,12 +2766,12 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic */ #ifdef CONFIG_PROC_FS -static void snd_cmipci_proc_read(struct snd_info_entry *entry, +static void snd_cmipci_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct cmipci *cm = entry->private_data; int i, v; - + snd_iprintf(buffer, "%s\n", cm->card->longname); for (i = 0; i < 0x94; i++) { if (i == 0x28) @@ -2788,7 +2788,7 @@ static void __devinit snd_cmipci_proc_init(struct cmipci *cm) { struct snd_info_entry *entry; - if (! snd_card_proc_new(cm->card, "cmipci", &entry)) + if (!snd_card_proc_new(cm->card, "cmipci", &entry)) snd_info_set_text_ops(entry, cm, snd_cmipci_proc_read); } #else /* !CONFIG_PROC_FS */ @@ -2816,7 +2816,7 @@ static void __devinit query_chip(struct cmipci *cm) /* check reg 0Ch, bit 24-31 */ detect = snd_cmipci_read(cm, CM_REG_INT_HLDCLR) & CM_CHIP_MASK2; - if (! detect) { + if (!detect) { /* check reg 08h, bit 24-28 */ detect = snd_cmipci_read(cm, CM_REG_CHFORMAT) & CM_CHIP_MASK1; switch (detect) { @@ -3115,7 +3115,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc switch (pci->device) { case PCI_DEVICE_ID_CMEDIA_CM8738: case PCI_DEVICE_ID_CMEDIA_CM8738B: - if (!pci_dev_present(intel_82437vx)) + if (!pci_dev_present(intel_82437vx)) snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_TXVX); break; default: @@ -3267,7 +3267,7 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci, if (dev >= SNDRV_CARDS) return -ENODEV; - if (! enable[dev]) { + if (!enable[dev]) { dev++; return -ENOENT; } @@ -3275,7 +3275,7 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci, card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; - + switch (pci->device) { case PCI_DEVICE_ID_CMEDIA_CM8738: case PCI_DEVICE_ID_CMEDIA_CM8738B: @@ -3343,7 +3343,7 @@ static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - + snd_pcm_suspend_all(cm->pcm); snd_pcm_suspend_all(cm->pcm2); snd_pcm_suspend_all(cm->pcm_spdif); @@ -3406,7 +3406,7 @@ static struct pci_driver driver = { .resume = snd_cmipci_resume, #endif }; - + static int __init alsa_card_cmipci_init(void) { return pci_register_driver(&driver); diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f..24cc52b 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -19,7 +19,7 @@ * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -573,13 +573,13 @@ static unsigned short snd_cs4281_ac97_read(struct snd_ac97 *ac97, struct cs4281 *chip = ac97->private_data; int count; unsigned short result; - // FIXME: volatile is necessary in the following due to a bug of - // some gcc versions + /* FIXME: volatile is necessary in the following due to a bug of */ + /* some gcc versions */ volatile int ac97_num = ((volatile struct snd_ac97 *)ac97)->num; /* * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address - * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97 + * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97 * 3. Write ACCTL = Control Register = 460h for initiating the write * 4. Read ACCTL = 460h, DCV should be reset by now and 460h = 17h * 5. if DCV not cleared, break and return error @@ -627,7 +627,7 @@ static unsigned short snd_cs4281_ac97_read(struct snd_ac97 *ac97, snd_printk(KERN_ERR "AC'97 read problem (ACCTL_DCV), reg = 0x%x\n", reg); result = 0xffff; goto __end; - + __ok1: /* * Wait for the valid status bit to go active. @@ -642,7 +642,7 @@ static unsigned short snd_cs4281_ac97_read(struct snd_ac97 *ac97, goto __ok2; udelay(10); } - + snd_printk(KERN_ERR "AC'97 read problem (ACSTS_VSTS), reg = 0x%x\n", reg); result = 0xffff; goto __end; @@ -654,7 +654,7 @@ static unsigned short snd_cs4281_ac97_read(struct snd_ac97 *ac97, */ result = snd_cs4281_peekBA0(chip, ac97_num ? BA0_ACSDA2 : BA0_ACSDA); - __end: +__end: return result; } @@ -707,7 +707,7 @@ static int snd_cs4281_trigger(struct snd_pcm_substream *substream, int cmd) static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate) { unsigned int val = ~0; - + if (real_rate) *real_rate = rate; /* special "hardcoded" rates */ @@ -721,7 +721,7 @@ static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate) default: goto __variable; } - __variable: +__variable: val = 1536000 / rate; if (real_rate) *real_rate = 1536000 / val; @@ -776,7 +776,7 @@ static void snd_cs4281_mode(struct cs4281 *chip, struct cs4281_dma *dma, snd_cs4281_pokeBA0(chip, BA0_ADCSR, val); } } - __skip_src: +__skip_src: /* Deactivate wave playback FIFO before changing slot assignments */ if (dma->regFCR == BA0_FCR0) snd_cs4281_pokeBA0(chip, dma->regFCR, snd_cs4281_peekBA0(chip, dma->regFCR) & ~BA0_FCR_FEN); @@ -834,7 +834,7 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream) struct cs4281_dma *dma = runtime->private_data; struct cs4281 *chip = snd_pcm_substream_chip(substream); - // printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies); + /* printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies); */ return runtime->buffer_size - snd_cs4281_peekBA0(chip, dma->regDCC) - 1; } @@ -965,7 +965,7 @@ static struct snd_pcm_ops snd_cs4281_capture_ops = { .pointer = snd_cs4281_pointer, }; -static int __devinit snd_cs4281_pcm(struct cs4281 * chip, int device, +static int __devinit snd_cs4281_pcm(struct cs4281 *chip, int device, struct snd_pcm ** rpcm) { struct snd_pcm *pcm; @@ -1008,7 +1008,7 @@ static int snd_cs4281_info_volume(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = CS_VOL_MASK; return 0; } - + static int snd_cs4281_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1052,24 +1052,24 @@ static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(db_scale_dsp, -4650, 150, 0); -static struct snd_kcontrol_new snd_cs4281_fm_vol = +static struct snd_kcontrol_new snd_cs4281_fm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Synth Playback Volume", - .info = snd_cs4281_info_volume, + .info = snd_cs4281_info_volume, .get = snd_cs4281_get_volume, - .put = snd_cs4281_put_volume, + .put = snd_cs4281_put_volume, .private_value = ((BA0_FMLVC << 16) | BA0_FMRVC), .tlv = { .p = db_scale_dsp }, }; -static struct snd_kcontrol_new snd_cs4281_pcm_vol = +static struct snd_kcontrol_new snd_cs4281_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Stream Playback Volume", - .info = snd_cs4281_info_volume, + .info = snd_cs4281_info_volume, .get = snd_cs4281_get_volume, - .put = snd_cs4281_put_volume, + .put = snd_cs4281_put_volume, .private_value = ((BA0_PPLVC << 16) | BA0_PPRVC), .tlv = { .p = db_scale_dsp }, }; @@ -1089,7 +1089,7 @@ static void snd_cs4281_mixer_free_ac97(struct snd_ac97 *ac97) chip->ac97 = NULL; } -static int __devinit snd_cs4281_mixer(struct cs4281 * chip) +static int __devinit snd_cs4281_mixer(struct cs4281 *chip) { struct snd_card *card = chip->card; struct snd_ac97_template ac97; @@ -1125,7 +1125,7 @@ static int __devinit snd_cs4281_mixer(struct cs4281 * chip) * proc interface */ -static void snd_cs4281_proc_read(struct snd_info_entry *entry, +static void snd_cs4281_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct cs4281 *chip = entry->private_data; @@ -1142,7 +1142,7 @@ static long snd_cs4281_BA0_read(struct snd_info_entry *entry, { long size; struct cs4281 *chip = entry->private_data; - + size = count; if (pos + size > CS4281_BA0_SIZE) size = (long)CS4281_BA0_SIZE - pos; @@ -1160,7 +1160,7 @@ static long snd_cs4281_BA1_read(struct snd_info_entry *entry, { long size; struct cs4281 *chip = entry->private_data; - + size = count; if (pos + size > CS4281_BA1_SIZE) size = (long)CS4281_BA1_SIZE - pos; @@ -1179,19 +1179,19 @@ static struct snd_info_entry_ops snd_cs4281_proc_ops_BA1 = { .read = snd_cs4281_BA1_read, }; -static void __devinit snd_cs4281_proc_init(struct cs4281 * chip) +static void __devinit snd_cs4281_proc_init(struct cs4281 *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "cs4281", &entry)) + if (!snd_card_proc_new(chip->card, "cs4281", &entry)) snd_info_set_text_ops(entry, chip, snd_cs4281_proc_read); - if (! snd_card_proc_new(chip->card, "cs4281_BA0", &entry)) { + if (!snd_card_proc_new(chip->card, "cs4281_BA0", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = chip; entry->c.ops = &snd_cs4281_proc_ops_BA0; entry->size = CS4281_BA0_SIZE; } - if (! snd_card_proc_new(chip->card, "cs4281_BA1", &entry)) { + if (!snd_card_proc_new(chip->card, "cs4281_BA1", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = chip; entry->c.ops = &snd_cs4281_proc_ops_BA1; @@ -1229,16 +1229,16 @@ static int snd_cs4281_gameport_cooked_read(struct gameport *gameport, { struct cs4281 *chip = gameport_get_port_data(gameport); unsigned js1, js2, jst; - + if (snd_BUG_ON(!chip)) return 0; js1 = snd_cs4281_peekBA0(chip, BA0_JSC1); js2 = snd_cs4281_peekBA0(chip, BA0_JSC2); jst = snd_cs4281_peekBA0(chip, BA0_JSPT); - - *buttons = (~jst >> 4) & 0x0F; - + + *buttons = (~jst >> 4) & 0x0F; + axes[0] = ((js1 & JSC1_Y1V_MASK) >> JSC1_Y1V_SHIFT) & 0xFFFF; axes[1] = ((js1 & JSC1_X1V_MASK) >> JSC1_X1V_SHIFT) & 0xFFFF; axes[2] = ((js2 & JSC2_Y2V_MASK) >> JSC2_Y2V_SHIFT) & 0xFFFF; @@ -1286,7 +1286,7 @@ static int __devinit snd_cs4281_create_gameport(struct cs4281 *chip) gp->cooked_read = snd_cs4281_gameport_cooked_read; gameport_set_port_data(gp, chip); - snd_cs4281_pokeBA0(chip, BA0_JSIO, 0xFF); // ? + snd_cs4281_pokeBA0(chip, BA0_JSIO, 0xFF); /* ? */ snd_cs4281_pokeBA0(chip, BA0_JSCTL, JSCTL_SP_MEDIUM_SLOW); gameport_register_port(gp); @@ -1388,7 +1388,7 @@ static int __devinit snd_cs4281_create(struct snd_card *card, snd_cs4281_free(chip); return -ENOMEM; } - + if (request_irq(pci->irq, snd_cs4281_interrupt, IRQF_SHARED, "CS4281", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); @@ -1427,7 +1427,7 @@ static int snd_cs4281_chip_init(struct cs4281 *chip) if (tmp & BA0_EPPMC_FPDN) snd_cs4281_pokeBA0(chip, BA0_EPPMC, tmp & ~BA0_EPPMC_FPDN); - __retry: +__retry: tmp = snd_cs4281_peekBA0(chip, BA0_CFLR); if (tmp != BA0_CFLR_DEFAULT) { snd_cs4281_pokeBA0(chip, BA0_CFLR, BA0_CFLR_DEFAULT); @@ -1440,9 +1440,9 @@ static int snd_cs4281_chip_init(struct cs4281 *chip) /* Set the 'Configuration Write Protect' register * to 4281h. Allows vendor-defined configuration - * space between 0e4h and 0ffh to be written. */ + * space between 0e4h and 0ffh to be written. */ snd_cs4281_pokeBA0(chip, BA0_CWPR, 0x4281); - + if ((tmp = snd_cs4281_peekBA0(chip, BA0_SERC1)) != (BA0_SERC1_SO1EN | BA0_SERC1_AC97)) { snd_printk(KERN_ERR "SERC1 AC'97 check failed (0x%x)\n", tmp); return -EIO; @@ -1469,7 +1469,7 @@ static int snd_cs4281_chip_init(struct cs4281 *chip) * the Sync pulse on the AC97 link. */ snd_cs4281_pokeBA0(chip, BA0_ACCTL, 0); udelay(50); - + /* Drive the ARST# pin low for a minimum of 1uS (as defined in the AC97 * spec) and then drive it high. This is done for non AC97 modes since * there might be logic external to the CS4281 that uses the ARST# line @@ -1549,7 +1549,7 @@ static int snd_cs4281_chip_init(struct cs4281 *chip) } while (time_after_eq(end_time, jiffies)); snd_printk(KERN_INFO "secondary codec doesn't respond. disable it...\n"); chip->dual_codec = 0; - __codec2_ok: ; + __codec2_ok:; } /* @@ -1787,7 +1787,7 @@ static struct snd_rawmidi_ops snd_cs4281_midi_input = .trigger = snd_cs4281_midi_input_trigger, }; -static int __devinit snd_cs4281_midi(struct cs4281 * chip, int device, +static int __devinit snd_cs4281_midi(struct cs4281 *chip, int device, struct snd_rawmidi **rrawmidi) { struct snd_rawmidi *rmidi; @@ -1855,7 +1855,7 @@ static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id) if ((status & BA0_HISR_MIDI) && chip->rmidi) { unsigned char c; - + spin_lock(&chip->reg_lock); while ((snd_cs4281_peekBA0(chip, BA0_MIDSR) & BA0_MIDSR_RBE) == 0) { c = snd_cs4281_peekBA0(chip, BA0_MIDRP); @@ -2102,7 +2102,7 @@ static struct pci_driver driver = { .resume = cs4281_resume, #endif }; - + static int __init alsa_card_cs4281_init(void) { return pci_register_driver(&driver); diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68..2e46ac7 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -10,8 +10,8 @@ * simultaneously play back audio at 16bit 44100kHz, the device actually plays * back in the same format in which it is capturing. By forcing the chip to * always play/capture in 16/44100, we can let alsa-lib convert the samples and - * that way we can hack up some full duplex audio. - * + * that way we can hack up some full duplex audio. + * * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. * The older version (VSA1) provides fairly good soundblaster emulation * although there are a couple of bugs: large DMA buffers break record, @@ -126,7 +126,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, err = pci_request_regions(pci, "CS5530"); if (err < 0) { - kfree(chip); + kfree(chip); pci_disable_device(pci); return err; } @@ -207,7 +207,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, return -ENODEV; } - printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, dma16); err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 1d8b160..07c908c 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -27,7 +27,7 @@ #include <linux/pci.h> #include <linux/slab.h> #include <linux/moduleparam.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/control.h> #include <sound/pcm.h> @@ -46,9 +46,9 @@ MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds."); static struct ac97_quirk ac97_quirks[] __devinitdata = { #if 0 /* Not yet confirmed if all 5536 boards are HP only */ { - .subvendor = PCI_VENDOR_ID_AMD, - .subdevice = PCI_DEVICE_ID_AMD_CS5536_AUDIO, - .name = "AMD RDK", + .subvendor = PCI_VENDOR_ID_AMD, + .subdevice = PCI_DEVICE_ID_AMD_CS5536_AUDIO, + .name = "AMD RDK", .type = AC97_TUNE_HP_ONLY }, #endif diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 564c33b..ea67963 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -32,18 +32,18 @@ static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au) { - /* + /* we depend on snd_ac97_suspend to tell the AC97 codec to shutdown. the amd spec suggests that the LNK_SHUTDOWN be done at the same time that the codec power-down is issued. instead, - we do it just after rather than at the same + we do it just after rather than at the same time. excluding codec specific build_ops->suspend ac97 powerdown hits: - 0x8000 EAPD - 0x4000 Headphone amplifier - 0x0300 ADC & DAC - 0x0400 Analog Mixer powerdown (Vref on) + 0x8000 EAPD + 0x4000 Headphone amplifier + 0x0300 ADC & DAC + 0x0400 Analog Mixer powerdown (Vref on) I am not sure if this is the best that we can do. The remainder to be investigated are: - analog mixer (vref off) 0x0800

1 0

[alsa-devel] [PATCH 05/24] sound/pci: coding style fixes: cs46xx
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 316 ++++++++++---------- sound/pci/cs46xx/dsp_spos.c | 428 ++++++++++++++-------------- sound/pci/cs46xx/dsp_spos.h | 8 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 546 +++++++++++++++++----------------- sound/pci/cs46xx/imgs/cwcbinhack.h | 2 +- 6 files changed, 651 insertions(+), 651 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index e876b32..d766d85 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -21,7 +21,7 @@ /* NOTES: - - sometimes the sound is metallic and sibilant, unloading and + - sometimes the sound is metallic and sibilant, unloading and reloading the module may solve this. */ diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index a10ab82..d8ab1ad 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -7,7 +7,7 @@ * KNOWN BUGS: * - Sometimes the SPDIF input DSP tasks get's unsynchronized * and the SPDIF get somewhat "distorcionated", or/and left right channel - * are swapped. To get around this problem when it happens, mute and unmute + * are swapped. To get around this problem when it happens, mute and unmute * the SPDIF input mixer control. * - On the Hercules Game Theater XP the amplifier are sometimes turned * off on inadecuate moments which causes distorcions on sound. @@ -22,13 +22,13 @@ * is default configuration), no SPDIF, no secondary codec, no * multi channel PCM. But known to work. * - * FINALLY: A credit to the developers Tom and Jordan + * FINALLY: A credit to the developers Tom and Jordan * at Cirrus for have helping me out with the DSP, however we * still don't have sufficient documentation and technical * references to be able to implement all fancy feutures - * supported by the cs46xx DSP's. + * supported by the cs46xx DSP's. * Benny <benny(a)hostmobility.com> - * + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -62,7 +62,7 @@ #include <sound/pcm_params.h> #include <sound/cs46xx.h> -#include <asm/io.h> +#include <linux/io.h> #include "cs46xx_lib.h" #include "dsp_spos.h" @@ -102,7 +102,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, /* * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address - * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97 + * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97 * 3. Write ACCTL = Control Register = 460h for initiating the write7---55 * 4. Read ACCTL = 460h, DCV should be reset by now and 460h = 17h * 5. if DCV not cleared, break and return error @@ -114,10 +114,10 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, tmp = snd_cs46xx_peekBA0(chip, BA0_ACCTL); if ((tmp & ACCTL_VFRM) == 0) { snd_printk(KERN_WARNING "cs46xx: ACCTL_VFRM not set 0x%x\n",tmp); - snd_cs46xx_pokeBA0(chip, BA0_ACCTL, (tmp & (~ACCTL_ESYN)) | ACCTL_VFRM ); + snd_cs46xx_pokeBA0(chip, BA0_ACCTL, (tmp & (~ACCTL_ESYN)) | ACCTL_VFRM); msleep(50); tmp = snd_cs46xx_peekBA0(chip, BA0_ACCTL + offset); - snd_cs46xx_pokeBA0(chip, BA0_ACCTL, tmp | ACCTL_ESYN | ACCTL_VFRM ); + snd_cs46xx_pokeBA0(chip, BA0_ACCTL, tmp | ACCTL_ESYN | ACCTL_VFRM); } @@ -137,7 +137,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, snd_cs46xx_pokeBA0(chip, BA0_ACCAD, reg); snd_cs46xx_pokeBA0(chip, BA0_ACCDA, 0); if (codec_index == CS46XX_PRIMARY_CODEC_INDEX) { - snd_cs46xx_pokeBA0(chip, BA0_ACCTL,/* clear ACCTL_DCV */ ACCTL_CRW | + snd_cs46xx_pokeBA0(chip, BA0_ACCTL,/* clear ACCTL_DCV */ ACCTL_CRW | ACCTL_VFRM | ACCTL_ESYN | ACCTL_RSTN); snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_DCV | ACCTL_CRW | @@ -168,7 +168,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, snd_printk(KERN_ERR "AC'97 read problem (ACCTL_DCV), reg = 0x%x\n", reg); result = 0xffff; goto end; - + ok1: /* * Wait for the valid status bit to go active. @@ -183,7 +183,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, goto ok2; udelay(10); } - + snd_printk(KERN_ERR "AC'97 read problem (ACSTS_VSTS), codec_index %d, reg = 0x%x\n", codec_index, reg); result = 0xffff; goto end; @@ -199,14 +199,14 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, snd_cs46xx_peekBA0(chip, BA0_ACCAD)); #endif - //snd_cs46xx_peekBA0(chip, BA0_ACCAD); + /* snd_cs46xx_peekBA0(chip, BA0_ACCAD); */ result = snd_cs46xx_peekBA0(chip, BA0_ACSDA + offset); end: chip->active_ctrl(chip, -1); return result; } -static unsigned short snd_cs46xx_ac97_read(struct snd_ac97 * ac97, +static unsigned short snd_cs46xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct snd_cs46xx *chip = ac97->private_data; @@ -339,7 +339,7 @@ int snd_cs46xx_download(struct snd_cs46xx *chip, int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip, unsigned long offset, - unsigned long len) + unsigned long len) { void __iomem *dst; unsigned int bank = offset >> 16; @@ -374,7 +374,7 @@ int snd_cs46xx_download_image(struct snd_cs46xx *chip) BA1Struct.memory[idx].size)) < 0) return err; offset += BA1Struct.memory[idx].size >> 2; - } + } return 0; } #endif /* CONFIG_SND_CS46XX_NEW_DSP */ @@ -412,23 +412,23 @@ static void snd_cs46xx_reset(struct snd_cs46xx *chip) snd_cs46xx_poke(chip, BA1_FRMT, 0xadf); } -static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout) +static int cs46xx_wait_for_fifo(struct snd_cs46xx *chip, int retry_timeout) { u32 i, status = 0; /* * Make sure the previous FIFO write operation has completed. */ - for(i = 0; i < 50; i++){ + for (i = 0; i < 50; i++){ status = snd_cs46xx_peekBA0(chip, BA0_SERBST); - - if( !(status & SERBST_WBSY) ) + + if (!(status & SERBST_WBSY)) break; mdelay(retry_timeout); } - - if(status & SERBST_WBSY) { - snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n"); + + if (status & SERBST_WBSY) { + snd_printk(KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n"); return -EINVAL; } @@ -470,7 +470,7 @@ static void snd_cs46xx_clear_serial_FIFOs(struct snd_cs46xx *chip) if (powerdown) snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp); - + break; } /* @@ -668,7 +668,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned frameGroupLength *= 3; } for (cnt = 5; cnt <= 125; cnt *= 5) { - if (((rate / cnt) * cnt) != rate) + if (((rate / cnt) * cnt) != rate) frameGroupLength *= 5; } @@ -778,7 +778,7 @@ static int snd_cs46xx_playback_trigger(struct snd_pcm_substream *substream, #ifdef CONFIG_SND_CS46XX_NEW_DSP struct snd_cs46xx_pcm *cpcm = substream->runtime->private_data; - if (! cpcm->pcm_channel) { + if (!cpcm->pcm_channel) { return -ENXIO; } #endif @@ -787,7 +787,7 @@ static int snd_cs46xx_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: #ifdef CONFIG_SND_CS46XX_NEW_DSP /* magic value to unmute PCM stream playback volume */ - snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + + snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + SCBVolumeCtrl) << 2, 0x80008000); if (cpcm->pcm_channel->unlinked) @@ -811,7 +811,7 @@ static int snd_cs46xx_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: #ifdef CONFIG_SND_CS46XX_NEW_DSP /* magic mute channel */ - snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + + snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + SCBVolumeCtrl) << 2, 0xffffffff); if (!cpcm->pcm_channel->unlinked) @@ -865,13 +865,13 @@ static int snd_cs46xx_capture_trigger(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_CS46XX_NEW_DSP -static int _cs46xx_adjust_sample_rate (struct snd_cs46xx *chip, struct snd_cs46xx_pcm *cpcm, - int sample_rate) +static int _cs46xx_adjust_sample_rate(struct snd_cs46xx *chip, struct snd_cs46xx_pcm *cpcm, + int sample_rate) { /* If PCMReaderSCB and SrcTaskSCB not created yet ... */ - if ( cpcm->pcm_channel == NULL) { - cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, + if (cpcm->pcm_channel == NULL) { + cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel(chip, sample_rate, cpcm, cpcm->hw_buf.addr,cpcm->pcm_channel_id); if (cpcm->pcm_channel == NULL) { snd_printk(KERN_ERR "cs46xx: failed to create virtual PCM channel\n"); @@ -882,9 +882,9 @@ static int _cs46xx_adjust_sample_rate (struct snd_cs46xx *chip, struct snd_cs46x /* if sample rate is changed */ if ((int)cpcm->pcm_channel->sample_rate != sample_rate) { int unlinked = cpcm->pcm_channel->unlinked; - cs46xx_dsp_destroy_pcm_channel (chip,cpcm->pcm_channel); + cs46xx_dsp_destroy_pcm_channel(chip,cpcm->pcm_channel); - if ( (cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, cpcm, + if ((cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel(chip, sample_rate, cpcm, cpcm->hw_buf.addr, cpcm->pcm_channel_id)) == NULL) { snd_printk(KERN_ERR "cs46xx: failed to re-create virtual PCM channel\n"); @@ -1017,7 +1017,7 @@ static int snd_cs46xx_playback_hw_free(struct snd_pcm_substream *substream) if (runtime->dma_area != cpcm->hw_buf.area) snd_pcm_lib_free_pages(substream); - + runtime->dma_area = NULL; runtime->dma_addr = 0; runtime->dma_bytes = 0; @@ -1039,7 +1039,7 @@ static int snd_cs46xx_playback_prepare(struct snd_pcm_substream *substream) if (snd_BUG_ON(!cpcm->pcm_channel)) return -ENXIO; - pfie = snd_cs46xx_peek(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 1) << 2 ); + pfie = snd_cs46xx_peek(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 1) << 2); pfie &= ~0x0000f03f; #else /* old dsp */ @@ -1068,7 +1068,7 @@ static int snd_cs46xx_playback_prepare(struct snd_pcm_substream *substream) if (snd_pcm_format_big_endian(runtime->format)) pfie |= 0x00004000; } - + memset(&cpcm->pcm_rec, 0, sizeof(cpcm->pcm_rec)); cpcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); cpcm->pcm_rec.hw_buffer_size = runtime->period_size * CS46XX_FRAGS << cpcm->shift; @@ -1161,7 +1161,7 @@ static irqreturn_t snd_cs46xx_interrupt(int irq, void *dev_id) struct snd_cs46xx *chip = dev_id; u32 status1; #ifdef CONFIG_SND_CS46XX_NEW_DSP - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; u32 status2; int i; struct snd_cs46xx_pcm *cpcm = NULL; @@ -1181,7 +1181,7 @@ static irqreturn_t snd_cs46xx_interrupt(int irq, void *dev_id) for (i = 0; i < DSP_MAX_PCM_CHANNELS; ++i) { if (i <= 15) { - if ( status1 & (1 << i) ) { + if (status1 & (1 << i)) { if (i == CS46XX_DSP_CAPTURE_CHANNEL) { if (chip->capt.substream) snd_pcm_period_elapsed(chip->capt.substream); @@ -1195,8 +1195,8 @@ static irqreturn_t snd_cs46xx_interrupt(int irq, void *dev_id) } } } else { - if ( status2 & (1 << (i - 16))) { - if (ins->pcm_channels[i].active && + if (status2 & (1 << (i - 16))) { + if (ins->pcm_channels[i].active && ins->pcm_channels[i].private_data && !ins->pcm_channels[i].unlinked) { cpcm = ins->pcm_channels[i].private_data; @@ -1220,7 +1220,7 @@ static irqreturn_t snd_cs46xx_interrupt(int irq, void *dev_id) if ((status1 & HISR_MIDI) && chip->rmidi) { unsigned char c; - + spin_lock(&chip->reg_lock); while ((snd_cs46xx_peekBA0(chip, BA0_MIDSR) & MIDSR_RBE) == 0) { c = snd_cs46xx_peekBA0(chip, BA0_MIDRP); @@ -1251,7 +1251,7 @@ static irqreturn_t snd_cs46xx_interrupt(int irq, void *dev_id) static struct snd_pcm_hardware snd_cs46xx_playback = { .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/ /*SNDRV_PCM_INFO_RESUME*/), .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | @@ -1307,7 +1307,7 @@ static void snd_cs46xx_pcm_free_substream(struct snd_pcm_runtime *runtime) kfree(runtime->private_data); } -static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,int pcm_channel_id) +static int _cs46xx_playback_open_channel(struct snd_pcm_substream *substream, int pcm_channel_id) { struct snd_cs46xx *chip = snd_pcm_substream_chip(substream); struct snd_cs46xx_pcm * cpcm; @@ -1329,12 +1329,12 @@ static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,in cpcm->substream = substream; #ifdef CONFIG_SND_CS46XX_NEW_DSP mutex_lock(&chip->spos_mutex); - cpcm->pcm_channel = NULL; + cpcm->pcm_channel = NULL; cpcm->pcm_channel_id = pcm_channel_id; snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_sizes); mutex_unlock(&chip->spos_mutex); @@ -1389,7 +1389,7 @@ static int snd_cs46xx_playback_close_iec958(struct snd_pcm_substream *substream) { int err; struct snd_cs46xx *chip = snd_pcm_substream_chip(substream); - + snd_printdd("close raw iec958 channel\n"); err = snd_cs46xx_playback_close(substream); @@ -1419,7 +1419,7 @@ static int snd_cs46xx_capture_open(struct snd_pcm_substream *substream) #ifdef CONFIG_SND_CS46XX_NEW_DSP snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_sizes); #endif return 0; @@ -1734,7 +1734,7 @@ static void snd_cs46xx_mixer_free_ac97(struct snd_ac97 *ac97) chip->ac97[CS46XX_SECONDARY_CODEC_INDEX] = NULL; } -static int snd_cs46xx_vol_info(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1758,7 +1758,7 @@ static int snd_cs46xx_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value; - unsigned int val = ((0xffff - ucontrol->value.integer.value[0]) << 16 | + unsigned int val = ((0xffff - ucontrol->value.integer.value[0]) << 16 | (0xffff - ucontrol->value.integer.value[1])); unsigned int old = snd_cs46xx_peek(chip, reg); int change = (old != val); @@ -1827,7 +1827,7 @@ static int snd_cs46xx_vol_iec958_put(struct snd_kcontrol *kcontrol, struct snd_c #define snd_mixer_boolean_info snd_ctl_boolean_mono_info -static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); @@ -1841,7 +1841,7 @@ static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); @@ -1851,7 +1851,7 @@ static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, case CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT: mutex_lock(&chip->spos_mutex); change = (chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED); - if (ucontrol->value.integer.value[0] && !change) + if (ucontrol->value.integer.value[0] && !change) cs46xx_dsp_enable_spdif_out(chip); else if (change && !ucontrol->value.integer.value[0]) cs46xx_dsp_disable_spdif_out(chip); @@ -1867,7 +1867,7 @@ static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, } else if (change && !ucontrol->value.integer.value[0]) cs46xx_dsp_disable_spdif_in(chip); - + res = (change != chip->dsp_spos_instance->spdif_status_in); break; default: @@ -1878,25 +1878,25 @@ static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, return res; } -static int snd_cs46xx_adc_capture_get(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_adc_capture_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; - if (ins->adc_input != NULL) + if (ins->adc_input != NULL) ucontrol->value.integer.value[0] = 1; - else + else ucontrol->value.integer.value[0] = 0; - + return 0; } -static int snd_cs46xx_adc_capture_put(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_adc_capture_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int change = 0; if (ucontrol->value.integer.value[0] && !ins->adc_input) { @@ -1909,26 +1909,26 @@ static int snd_cs46xx_adc_capture_put(struct snd_kcontrol *kcontrol, return change; } -static int snd_cs46xx_pcm_capture_get(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_pcm_capture_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; - if (ins->pcm_input != NULL) + if (ins->pcm_input != NULL) ucontrol->value.integer.value[0] = 1; - else + else ucontrol->value.integer.value[0] = 0; return 0; } -static int snd_cs46xx_pcm_capture_put(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_pcm_capture_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int change = 0; if (ucontrol->value.integer.value[0] && !ins->pcm_input) { @@ -1942,7 +1942,7 @@ static int snd_cs46xx_pcm_capture_put(struct snd_kcontrol *kcontrol, return change; } -static int snd_herc_spdif_select_get(struct snd_kcontrol *kcontrol, +static int snd_herc_spdif_select_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); @@ -1959,8 +1959,8 @@ static int snd_herc_spdif_select_get(struct snd_kcontrol *kcontrol, /* * Game Theatre XP card - EGPIO[0] is used to select SPDIF input optical or coaxial. - */ -static int snd_herc_spdif_select_put(struct snd_kcontrol *kcontrol, + */ +static int snd_herc_spdif_select_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); @@ -1969,9 +1969,9 @@ static int snd_herc_spdif_select_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) { /* optical is default */ - snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, + snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, EGPIODR_GPOE0 | val1); /* enable EGPIO0 output */ - snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, + snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, EGPIOPTR_GPPT0 | val2); /* open-drain on output */ } else { /* coaxial */ @@ -1979,7 +1979,7 @@ static int snd_herc_spdif_select_put(struct snd_kcontrol *kcontrol, snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, val2 & ~EGPIOPTR_GPPT0); /* disable */ } - /* checking diff from the EGPIO direction register + /* checking diff from the EGPIO direction register should be enough */ return (val1 != (int)snd_cs46xx_peekBA0(chip, BA0_EGPIODR)); } @@ -1996,7 +1996,7 @@ static int snd_cs46xx_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; mutex_lock(&chip->spos_mutex); ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_default >> 24) & 0xff); @@ -2012,7 +2012,7 @@ static int snd_cs46xx_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx * chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; unsigned int val; int change; @@ -2027,7 +2027,7 @@ static int snd_cs46xx_spdif_default_put(struct snd_kcontrol *kcontrol, change = (unsigned int)ins->spdif_csuv_default != val; ins->spdif_csuv_default = val; - if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) ) + if (!(ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN)) cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val); mutex_unlock(&chip->spos_mutex); @@ -2049,7 +2049,7 @@ static int snd_cs46xx_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; mutex_lock(&chip->spos_mutex); ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_stream >> 24) & 0xff); @@ -2065,7 +2065,7 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx * chip = snd_kcontrol_chip(kcontrol); - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; unsigned int val; int change; @@ -2080,7 +2080,7 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, change = ins->spdif_csuv_stream != val; ins->spdif_csuv_stream = val; - if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN ) + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val); mutex_unlock(&chip->spos_mutex); @@ -2187,7 +2187,7 @@ static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { #ifdef CONFIG_SND_CS46XX_NEW_DSP /* set primary cs4294 codec into Extended Audio Mode */ -static int snd_cs46xx_front_dup_get(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_front_dup_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); @@ -2197,7 +2197,7 @@ static int snd_cs46xx_front_dup_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_cs46xx_front_dup_put(struct snd_kcontrol *kcontrol, +static int snd_cs46xx_front_dup_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); @@ -2228,13 +2228,13 @@ static struct snd_kcontrol_new snd_hercules_controls[] = { }; -static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) +static void snd_cs46xx_codec_reset(struct snd_ac97 *ac97) { unsigned long end_time; int err; /* reset to defaults */ - snd_ac97_write(ac97, AC97_RESET, 0); + snd_ac97_write(ac97, AC97_RESET, 0); /* set the desired CODEC mode */ if (ac97->num == CS46XX_PRIMARY_CODEC_INDEX) { @@ -2255,7 +2255,7 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) end_time = jiffies + HZ; do { unsigned short ext_mid; - + /* use preliminary reads to settle the communication */ snd_ac97_read(ac97, AC97_RESET); snd_ac97_read(ac97, AC97_VENDOR_ID1); @@ -2273,7 +2273,7 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) msleep(10); } while (time_after_eq(end_time, jiffies)); - snd_printk(KERN_ERR "CS46xx secondary codec doesn't respond!\n"); + snd_printk(KERN_ERR "CS46xx secondary codec doesn't respond!\n"); } #endif @@ -2338,7 +2338,7 @@ int __devinit snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device) #ifdef CONFIG_SND_CS46XX_NEW_DSP snd_printdd("snd_cs46xx: detecting seconadry codec\n"); /* try detect a secondary codec */ - if (! cs46xx_detect_codec(chip, CS46XX_SECONDARY_CODEC_INDEX)) + if (!cs46xx_detect_codec(chip, CS46XX_SECONDARY_CODEC_INDEX)) chip->nr_ac97_codecs = 2; #endif /* CONFIG_SND_CS46XX_NEW_DSP */ @@ -2357,7 +2357,7 @@ int __devinit snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device) id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; strcpy(id.name, "External Amplifier"); chip->eapd_switch = snd_ctl_find_id(chip->card, &id); - + #ifdef CONFIG_SND_CS46XX_NEW_DSP if (chip->nr_ac97_codecs == 1) { unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff; @@ -2378,7 +2378,7 @@ int __devinit snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device) /* turn on amplifier */ chip->amplifier_ctrl(chip, 1); - + return 0; } @@ -2563,7 +2563,7 @@ static void snd_cs46xx_gameport_trigger(struct gameport *gameport) if (snd_BUG_ON(!chip)) return; - snd_cs46xx_pokeBA0(chip, BA0_JSPT, 0xFF); //outb(gameport->io, 0xFF); + snd_cs46xx_pokeBA0(chip, BA0_JSPT, 0xFF); /* outb(gameport->io, 0xFF); */ } static unsigned char snd_cs46xx_gameport_read(struct gameport *gameport) @@ -2572,7 +2572,7 @@ static unsigned char snd_cs46xx_gameport_read(struct gameport *gameport) if (snd_BUG_ON(!chip)) return 0; - return snd_cs46xx_peekBA0(chip, BA0_JSPT); //inb(gameport->io); + return snd_cs46xx_peekBA0(chip, BA0_JSPT); /* inb(gameport->io); */ } static int snd_cs46xx_gameport_cooked_read(struct gameport *gameport, int *axes, int *buttons) @@ -2586,16 +2586,16 @@ static int snd_cs46xx_gameport_cooked_read(struct gameport *gameport, int *axes, js1 = snd_cs46xx_peekBA0(chip, BA0_JSC1); js2 = snd_cs46xx_peekBA0(chip, BA0_JSC2); jst = snd_cs46xx_peekBA0(chip, BA0_JSPT); - - *buttons = (~jst >> 4) & 0x0F; - + + *buttons = (~jst >> 4) & 0x0F; + axes[0] = ((js1 & JSC1_Y1V_MASK) >> JSC1_Y1V_SHIFT) & 0xFFFF; axes[1] = ((js1 & JSC1_X1V_MASK) >> JSC1_X1V_SHIFT) & 0xFFFF; axes[2] = ((js2 & JSC2_Y2V_MASK) >> JSC2_Y2V_SHIFT) & 0xFFFF; axes[3] = ((js2 & JSC2_X2V_MASK) >> JSC2_X2V_SHIFT) & 0xFFFF; - for(jst=0;jst<4;++jst) - if(axes[jst]==0xFFFF) axes[jst] = -1; + for (jst=0; jst<4; ++jst) + if (axes[jst]==0xFFFF) axes[jst] = -1; return 0; } @@ -2632,7 +2632,7 @@ int __devinit snd_cs46xx_gameport(struct snd_cs46xx *chip) gp->trigger = snd_cs46xx_gameport_trigger; gp->cooked_read = snd_cs46xx_gameport_cooked_read; - snd_cs46xx_pokeBA0(chip, BA0_JSIO, 0xFF); // ? + snd_cs46xx_pokeBA0(chip, BA0_JSIO, 0xFF); /* ? */ snd_cs46xx_pokeBA0(chip, BA0_JSCTL, JSCTL_SP_MEDIUM_SLOW); gameport_register_port(gp); @@ -2663,7 +2663,7 @@ static long snd_cs46xx_io_read(struct snd_info_entry *entry, void *file_private_ { long size; struct snd_cs46xx_region *region = entry->private_data; - + size = count; if (pos + (size_t)size > region->size) size = region->size - pos; @@ -2682,10 +2682,10 @@ static int __devinit snd_cs46xx_proc_init(struct snd_card *card, struct snd_cs46 { struct snd_info_entry *entry; int idx; - + for (idx = 0; idx < 5; idx++) { struct snd_cs46xx_region *region = &chip->region.idx[idx]; - if (! snd_card_proc_new(card, region->name, &entry)) { + if (!snd_card_proc_new(card, region->name, &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = chip; entry->c.ops = &snd_cs46xx_proc_io_ops; @@ -2753,7 +2753,7 @@ static void snd_cs46xx_hw_stop(struct snd_cs46xx *chip) snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, 0); /* - * Turn off the Processor by turning off the software clock enable flag in + * Turn off the Processor by turning off the software clock enable flag in * the clock control register. */ tmp = snd_cs46xx_peekBA0(chip, BA0_CLKCR1) & ~CLKCR1_SWCE; @@ -2775,7 +2775,7 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) if (chip->amplifier_ctrl) chip->amplifier_ctrl(chip, -chip->amplifier); /* force to off */ - + snd_cs46xx_proc_done(chip); if (chip->region.idx[0].resource) @@ -2800,7 +2800,7 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) chip->dsp_spos_instance = NULL; } #endif - + #ifdef CONFIG_PM kfree(chip->saved_regs); #endif @@ -2823,7 +2823,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) { int timeout; - /* + /* * First, blast the clock control register to zero so that the PLL starts * out in a known state, and blast the master serial port control register * to zero so that the serial ports also start out in a known state. @@ -2834,9 +2834,9 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) /* * If we are in AC97 mode, then we must set the part to a host controlled * AC-link. Otherwise, we won't be able to bring up the link. - */ + */ #ifdef CONFIG_SND_CS46XX_NEW_DSP - snd_cs46xx_pokeBA0(chip, BA0_SERACC, SERACC_HSP | SERACC_CHIP_TYPE_2_0 | + snd_cs46xx_pokeBA0(chip, BA0_SERACC, SERACC_HSP | SERACC_CHIP_TYPE_2_0 | SERACC_TWO_CODECS); /* 2.00 dual codecs */ /* snd_cs46xx_pokeBA0(chip, BA0_SERACC, SERACC_HSP | SERACC_CHIP_TYPE_2_0); */ /* 2.00 codec */ #else @@ -2858,7 +2858,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) #ifdef CONFIG_SND_CS46XX_NEW_DSP snd_cs46xx_pokeBA0(chip, BA0_ACCTL2, ACCTL_RSTN); #endif - + /* * The first thing we do here is to enable sync generation. As soon * as we start receiving bit clock, we'll start producing the SYNC @@ -2966,7 +2966,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) for (count = 0; count < 150; count++) { /* First, we want to wait for a short time. */ udelay(25); - + if (snd_cs46xx_peekBA0(chip, BA0_ACSTS2) & ACSTS_CRDY) break; } @@ -3009,7 +3009,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) return -EIO; #else /* This may happen on a cold boot with a Terratec SiXPack 5.1. - Reloading the driver may help, if there's other soundcards + Reloading the driver may help, if there's other soundcards with the same problem I would like to know. (Benny) */ snd_printk(KERN_ERR "ERROR: snd-cs46xx: never read ISV3 & ISV4 from AC'97\n"); @@ -3036,7 +3036,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) /* snd_cs46xx_pokeBA0(chip, BA0_AC97_POWERDOWN, 0x300); */ /* - * Turn off the Processor by turning off the software clock enable flag in + * Turn off the Processor by turning off the software clock enable flag in * the clock control register. */ /* tmp = snd_cs46xx_peekBA0(chip, BA0_CLKCR1) & ~CLKCR1_SWCE; */ @@ -3046,7 +3046,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) } /* - * start and load DSP + * start and load DSP */ static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) @@ -3054,7 +3054,7 @@ static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) unsigned int tmp; snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); - + tmp = snd_cs46xx_peek(chip, BA1_PFIE); tmp &= ~0x0000f03f; snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ @@ -3066,7 +3066,7 @@ static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) } int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) -{ +{ unsigned int tmp; /* * Reset the processor. @@ -3140,9 +3140,9 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) snd_cs46xx_proc_start(chip); cs46xx_enable_stream_irqs(chip); - + #ifndef CONFIG_SND_CS46XX_NEW_DSP - /* set the attenuation to 0dB */ + /* set the attenuation to 0dB */ snd_cs46xx_poke(chip, BA1_PVOL, 0x80008000); snd_cs46xx_poke(chip, BA1_CVOL, 0x80008000); #endif @@ -3154,15 +3154,15 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) /* * AMP control - null AMP */ - + static void amp_none(struct snd_cs46xx *chip, int change) -{ +{ } #ifdef CONFIG_SND_CS46XX_NEW_DSP static int voyetra_setup_eapd_slot(struct snd_cs46xx *chip) { - + u32 idx, valid_slots,tmp,powerdown = 0; u16 modem_power,pin_config,logic_type; @@ -3183,39 +3183,39 @@ static int voyetra_setup_eapd_slot(struct snd_cs46xx *chip) * Clear PRA. The Bonzo chip will be used for GPIO not for modem * stuff. */ - if(chip->nr_ac97_codecs != 2) { + if (chip->nr_ac97_codecs != 2) { snd_printk (KERN_ERR "cs46xx: cs46xx_setup_eapd_slot() - no secondary codec configured\n"); return -EINVAL; } - modem_power = snd_cs46xx_codec_read (chip, + modem_power = snd_cs46xx_codec_read (chip, AC97_EXTENDED_MSTATUS, CS46XX_SECONDARY_CODEC_INDEX); modem_power &=0xFEFF; - snd_cs46xx_codec_write(chip, + snd_cs46xx_codec_write(chip, AC97_EXTENDED_MSTATUS, modem_power, CS46XX_SECONDARY_CODEC_INDEX); /* * Set GPIO pin's 7 and 8 so that they are configured for output. */ - pin_config = snd_cs46xx_codec_read (chip, + pin_config = snd_cs46xx_codec_read (chip, AC97_GPIO_CFG, CS46XX_SECONDARY_CODEC_INDEX); pin_config &=0x27F; - snd_cs46xx_codec_write(chip, + snd_cs46xx_codec_write(chip, AC97_GPIO_CFG, pin_config, CS46XX_SECONDARY_CODEC_INDEX); - + /* * Set GPIO pin's 7 and 8 so that they are compatible with CMOS logic. */ logic_type = snd_cs46xx_codec_read(chip, AC97_GPIO_POLARITY, CS46XX_SECONDARY_CODEC_INDEX); - logic_type &=0x27F; + logic_type &=0x27F; snd_cs46xx_codec_write (chip, AC97_GPIO_POLARITY, logic_type, CS46XX_SECONDARY_CODEC_INDEX); @@ -3224,16 +3224,16 @@ static int voyetra_setup_eapd_slot(struct snd_cs46xx *chip) valid_slots |= 0x200; snd_cs46xx_pokeBA0(chip, BA0_ACOSV, valid_slots); - if ( cs46xx_wait_for_fifo(chip,1) ) { + if (cs46xx_wait_for_fifo(chip,1)) { snd_printdd("FIFO is busy\n"); - + return -EINVAL; } /* - * Fill slots 12 with the correct value for the GPIO pins. + * Fill slots 12 with the correct value for the GPIO pins. */ - for(idx = 0x90; idx <= 0x9F; idx++) { + for (idx = 0x90; idx <= 0x9F; idx++) { /* * Initialize the fifo so that bits 7 and 8 are on. * @@ -3245,17 +3245,17 @@ static int voyetra_setup_eapd_slot(struct snd_cs46xx *chip) /* * Wait for command to complete */ - if ( cs46xx_wait_for_fifo(chip,200) ) { + if (cs46xx_wait_for_fifo(chip,200)) { snd_printdd("failed waiting for FIFO at addr (%02X)\n",idx); return -EINVAL; } - + /* * Write the serial port FIFO index. */ snd_cs46xx_pokeBA0(chip, BA0_SERBAD, idx); - + /* * Tell the serial port to load the new value into the FIFO location. */ @@ -3279,17 +3279,17 @@ static int voyetra_setup_eapd_slot(struct snd_cs46xx *chip) /* * Crystal EAPD mode */ - + static void amp_voyetra(struct snd_cs46xx *chip, int change) { - /* Manage the EAPD bit on the Crystal 4297 + /* Manage the EAPD bit on the Crystal 4297 and the Analog AD1885 */ - + #ifdef CONFIG_SND_CS46XX_NEW_DSP int old = chip->amplifier; #endif int oval, val; - + chip->amplifier += change; oval = snd_cs46xx_codec_read(chip, AC97_POWERDOWN, CS46XX_PRIMARY_CODEC_INDEX); @@ -3316,7 +3316,7 @@ static void amp_voyetra(struct snd_cs46xx *chip, int change) #endif } -static void hercules_init(struct snd_cs46xx *chip) +static void hercules_init(struct snd_cs46xx *chip) { /* default: AMP off, and SPDIF input optical */ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, EGPIODR_GPOE0); @@ -3326,7 +3326,7 @@ static void hercules_init(struct snd_cs46xx *chip) /* * Game Theatre XP card - EGPIO[2] is used to enable the external amp. - */ + */ static void amp_hercules(struct snd_cs46xx *chip, int change) { int old = chip->amplifier; @@ -3337,9 +3337,9 @@ static void amp_hercules(struct snd_cs46xx *chip, int change) if (chip->amplifier && !old) { snd_printdd ("Hercules amplifier ON\n"); - snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, + snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, EGPIODR_GPOE2 | val1); /* enable EGPIO2 output */ - snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, + snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, EGPIOPTR_GPPT2 | val2); /* open-drain on output */ } else if (old && !chip->amplifier) { snd_printdd ("Hercules amplifier OFF\n"); @@ -3348,7 +3348,7 @@ static void amp_hercules(struct snd_cs46xx *chip, int change) } } -static void voyetra_mixer_init (struct snd_cs46xx *chip) +static void voyetra_mixer_init(struct snd_cs46xx *chip) { snd_printdd ("initializing Voyetra mixer\n"); @@ -3357,7 +3357,7 @@ static void voyetra_mixer_init (struct snd_cs46xx *chip) snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, EGPIODR_GPOE0); } -static void hercules_mixer_init (struct snd_cs46xx *chip) +static void hercules_mixer_init(struct snd_cs46xx *chip) { #ifdef CONFIG_SND_CS46XX_NEW_DSP unsigned int idx; @@ -3374,7 +3374,7 @@ static void hercules_mixer_init (struct snd_cs46xx *chip) if (chip->in_suspend) return; - for (idx = 0 ; idx < ARRAY_SIZE(snd_hercules_controls); idx++) { + for (idx = 0; idx < ARRAY_SIZE(snd_hercules_controls); idx++) { struct snd_kcontrol *kctl; kctl = snd_ctl_new1(&snd_hercules_controls[idx], chip); @@ -3391,7 +3391,7 @@ static void hercules_mixer_init (struct snd_cs46xx *chip) /* * Untested */ - + static void amp_voyetra_4294(struct snd_cs46xx *chip, int change) { chip->amplifier += change; @@ -3421,21 +3421,21 @@ static void amp_voyetra_4294(struct snd_cs46xx *chip, int change) * Linuxcare. Perhaps one day Crystal will document their chips well * enough to make them useful. */ - + static void clkrun_hack(struct snd_cs46xx *chip, int change) { u16 control, nval; - + if (!chip->acpi_port) return; chip->amplifier += change; - - /* Read ACPI port */ + + /* Read ACPI port */ nval = control = inw(chip->acpi_port + 0x10); /* Flip CLKRUN off while running */ - if (! chip->amplifier) + if (!chip->amplifier) nval |= 0x2000; else nval &= ~0x2000; @@ -3443,7 +3443,7 @@ static void clkrun_hack(struct snd_cs46xx *chip, int change) outw(nval, chip->acpi_port + 0x10); } - + /* * detect intel piix4 */ @@ -3453,13 +3453,13 @@ static void clkrun_init(struct snd_cs46xx *chip) u8 pp; chip->acpi_port = 0; - + pdev = pci_get_device(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82371AB_3, NULL); if (pdev == NULL) return; /* Not a thinkpad thats for sure */ - /* Find the control port */ + /* Find the control port */ pci_read_config_byte(pdev, 0x41, &pp); chip->acpi_port = pp << 8; pci_dev_put(pdev); @@ -3469,7 +3469,7 @@ static void clkrun_init(struct snd_cs46xx *chip) /* * Card subid table */ - + struct cs_card_type { u16 vendor; @@ -3595,8 +3595,8 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; snd_pcm_suspend_all(chip->pcm); - // chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL); - // chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE); + /* chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL); */ + /* chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE); */ snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); @@ -3652,9 +3652,9 @@ int snd_cs46xx_resume(struct pci_dev *pci) #endif #if 0 - snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, + snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, chip->ac97_general_purpose); - snd_cs46xx_codec_write(chip, AC97_POWER_CONTROL, + snd_cs46xx_codec_write(chip, AC97_POWER_CONTROL, chip->ac97_powerdown); mdelay(10); snd_cs46xx_codec_write(chip, BA0_AC97_POWERDOWN, @@ -3700,7 +3700,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_cs46xx_dev_free, }; - + *rchip = NULL; /* enable PCI device */ @@ -3782,7 +3782,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card, chip->active_ctrl = clkrun_hack; clkrun_init(chip); } - + if (chip->amplifier_ctrl == NULL) chip->amplifier_ctrl = amp_none; if (chip->active_ctrl == NULL) @@ -3835,7 +3835,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card, snd_cs46xx_free(chip); return err; } - + snd_cs46xx_proc_init(card, chip); #ifdef CONFIG_PM diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f4f0c8f..ad6e433 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -20,7 +20,7 @@ */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/init.h> @@ -37,10 +37,10 @@ #include "cs46xx_lib.h" #include "dsp_spos.h" -static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, - struct dsp_scb_descriptor * fg_entry); +static int cs46xx_dsp_async_init(struct snd_cs46xx *chip, + struct dsp_scb_descriptor *fg_entry); -static enum wide_opcode wide_opcodes[] = { +static enum wide_opcode wide_opcodes[] = { WIDE_FOR_BEGIN_LOOP, WIDE_FOR_BEGIN_LOOP2, WIDE_COND_GOTO_ADDR, @@ -55,17 +55,17 @@ static enum wide_opcode wide_opcodes[] = { WIDE_TBEQ_NCOND_CALL1_ADDR }; -static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 size, +static int shadow_and_reallocate_code(struct snd_cs46xx *chip, u32 *data, u32 size, u32 overlay_begin_address) { unsigned int i = 0, j, nreallocated = 0; u32 hival,loval,address; u32 mop_operands,mop_type,wide_op; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(size %2)) return -EINVAL; - + while (i < size) { loval = data[i++]; hival = data[i++]; @@ -73,37 +73,37 @@ static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 if (ins->code.offset > 0) { mop_operands = (hival >> 6) & 0x03fff; mop_type = mop_operands >> 10; - + /* check for wide type instruction */ if (mop_type == 0 && (mop_operands & WIDE_LADD_INSTR_MASK) == 0 && (mop_operands & WIDE_INSTR_MASK) != 0) { wide_op = loval & 0x7f; - for (j = 0;j < ARRAY_SIZE(wide_opcodes); ++j) { + for (j = 0; j < ARRAY_SIZE(wide_opcodes); ++j) { if (wide_opcodes[j] == wide_op) { /* need to reallocate instruction */ address = (hival & 0x00FFF) << 5; address |= loval >> 15; - + snd_printdd("handle_wideop[1]: %05x:%05x addr %04x\n",hival,loval,address); - - if ( !(address & 0x8000) ) { + + if (!(address & 0x8000)) { address += (ins->code.offset / 2) - overlay_begin_address; } else { snd_printdd("handle_wideop[1]: ROM symbol not reallocated\n"); } - + hival &= 0xFF000; loval &= 0x07FFF; - + hival |= ( (address >> 5) & 0x00FFF); loval |= ( (address << 15) & 0xF8000); - + address = (hival & 0x00FFF) << 5; address |= loval >> 15; - - snd_printdd("handle_wideop:[2] %05x:%05x addr %04x\n",hival,loval,address); - nreallocated ++; + + snd_printdd("handle_wideop:[2] %05x:%05x addr %04x\n",hival,loval,address); + nreallocated++; } /* wide_opcodes[j] == wide_op */ } /* for */ } /* mod_type == 0 ... */ @@ -117,10 +117,10 @@ static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 return nreallocated; } -static struct dsp_segment_desc * get_segment_desc (struct dsp_module_desc * module, int seg_type) +static struct dsp_segment_desc *get_segment_desc(struct dsp_module_desc *module, int seg_type) { int i; - for (i = 0;i < module->nsegments; ++i) { + for (i = 0; i < module->nsegments; ++i) { if (module->segments[i].segment_type == seg_type) { return (module->segments + i); } @@ -129,7 +129,7 @@ static struct dsp_segment_desc * get_segment_desc (struct dsp_module_desc * modu return NULL; }; -static int find_free_symbol_index (struct dsp_spos_instance * ins) +static int find_free_symbol_index(struct dsp_spos_instance *ins) { int index = ins->symbol_table.nsymbols,i; @@ -143,19 +143,19 @@ static int find_free_symbol_index (struct dsp_spos_instance * ins) return index; } -static int add_symbols (struct snd_cs46xx * chip, struct dsp_module_desc * module) +static int add_symbols(struct snd_cs46xx *chip, struct dsp_module_desc *module) { int i; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (module->symbol_table.nsymbols > 0) { if (!strcmp(module->symbol_table.symbols[0].symbol_name, "OVERLAYBEGINADDRESS") && - module->symbol_table.symbols[0].symbol_type == SYMBOL_CONSTANT ) { + module->symbol_table.symbols[0].symbol_type == SYMBOL_CONSTANT) { module->overlay_begin_address = module->symbol_table.symbols[0].address; } } - for (i = 0;i < module->symbol_table.nsymbols; ++i) { + for (i = 0; i < module->symbol_table.nsymbols; ++i) { if (ins->symbol_table.nsymbols == (DSP_MAX_SYMBOLS - 1)) { snd_printk(KERN_ERR "dsp_spos: symbol table is full\n"); return -ENOMEM; @@ -171,7 +171,7 @@ static int add_symbols (struct snd_cs46xx * chip, struct dsp_module_desc * modul ins->symbol_table.symbols[ins->symbol_table.nsymbols].module = module; ins->symbol_table.symbols[ins->symbol_table.nsymbols].deleted = 0; - if (ins->symbol_table.nsymbols > ins->symbol_table.highest_frag_index) + if (ins->symbol_table.nsymbols > ins->symbol_table.highest_frag_index) ins->symbol_table.highest_frag_index = ins->symbol_table.nsymbols; ins->symbol_table.nsymbols++; @@ -185,17 +185,17 @@ static int add_symbols (struct snd_cs46xx * chip, struct dsp_module_desc * modul } static struct dsp_symbol_entry * -add_symbol (struct snd_cs46xx * chip, char * symbol_name, u32 address, int type) +add_symbol(struct snd_cs46xx *chip, char *symbol_name, u32 address, int type) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_symbol_entry * symbol = NULL; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_symbol_entry *symbol = NULL; int index; if (ins->symbol_table.nsymbols == (DSP_MAX_SYMBOLS - 1)) { snd_printk(KERN_ERR "dsp_spos: symbol table is full\n"); return NULL; } - + if (cs46xx_dsp_lookup_symbol(chip, symbol_name, type) != NULL) { @@ -212,7 +212,7 @@ add_symbol (struct snd_cs46xx * chip, char * symbol_name, u32 address, int type) ins->symbol_table.symbols[index].deleted = 0; symbol = (ins->symbol_table.symbols + index); - if (index > ins->symbol_table.highest_frag_index) + if (index > ins->symbol_table.highest_frag_index) ins->symbol_table.highest_frag_index = index; if (index == ins->symbol_table.nsymbols) @@ -221,11 +221,11 @@ add_symbol (struct snd_cs46xx * chip, char * symbol_name, u32 address, int type) return symbol; } -struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) +struct dsp_spos_instance *cs46xx_dsp_spos_create(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = kzalloc(sizeof(struct dsp_spos_instance), GFP_KERNEL); + struct dsp_spos_instance *ins = kzalloc(sizeof(struct dsp_spos_instance), GFP_KERNEL); - if (ins == NULL) + if (ins == NULL) return NULL; /* better to use vmalloc for this big table */ @@ -271,8 +271,8 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) /* set left and right validity bits and default channel status */ - ins->spdif_csuv_default = - ins->spdif_csuv_stream = + ins->spdif_csuv_default = + ins->spdif_csuv_stream = /* byte 0 */ ((unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF & 0xff)) << 24) | /* byte 1 */ ((unsigned int)_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 8) & 0xff)) << 16) | /* byte 3 */ (unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) | @@ -285,10 +285,10 @@ error: return NULL; } -void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) +void cs46xx_dsp_spos_destroy(struct snd_cs46xx *chip) { int i; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(!ins)) return; @@ -297,7 +297,7 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) for (i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) continue; - cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); + cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i)); } kfree(ins->code.data); @@ -354,10 +354,10 @@ static int dsp_load_sample(struct snd_cs46xx *chip, return 0; } -int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module) +int cs46xx_dsp_load_module(struct snd_cs46xx *chip, struct dsp_module_desc *module) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM); + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_segment_desc *code = get_segment_desc (module,SEGTYPE_SP_PROGRAM); u32 doffset, dsize; int err; @@ -367,12 +367,12 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m } snd_printdd("dsp_spos: loading module %s into DSP\n", module->module_name); - + if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing parameter area\n"); snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE); } - + err = dsp_load_parameter(chip, get_segment_desc(module, SEGTYPE_SP_PARAMETER)); if (err < 0) @@ -412,11 +412,11 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m snd_printk(KERN_ERR "dsp_spos: failed to load symbol table\n"); return -ENOMEM; } - + doffset = (code->offset * 4 + ins->code.offset * 4 + DSP_CODE_BYTE_OFFSET); dsize = code->size * 4; snd_printdd("dsp_spos: downloading code to chip (%08x-%08x)\n", - doffset,doffset + dsize); + doffset,doffset + dsize); module->nfixups = shadow_and_reallocate_code(chip,code->data,code->size,module->overlay_begin_address); @@ -438,12 +438,12 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m } struct dsp_symbol_entry * -cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symbol_type) +cs46xx_dsp_lookup_symbol(struct snd_cs46xx *chip, char *symbol_name, int symbol_type) { int i; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; - for ( i = 0; i < ins->symbol_table.nsymbols; ++i ) { + for (i = 0; i < ins->symbol_table.nsymbols; ++i) { if (ins->symbol_table.symbols[i].deleted) continue; @@ -465,12 +465,12 @@ cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symb #ifdef CONFIG_PROC_FS static struct dsp_symbol_entry * -cs46xx_dsp_lookup_symbol_addr (struct snd_cs46xx * chip, u32 address, int symbol_type) +cs46xx_dsp_lookup_symbol_addr(struct snd_cs46xx *chip, u32 address, int symbol_type) { int i; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; - for ( i = 0; i < ins->symbol_table.nsymbols; ++i ) { + for (i = 0; i < ins->symbol_table.nsymbols; ++i) { if (ins->symbol_table.symbols[i].deleted) continue; @@ -486,15 +486,15 @@ cs46xx_dsp_lookup_symbol_addr (struct snd_cs46xx * chip, u32 address, int symbol } -static void cs46xx_dsp_proc_symbol_table_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_symbol_table_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_cs46xx *chip = entry->private_data; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i; snd_iprintf(buffer, "SYMBOLS:\n"); - for ( i = 0; i < ins->symbol_table.nsymbols; ++i ) { + for (i = 0; i < ins->symbol_table.nsymbols; ++i) { char *module_str = "system"; if (ins->symbol_table.symbols[i].deleted) @@ -504,32 +504,32 @@ static void cs46xx_dsp_proc_symbol_table_read (struct snd_info_entry *entry, module_str = ins->symbol_table.symbols[i].module->module_name; } - + snd_iprintf(buffer, "%04X <%02X> %s [%s]\n", ins->symbol_table.symbols[i].address, ins->symbol_table.symbols[i].symbol_type, ins->symbol_table.symbols[i].symbol_name, - module_str); + module_str); } } -static void cs46xx_dsp_proc_modules_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_modules_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_cs46xx *chip = entry->private_data; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i,j; mutex_lock(&chip->spos_mutex); snd_iprintf(buffer, "MODULES:\n"); - for ( i = 0; i < ins->nmodules; ++i ) { + for (i = 0; i < ins->nmodules; ++i) { snd_iprintf(buffer, "\n%s:\n", ins->modules[i].module_name); snd_iprintf(buffer, " %d symbols\n", ins->modules[i].symbol_table.nsymbols); snd_iprintf(buffer, " %d fixups\n", ins->modules[i].nfixups); for (j = 0; j < ins->modules[i].nsegments; ++ j) { - struct dsp_segment_desc * desc = (ins->modules[i].segments + j); + struct dsp_segment_desc *desc = (ins->modules[i].segments + j); snd_iprintf(buffer, " segment %02x offset %08x size %08x\n", desc->segment_type,desc->offset, desc->size); } @@ -537,20 +537,20 @@ static void cs46xx_dsp_proc_modules_read (struct snd_info_entry *entry, mutex_unlock(&chip->spos_mutex); } -static void cs46xx_dsp_proc_task_tree_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_task_tree_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_cs46xx *chip = entry->private_data; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i, j, col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; mutex_lock(&chip->spos_mutex); snd_iprintf(buffer, "TASK TREES:\n"); - for ( i = 0; i < ins->ntask; ++i) { + for (i = 0; i < ins->ntask; ++i) { snd_iprintf(buffer,"\n%04x %s:\n",ins->tasks[i].address,ins->tasks[i].task_name); - for (col = 0,j = 0;j < ins->tasks[i].size; j++,col++) { + for (col = 0,j = 0; j < ins->tasks[i].size; j++, col++) { u32 val; if (col == 4) { snd_iprintf(buffer,"\n"); @@ -561,26 +561,26 @@ static void cs46xx_dsp_proc_task_tree_read (struct snd_info_entry *entry, } } - snd_iprintf(buffer,"\n"); + snd_iprintf(buffer,"\n"); mutex_unlock(&chip->spos_mutex); } -static void cs46xx_dsp_proc_scb_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_scb_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_cs46xx *chip = entry->private_data; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i; mutex_lock(&chip->spos_mutex); snd_iprintf(buffer, "SCB's:\n"); - for ( i = 0; i < ins->nscb; ++i) { + for (i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) continue; snd_iprintf(buffer,"\n%04x %s:\n\n",ins->scbs[i].address,ins->scbs[i].scb_name); if (ins->scbs[i].parent_scb_ptr != NULL) { - snd_iprintf(buffer,"parent [%s:%04x] ", + snd_iprintf(buffer,"parent [%s:%04x] ", ins->scbs[i].parent_scb_ptr->scb_name, ins->scbs[i].parent_scb_ptr->address); } else snd_iprintf(buffer,"parent [none] "); @@ -598,22 +598,22 @@ static void cs46xx_dsp_proc_scb_read (struct snd_info_entry *entry, mutex_unlock(&chip->spos_mutex); } -static void cs46xx_dsp_proc_parameter_dump_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_parameter_dump_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_cs46xx *chip = entry->private_data; - /*struct dsp_spos_instance * ins = chip->dsp_spos_instance; */ + /*struct dsp_spos_instance *ins = chip->dsp_spos_instance; */ unsigned int i, col = 0; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; - struct dsp_symbol_entry * symbol; + struct dsp_symbol_entry *symbol; - for (i = 0;i < DSP_PARAMETER_BYTE_SIZE; i += sizeof(u32),col ++) { + for (i = 0; i < DSP_PARAMETER_BYTE_SIZE; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; } - if ( (symbol = cs46xx_dsp_lookup_symbol_addr (chip,i / sizeof(u32), SYMBOL_PARAMETER)) != NULL) { + if ((symbol = cs46xx_dsp_lookup_symbol_addr (chip,i / sizeof(u32), SYMBOL_PARAMETER)) != NULL) { col = 0; snd_iprintf (buffer,"\n%s:\n",symbol->symbol_name); } @@ -626,15 +626,15 @@ static void cs46xx_dsp_proc_parameter_dump_read (struct snd_info_entry *entry, } } -static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_sample_dump_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_cs46xx *chip = entry->private_data; - int i,col = 0; + int i, col = 0; void __iomem *dst = chip->region.idx[2].remap_addr; snd_iprintf(buffer,"PCMREADER:\n"); - for (i = PCM_READER_BUF1;i < PCM_READER_BUF1 + 0x30; i += sizeof(u32),col ++) { + for (i = PCM_READER_BUF1; i < PCM_READER_BUF1 + 0x30; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -650,7 +650,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\nMIX_SAMPLE_BUF1:\n"); col = 0; - for (i = MIX_SAMPLE_BUF1;i < MIX_SAMPLE_BUF1 + 0x40; i += sizeof(u32),col ++) { + for (i = MIX_SAMPLE_BUF1; i < MIX_SAMPLE_BUF1 + 0x40; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -665,12 +665,12 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\nSRC_TASK_SCB1:\n"); col = 0; - for (i = 0x2480 ; i < 0x2480 + 0x40 ; i += sizeof(u32),col ++) { + for (i = 0x2480; i < 0x2480 + 0x40; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; } - + if (col == 0) { snd_iprintf(buffer, "%04X ",i); } @@ -681,7 +681,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\nSPDIFO_BUFFER:\n"); col = 0; - for (i = SPDIFO_IP_OUTPUT_BUFFER1;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x30; i += sizeof(u32),col ++) { + for (i = SPDIFO_IP_OUTPUT_BUFFER1; i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x30; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -697,7 +697,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\n...\n"); col = 0; - for (i = SPDIFO_IP_OUTPUT_BUFFER1+0xD0;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x110; i += sizeof(u32),col ++) { + for (i = SPDIFO_IP_OUTPUT_BUFFER1+0xD0; i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x110; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -713,7 +713,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\nOUTPUT_SNOOP:\n"); col = 0; - for (i = OUTPUT_SNOOP_BUFFER;i < OUTPUT_SNOOP_BUFFER + 0x40; i += sizeof(u32),col ++) { + for (i = OUTPUT_SNOOP_BUFFER; i < OUTPUT_SNOOP_BUFFER + 0x40; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -728,7 +728,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\nCODEC_INPUT_BUF1: \n"); col = 0; - for (i = CODEC_INPUT_BUF1;i < CODEC_INPUT_BUF1 + 0x40; i += sizeof(u32),col ++) { + for (i = CODEC_INPUT_BUF1; i < CODEC_INPUT_BUF1 + 0x40; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -743,7 +743,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, #if 0 snd_iprintf(buffer,"\nWRITE_BACK_BUF1: \n"); col = 0; - for (i = WRITE_BACK_BUF1;i < WRITE_BACK_BUF1 + 0x40; i += sizeof(u32),col ++) { + for (i = WRITE_BACK_BUF1; i < WRITE_BACK_BUF1 + 0x40; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -759,7 +759,7 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, snd_iprintf(buffer,"\nSPDIFI_IP_OUTPUT_BUFFER1: \n"); col = 0; - for (i = SPDIFI_IP_OUTPUT_BUFFER1;i < SPDIFI_IP_OUTPUT_BUFFER1 + 0x80; i += sizeof(u32),col ++) { + for (i = SPDIFI_IP_OUTPUT_BUFFER1; i < SPDIFI_IP_OUTPUT_BUFFER1 + 0x80; i += sizeof(u32), col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; @@ -768,16 +768,16 @@ static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry, if (col == 0) { snd_iprintf(buffer, "%04X ",i); } - + snd_iprintf(buffer,"%08X ",readl(dst + i)); } snd_iprintf(buffer,"\n"); } -int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) +int cs46xx_dsp_proc_init(struct snd_card *card, struct snd_cs46xx *chip) { struct snd_info_entry *entry; - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i; ins->snd_card = card; @@ -785,7 +785,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; - + if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -808,7 +808,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) } } ins->proc_sym_info_entry = entry; - + if ((entry = snd_info_create_card_entry(card, "spos_modules", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; @@ -881,9 +881,9 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) return 0; } -int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) +int cs46xx_dsp_proc_done(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i; snd_info_free_entry(ins->proc_sym_info_entry); @@ -907,7 +907,7 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) mutex_lock(&chip->spos_mutex); for (i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) continue; - cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); + cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i)); } mutex_unlock(&chip->spos_mutex); @@ -919,10 +919,10 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) #endif /* CONFIG_PROC_FS */ static int debug_tree; -static void _dsp_create_task_tree (struct snd_cs46xx *chip, u32 * task_data, +static void _dsp_create_task_tree(struct snd_cs46xx *chip, u32 *task_data, u32 dest, int size) { - void __iomem *spdst = chip->region.idx[1].remap_addr + + void __iomem *spdst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET + dest * sizeof(u32); int i; @@ -934,9 +934,9 @@ static void _dsp_create_task_tree (struct snd_cs46xx *chip, u32 * task_data, } static int debug_scb; -static void _dsp_create_scb (struct snd_cs46xx *chip, u32 * scb_data, u32 dest) +static void _dsp_create_scb(struct snd_cs46xx *chip, u32 *scb_data, u32 dest) { - void __iomem *spdst = chip->region.idx[1].remap_addr + + void __iomem *spdst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET + dest * sizeof(u32); int i; @@ -947,7 +947,7 @@ static void _dsp_create_scb (struct snd_cs46xx *chip, u32 * scb_data, u32 dest) } } -static int find_free_scb_index (struct dsp_spos_instance * ins) +static int find_free_scb_index(struct dsp_spos_instance *ins) { int index = ins->nscb, i; @@ -961,10 +961,10 @@ static int find_free_scb_index (struct dsp_spos_instance * ins) return index; } -static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * name, u32 dest) +static struct dsp_scb_descriptor *_map_scb(struct snd_cs46xx *chip, char *name, u32 dest) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * desc = NULL; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *desc = NULL; int index; if (ins->nscb == DSP_MAX_SCB_DESC - 1) { @@ -995,10 +995,10 @@ static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * nam } static struct dsp_task_descriptor * -_map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) +_map_task_tree(struct snd_cs46xx *chip, char *name, u32 dest, u32 size) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_task_descriptor * desc = NULL; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_task_descriptor *desc = NULL; if (ins->ntask == DSP_MAX_TASK_DESC - 1) { snd_printk(KERN_ERR "dsp_spos: got no place for other TASK\n"); @@ -1023,9 +1023,9 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) } struct dsp_scb_descriptor * -cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest) +cs46xx_dsp_create_scb(struct snd_cs46xx *chip, char *name, u32 *scb_data, u32 dest) { - struct dsp_scb_descriptor * desc; + struct dsp_scb_descriptor *desc; desc = _map_scb (chip,name,dest); if (desc) { @@ -1040,10 +1040,10 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 static struct dsp_task_descriptor * -cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_data, +cs46xx_dsp_create_task_tree(struct snd_cs46xx *chip, char *name, u32 *task_data, u32 dest, int size) { - struct dsp_task_descriptor * desc; + struct dsp_task_descriptor *desc; desc = _map_task_tree (chip,name,dest,size); if (desc) { @@ -1056,28 +1056,28 @@ cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_da return desc; } -int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) +int cs46xx_dsp_scb_and_task_init(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_symbol_entry * fg_task_tree_header_code; - struct dsp_symbol_entry * task_tree_header_code; - struct dsp_symbol_entry * task_tree_thread; - struct dsp_symbol_entry * null_algorithm; - struct dsp_symbol_entry * magic_snoop_task; - - struct dsp_scb_descriptor * timing_master_scb; - struct dsp_scb_descriptor * codec_out_scb; - struct dsp_scb_descriptor * codec_in_scb; - struct dsp_scb_descriptor * src_task_scb; - struct dsp_scb_descriptor * master_mix_scb; - struct dsp_scb_descriptor * rear_mix_scb; - struct dsp_scb_descriptor * record_mix_scb; - struct dsp_scb_descriptor * write_back_scb; - struct dsp_scb_descriptor * vari_decimate_scb; - struct dsp_scb_descriptor * rear_codec_out_scb; - struct dsp_scb_descriptor * clfe_codec_out_scb; - struct dsp_scb_descriptor * magic_snoop_scb; - + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_symbol_entry *fg_task_tree_header_code; + struct dsp_symbol_entry *task_tree_header_code; + struct dsp_symbol_entry *task_tree_thread; + struct dsp_symbol_entry *null_algorithm; + struct dsp_symbol_entry *magic_snoop_task; + + struct dsp_scb_descriptor *timing_master_scb; + struct dsp_scb_descriptor *codec_out_scb; + struct dsp_scb_descriptor *codec_in_scb; + struct dsp_scb_descriptor *src_task_scb; + struct dsp_scb_descriptor *master_mix_scb; + struct dsp_scb_descriptor *rear_mix_scb; + struct dsp_scb_descriptor *record_mix_scb; + struct dsp_scb_descriptor *write_back_scb; + struct dsp_scb_descriptor *vari_decimate_scb; + struct dsp_scb_descriptor *rear_codec_out_scb; + struct dsp_scb_descriptor *clfe_codec_out_scb; + struct dsp_scb_descriptor *magic_snoop_scb; + int fifo_addr, fifo_span, valid_slots; static struct dsp_spos_control_block sposcb = { @@ -1107,18 +1107,18 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) return -EIO; } - fg_task_tree_header_code = cs46xx_dsp_lookup_symbol(chip, "FGTASKTREEHEADERCODE", SYMBOL_CODE); + fg_task_tree_header_code = cs46xx_dsp_lookup_symbol(chip, "FGTASKTREEHEADERCODE", SYMBOL_CODE); if (fg_task_tree_header_code == NULL) { snd_printk(KERN_ERR "dsp_spos: symbol FGTASKTREEHEADERCODE not found\n"); return -EIO; } - task_tree_header_code = cs46xx_dsp_lookup_symbol(chip, "TASKTREEHEADERCODE", SYMBOL_CODE); + task_tree_header_code = cs46xx_dsp_lookup_symbol(chip, "TASKTREEHEADERCODE", SYMBOL_CODE); if (task_tree_header_code == NULL) { snd_printk(KERN_ERR "dsp_spos: symbol TASKTREEHEADERCODE not found\n"); return -EIO; } - + task_tree_thread = cs46xx_dsp_lookup_symbol(chip, "TASKTREETHREAD", SYMBOL_CODE); if (task_tree_thread == NULL) { snd_printk(KERN_ERR "dsp_spos: symbol TASKTREETHREAD not found\n"); @@ -1130,7 +1130,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) snd_printk(KERN_ERR "dsp_spos: symbol MAGICSNOOPTASK not found\n"); return -EIO; } - + { /* create the null SCB */ static struct dsp_generic_scb null_scb = { @@ -1165,14 +1165,14 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) DSP_SPOS_DC_DC, DSP_SPOS_DC_DC, DSP_SPOS_DC,DSP_SPOS_DC }, - + { - BG_TREE_SCB_ADDR,TIMINGMASTER_SCB_ADDR, + BG_TREE_SCB_ADDR,TIMINGMASTER_SCB_ADDR, 0, - FG_TASK_HEADER_ADDR + TCBData, + FG_TASK_HEADER_ADDR + TCBData, }, - { + { 4,0, 1,0, 2,SPOSCB_ADDR + HFGFlags, @@ -1215,9 +1215,9 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) DSP_SPOS_DCDC, DSP_SPOS_DCDC, DSP_SPOS_DCDC, - DSP_SPOS_DCDC - }, - { + DSP_SPOS_DCDC + }, + { FG_INTERVAL_TIMER_PERIOD,DSP_SPOS_UU, 0,0 } @@ -1241,14 +1241,14 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) DSP_SPOS_DC_DC, DSP_SPOS_DC_DC, DSP_SPOS_DC,DSP_SPOS_DC }, - + { NULL_SCB_ADDR,NULL_SCB_ADDR, /* Set up the background to do nothing */ 0, BG_TREE_SCB_ADDR + TCBData, }, - { + { 9999,0, 0,1, 0,SPOSCB_ADDR + HFGFlags, @@ -1291,9 +1291,9 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) DSP_SPOS_DCDC, DSP_SPOS_DCDC, DSP_SPOS_DCDC, - DSP_SPOS_DCDC - }, - { + DSP_SPOS_DCDC + }, + { BG_INTERVAL_TIMER_PERIOD,DSP_SPOS_UU, 0,0 } @@ -1344,7 +1344,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) 0x00020000, 0x0000ffff }; - + if (!cs46xx_dsp_create_task_tree(chip, NULL, (u32 *)&mix2_ostream_spb, WRITE_BACK_SPB, 2)) @@ -1376,7 +1376,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) goto _fail_end; if (chip->nr_ac97_codecs == 1) { - /* output on slot 5 and 11 + /* output on slot 5 and 11 on primary CODEC */ fifo_addr = 0x20; fifo_span = 0x60; @@ -1384,7 +1384,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) /* enable slot 5 and 11 */ valid_slots |= ACOSV_SLV5 | ACOSV_SLV11; } else { - /* output on slot 7 and 8 + /* output on slot 7 and 8 on secondary CODEC */ fifo_addr = 0x40; fifo_span = 0x10; @@ -1398,8 +1398,8 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) REAR_CODECOUT_SCB_ADDR,codec_in_scb, SCB_ON_PARENT_NEXT_SCB); if (!rear_codec_out_scb) goto _fail_end; - - + + /* create the rear PCM channel mixer SCB */ rear_mix_scb = cs46xx_dsp_create_mix_only_scb(chip,"RearMixerSCB", MIX_SAMPLE_BUF3, @@ -1408,9 +1408,9 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) SCB_ON_PARENT_SUBLIST_SCB); ins->rear_mix_scb = rear_mix_scb; if (!rear_mix_scb) goto _fail_end; - + if (chip->nr_ac97_codecs == 2) { - /* create CODEC tasklet for rear Center/LFE output + /* create CODEC tasklet for rear Center/LFE output slot 6 and 9 on seconadry CODEC */ clfe_codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,"CodecOutSCB_CLFE",0x0030,0x0030, CLFE_MIXER_SCB_ADDR, @@ -1418,8 +1418,8 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) rear_codec_out_scb, SCB_ON_PARENT_NEXT_SCB); if (!clfe_codec_out_scb) goto _fail_end; - - + + /* create the rear PCM channel mixer SCB */ ins->center_lfe_mix_scb = cs46xx_dsp_create_mix_only_scb(chip,"CLFEMixerSCB", MIX_SAMPLE_BUF4, @@ -1445,7 +1445,7 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) clfe_codec_out_scb, SCB_ON_PARENT_NEXT_SCB); - + if (!magic_snoop_scb) goto _fail_end; ins->ref_snoop_scb = magic_snoop_scb; @@ -1473,19 +1473,19 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) cs46xx_dsp_async_init(chip,timing_master_scb); return 0; - _fail_end: +_fail_end: snd_printk(KERN_ERR "dsp_spos: failed to setup SCB's in DSP\n"); return -EINVAL; } -static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, - struct dsp_scb_descriptor * fg_entry) +static int cs46xx_dsp_async_init(struct snd_cs46xx *chip, + struct dsp_scb_descriptor *fg_entry) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_symbol_entry * s16_async_codec_input_task; - struct dsp_symbol_entry * spdifo_task; - struct dsp_symbol_entry * spdifi_task; - struct dsp_scb_descriptor * spdifi_scb_desc, * spdifo_scb_desc, * async_codec_scb_desc; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_symbol_entry *s16_async_codec_input_task; + struct dsp_symbol_entry *spdifo_task; + struct dsp_symbol_entry *spdifi_task; + struct dsp_scb_descriptor *spdifi_scb_desc, *spdifo_scb_desc, *async_codec_scb_desc; s16_async_codec_input_task = cs46xx_dsp_lookup_symbol(chip, "S16_ASYNCCODECINPUTTASK", SYMBOL_CODE); if (s16_async_codec_input_task == NULL) { @@ -1509,10 +1509,10 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, struct dsp_spdifoscb spdifo_scb = { /* 0 */ DSP_SPOS_UUUU, { - /* 1 */ 0xb0, - /* 2 */ 0, - /* 3 */ 0, - /* 4 */ 0, + /* 1 */ 0xb0, + /* 2 */ 0, + /* 3 */ 0, + /* 4 */ 0, }, /* NOTE: the SPDIF output task read samples in mono format, the AsynchFGTxSCB task writes to buffer @@ -1520,9 +1520,9 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, */ /* 5 */ RSCONFIG_SAMPLE_16MONO + RSCONFIG_MODULO_256, /* 6 */ ( SPDIFO_IP_OUTPUT_BUFFER1 << 0x10 ) | 0xFFFC, - /* 7 */ 0,0, - /* 8 */ 0, - /* 9 */ FG_TASK_HEADER_ADDR, NULL_SCB_ADDR, + /* 7 */ 0,0, + /* 8 */ 0, + /* 9 */ FG_TASK_HEADER_ADDR, NULL_SCB_ADDR, /* A */ spdifo_task->address, SPDIFO_SCB_INST + SPDIFOFIFOPointer, { @@ -1539,7 +1539,7 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, /* 0 */ DSP_SPOS_UULO,DSP_SPOS_UUHI, /* 1 */ 0, /* 2 */ 0, - /* 3 */ 1,4000, /* SPDIFICountLimit SPDIFICount */ + /* 3 */ 1,4000, /* SPDIFICountLimit SPDIFICount */ /* 4 */ DSP_SPOS_UUUU, /* SPDIFIStatusData */ /* 5 */ 0,DSP_SPOS_UUHI, /* StatusData, Free4 */ /* 6 */ DSP_SPOS_UUUU, /* Free3 */ @@ -1549,8 +1549,8 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, /* A */ spdifi_task->address, SPDIFI_SCB_INST + SPDIFIFIFOPointer, /* NOTE: The SPDIF input task write the sample in mono - format from the HW FIFO, the AsynchFGRxSCB task reads - them in stereo + format from the HW FIFO, the AsynchFGRxSCB task reads + them in stereo */ /* B */ RSCONFIG_SAMPLE_16MONO + RSCONFIG_MODULO_128, /* C */ (SPDIFI_IP_OUTPUT_BUFFER1 << 0x10) | 0xFFFC, @@ -1573,40 +1573,40 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, /* 9 */ SPDIFI_SCB_INST,NULL_SCB_ADDR, /* A */ s16_async_codec_input_task->address, HFG_TREE_SCB + AsyncCIOFIFOPointer, - + /* B */ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64, /* C */ (ASYNC_IP_OUTPUT_BUFFER1 << 0x10), /*(ASYNC_IP_OUTPUT_BUFFER1 << 0x10) | 0xFFFC,*/ - + #ifdef UseASER1Input - /* short AsyncCIFIFOPointer:AsyncCIStatRegAddr; + /* short AsyncCIFIFOPointer:AsyncCIStatRegAddr; Init. 0000:8042: for ASER1 0000:8044: for ASER2 */ /* D */ 0x8042,0, - + /* short AsyncCIStMoFormat:AsyncCIFIFOBaseAddr; Init 1 stero:8050 ASER1 Init 0 mono:8070 ASER2 Init 1 Stereo : 0100 ASER1 (Set by script) */ /* E */ 0x0100,0x0001, - + #endif - + #ifdef UseASER2Input /* short AsyncCIFIFOPointer:AsyncCIStatRegAddr; Init. 0000:8042: for ASER1 0000:8044: for ASER2 */ /* D */ 0x8044,0, - + /* short AsyncCIStMoFormat:AsyncCIFIFOBaseAddr; Init 1 stero:8050 ASER1 Init 0 mono:8070 ASER2 Init 1 Stereo : 0100 ASER1 (Set by script) */ /* E */ 0x0110,0x0001, - + #endif - + /* short AsyncCIOutputBufModulo:AsyncCIFree; - AsyncCIOutputBufModulo: The modulo size for + AsyncCIOutputBufModulo: The modulo size for the output buffer of this task */ /* F */ 0, /* DSP_SPOS_UUUU */ }; @@ -1647,15 +1647,15 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, cs46xx_dsp_proc_register_scb_desc (chip,async_codec_scb_desc); /* Async MASTER ENABLE, affects both SPDIF input and output */ - snd_cs46xx_pokeBA0(chip, BA0_ASER_MASTER, 0x1 ); + snd_cs46xx_pokeBA0(chip, BA0_ASER_MASTER, 0x1); } return 0; } -static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip) +static void cs46xx_dsp_disable_spdif_hw(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; /* set SPDIF output FIFO slot */ snd_cs46xx_pokeBA0(chip, BA0_ASER_FADDR, 0); @@ -1674,16 +1674,16 @@ static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip) ins->spdif_status_out &= ~DSP_SPDIF_STATUS_HW_ENABLED; } -int cs46xx_dsp_enable_spdif_hw (struct snd_cs46xx *chip) +int cs46xx_dsp_enable_spdif_hw(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; /* if hw-ctrl already enabled, turn off to reset logic ... */ cs46xx_dsp_disable_spdif_hw (chip); udelay(50); /* set SPDIF output FIFO slot */ - snd_cs46xx_pokeBA0(chip, BA0_ASER_FADDR, ( 0x8000 | ((SP_SPDOUT_FIFO >> 4) << 4) )); + snd_cs46xx_pokeBA0(chip, BA0_ASER_FADDR, ( 0x8000 | ((SP_SPDOUT_FIFO >> 4) << 4))); /* SPDIF output MASTER ENABLE */ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CONTROL, 0x80000000); @@ -1699,7 +1699,7 @@ int cs46xx_dsp_enable_spdif_hw (struct snd_cs46xx *chip) int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; /* turn on amplifier */ chip->active_ctrl(chip, 1); @@ -1712,7 +1712,7 @@ int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip) mutex_lock(&chip->spos_mutex); - if ( ! (ins->spdif_status_out & DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED) ) { + if (!(ins->spdif_status_out & DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED)) { /* time countdown enable */ cs46xx_poke_via_dsp (chip,SP_ASER_COUNTDOWN, 0x80000005); /* NOTE: 80000005 value is just magic. With all values @@ -1761,7 +1761,7 @@ int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip) int cs46xx_dsp_disable_spdif_in (struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(!ins->asynch_rx_scb)) return -EINVAL; @@ -1787,9 +1787,9 @@ int cs46xx_dsp_disable_spdif_in (struct snd_cs46xx *chip) return 0; } -int cs46xx_dsp_enable_pcm_capture (struct snd_cs46xx *chip) +int cs46xx_dsp_enable_pcm_capture(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(ins->pcm_input)) return -EINVAL; @@ -1804,9 +1804,9 @@ int cs46xx_dsp_enable_pcm_capture (struct snd_cs46xx *chip) return 0; } -int cs46xx_dsp_disable_pcm_capture (struct snd_cs46xx *chip) +int cs46xx_dsp_disable_pcm_capture(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(!ins->pcm_input)) return -EINVAL; @@ -1819,9 +1819,9 @@ int cs46xx_dsp_disable_pcm_capture (struct snd_cs46xx *chip) return 0; } -int cs46xx_dsp_enable_adc_capture (struct snd_cs46xx *chip) +int cs46xx_dsp_enable_adc_capture(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(ins->adc_input)) return -EINVAL; @@ -1836,9 +1836,9 @@ int cs46xx_dsp_enable_adc_capture (struct snd_cs46xx *chip) return 0; } -int cs46xx_dsp_disable_adc_capture (struct snd_cs46xx *chip) +int cs46xx_dsp_disable_adc_capture(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(!ins->adc_input)) return -EINVAL; @@ -1851,7 +1851,7 @@ int cs46xx_dsp_disable_adc_capture (struct snd_cs46xx *chip) return 0; } -int cs46xx_poke_via_dsp (struct snd_cs46xx *chip, u32 address, u32 data) +int cs46xx_poke_via_dsp(struct snd_cs46xx *chip, u32 address, u32 data) { u32 temp; int i; @@ -1860,19 +1860,19 @@ int cs46xx_poke_via_dsp (struct snd_cs46xx *chip, u32 address, u32 data) a rough guess from looking at the controller spec.) */ if (address < 0x8000 || address >= 0x9000) return -EINVAL; - + /* initialize the SP_IO_WRITE SCB with the data. */ temp = ( address << 16 ) | ( address & 0x0000FFFF); /* offset 0 <-- address2 : address1 */ snd_cs46xx_poke(chip,( SPIOWRITE_SCB_ADDR << 2), temp); snd_cs46xx_poke(chip,((SPIOWRITE_SCB_ADDR + 1) << 2), data); /* offset 1 <-- data1 */ snd_cs46xx_poke(chip,((SPIOWRITE_SCB_ADDR + 2) << 2), data); /* offset 1 <-- data2 */ - + /* Poke this location to tell the task to start */ snd_cs46xx_poke(chip,((SPIOWRITE_SCB_ADDR + 6) << 2), SPIOWRITE_SCB_ADDR << 0x10); /* Verify that the task ran */ - for (i=0; i<25; i++) { + for (i = 0; i < 25; i++) { udelay(125); temp = snd_cs46xx_peek(chip,((SPIOWRITE_SCB_ADDR + 6) << 2)); @@ -1888,13 +1888,13 @@ int cs46xx_poke_via_dsp (struct snd_cs46xx *chip, u32 address, u32 data) return 0; } -int cs46xx_dsp_set_dac_volume (struct snd_cs46xx * chip, u16 left, u16 right) +int cs46xx_dsp_set_dac_volume(struct snd_cs46xx *chip, u16 left, u16 right) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * scb; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *scb; mutex_lock(&chip->spos_mutex); - + /* main output */ scb = ins->master_mix_scb->sub_list_ptr; while (scb != ins->the_null_scb) { @@ -1917,9 +1917,9 @@ int cs46xx_dsp_set_dac_volume (struct snd_cs46xx * chip, u16 left, u16 right) return 0; } -int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) +int cs46xx_dsp_set_iec958_volume(struct snd_cs46xx *chip, u16 left, u16 right) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; mutex_lock(&chip->spos_mutex); @@ -1936,9 +1936,9 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) } #ifdef CONFIG_PM -int cs46xx_dsp_resume(struct snd_cs46xx * chip) +int cs46xx_dsp_resume(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int i, err; /* clear parameter, sample and code areas */ diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h index f9e169d..a005f85 100644 --- a/sound/pci/cs46xx/dsp_spos.h +++ b/sound/pci/cs46xx/dsp_spos.h @@ -189,9 +189,9 @@ enum wide_opcode { static inline u8 _wrap_all_bits (u8 val) { u8 wrapped; - + /* wrap all 8 bits */ - wrapped = + wrapped = ((val & 0x1 ) << 7) | ((val & 0x2 ) << 5) | ((val & 0x4 ) << 3) | @@ -205,13 +205,13 @@ static inline u8 _wrap_all_bits (u8 val) } static inline void cs46xx_dsp_spos_update_scb (struct snd_cs46xx * chip, - struct dsp_scb_descriptor * scb) + struct dsp_scb_descriptor * scb) { /* update nextSCB and subListPtr in SCB */ snd_cs46xx_poke(chip, (scb->address + SCBsubListPtr) << 2, (scb->sub_list_ptr->address << 0x10) | - (scb->next_scb_ptr->address)); + (scb->next_scb_ptr->address)); } static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index dd7c41b..ac14c46 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -21,7 +21,7 @@ */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/init.h> @@ -37,13 +37,13 @@ #include "dsp_spos.h" struct proc_scb_info { - struct dsp_scb_descriptor * scb_desc; + struct dsp_scb_descriptor *scb_desc; struct snd_cs46xx *chip; }; -static void remove_symbol (struct snd_cs46xx * chip, struct dsp_symbol_entry * symbol) +static void remove_symbol(struct snd_cs46xx *chip, struct dsp_symbol_entry *symbol) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; int symbol_index = (int)(symbol - ins->symbol_table.symbols); if (snd_BUG_ON(ins->symbol_table.nsymbols <= 0)) @@ -57,7 +57,7 @@ static void remove_symbol (struct snd_cs46xx * chip, struct dsp_symbol_entry * s if (symbol_index < ins->symbol_table.highest_frag_index) { ins->symbol_table.highest_frag_index = symbol_index; } - + if (symbol_index == ins->symbol_table.nsymbols - 1) ins->symbol_table.nsymbols --; @@ -68,14 +68,14 @@ static void remove_symbol (struct snd_cs46xx * chip, struct dsp_symbol_entry * s } #ifdef CONFIG_PROC_FS -static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, +static void cs46xx_dsp_proc_scb_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct proc_scb_info * scb_info = entry->private_data; - struct dsp_scb_descriptor * scb = scb_info->scb_desc; - struct dsp_spos_instance * ins; + struct proc_scb_info *scb_info = entry->private_data; + struct dsp_scb_descriptor *scb = scb_info->scb_desc; + struct dsp_spos_instance *ins; struct snd_cs46xx *chip = scb_info->chip; - int j,col; + int j, col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; ins = chip->dsp_spos_instance; @@ -83,22 +83,22 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, mutex_lock(&chip->spos_mutex); snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name); - for (col = 0,j = 0;j < 0x10; j++,col++) { + for (col = 0,j = 0; j < 0x10; j++, col++) { if (col == 4) { snd_iprintf(buffer,"\n"); col = 0; } snd_iprintf(buffer,"%08x ",readl(dst + (scb->address + j) * sizeof(u32))); } - + snd_iprintf(buffer,"\n"); if (scb->parent_scb_ptr != NULL) { - snd_iprintf(buffer,"parent [%s:%04x] ", + snd_iprintf(buffer,"parent [%s:%04x] ", scb->parent_scb_ptr->scb_name, scb->parent_scb_ptr->address); } else snd_iprintf(buffer,"parent [none] "); - + snd_iprintf(buffer,"sub_list_ptr [%s:%04x]\nnext_scb_ptr [%s:%04x] task_entry [%s:%04x]\n", scb->sub_list_ptr->scb_name, scb->sub_list_ptr->address, @@ -107,22 +107,22 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, scb->task_entry->symbol_name, scb->task_entry->address); - snd_iprintf(buffer,"index [%d] ref_count [%d]\n",scb->index,scb->ref_count); + snd_iprintf(buffer,"index [%d] ref_count [%d]\n",scb->index,scb->ref_count); mutex_unlock(&chip->spos_mutex); } #endif -static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb) +static void _dsp_unlink_scb(struct snd_cs46xx *chip, struct dsp_scb_descriptor *scb) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; unsigned long flags; - if ( scb->parent_scb_ptr ) { + if (scb->parent_scb_ptr) { /* unlink parent SCB */ if (snd_BUG_ON(scb->parent_scb_ptr->sub_list_ptr != scb && scb->parent_scb_ptr->next_scb_ptr != scb)) return; - + if (scb->parent_scb_ptr->sub_list_ptr == scb) { if (scb->next_scb_ptr == ins->the_null_scb) { @@ -153,7 +153,7 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor scb->next_scb_ptr = ins->the_null_scb; } - spin_lock_irqsave(&chip->reg_lock, flags); + spin_lock_irqsave(&chip->reg_lock, flags); /* update parent first entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr); @@ -166,21 +166,21 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor } } -static void _dsp_clear_sample_buffer (struct snd_cs46xx *chip, u32 sample_buffer_addr, - int dword_count) +static void _dsp_clear_sample_buffer(struct snd_cs46xx *chip, u32 sample_buffer_addr, + int dword_count) { void __iomem *dst = chip->region.idx[2].remap_addr + sample_buffer_addr; int i; - - for (i = 0; i < dword_count ; ++i ) { + + for (i = 0; i < dword_count; ++i) { writel(0, dst); dst += 4; - } + } } -void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb) +void cs46xx_dsp_remove_scb(struct snd_cs46xx *chip, struct dsp_scb_descriptor *scb) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; unsigned long flags; /* check integrety */ @@ -190,7 +190,7 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * return; #if 0 - /* can't remove a SCB with childs before + /* can't remove a SCB with childs before removing childs first */ if (snd_BUG_ON(scb->sub_list_ptr != ins->the_null_scb || scb->next_scb_ptr != ins->the_null_scb)) @@ -221,7 +221,7 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * #if 0 /* !!!! THIS IS A PIECE OF SHIT MADE BY ME !!! */ - for(i = scb->index + 1;i < ins->nscb; ++i) { + for (i = scb->index + 1; i < ins->nscb; ++i) { ins->scbs[i - 1].index = i - 1; } #endif @@ -229,10 +229,10 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * #ifdef CONFIG_PROC_FS -void cs46xx_dsp_proc_free_scb_desc (struct dsp_scb_descriptor * scb) +void cs46xx_dsp_proc_free_scb_desc(struct dsp_scb_descriptor *scb) { if (scb->proc_info) { - struct proc_scb_info * scb_info = scb->proc_info->private_data; + struct proc_scb_info *scb_info = scb->proc_info->private_data; snd_printdd("cs46xx_dsp_proc_free_scb_desc: freeing %s\n",scb->scb_name); @@ -243,18 +243,18 @@ void cs46xx_dsp_proc_free_scb_desc (struct dsp_scb_descriptor * scb) } } -void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, - struct dsp_scb_descriptor * scb) +void cs46xx_dsp_proc_register_scb_desc(struct snd_cs46xx *chip, + struct dsp_scb_descriptor *scb) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; struct snd_info_entry * entry; - struct proc_scb_info * scb_info; + struct proc_scb_info *scb_info; /* register to proc */ if (ins->snd_card != NULL && ins->proc_dsp_dir != NULL && scb->proc_info == NULL) { - - if ((entry = snd_info_create_card_entry(ins->snd_card, scb->scb_name, + + if ((entry = snd_info_create_card_entry(ins->snd_card, scb->scb_name, ins->proc_dsp_dir)) != NULL) { scb_info = kmalloc(sizeof(struct proc_scb_info), GFP_KERNEL); if (!scb_info) { @@ -265,13 +265,13 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, scb_info->chip = chip; scb_info->scb_desc = scb; - + entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = scb_info; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - + entry->c.text.read = cs46xx_dsp_proc_scb_info_read; - + if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); kfree (scb_info); @@ -284,22 +284,22 @@ out: } #endif /* CONFIG_PROC_FS */ -static struct dsp_scb_descriptor * -_dsp_create_generic_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest, - struct dsp_symbol_entry * task_entry, - struct dsp_scb_descriptor * parent_scb, +static struct dsp_scb_descriptor * +_dsp_create_generic_scb(struct snd_cs46xx *chip, char *name, u32 *scb_data, u32 dest, + struct dsp_symbol_entry *task_entry, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * scb; - + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *scb; + unsigned long flags; if (snd_BUG_ON(!ins->the_null_scb)) return NULL; /* fill the data that will be wroten to DSP */ - scb_data[SCBsubListPtr] = + scb_data[SCBsubListPtr] = (ins->the_null_scb->address << 0x10) | ins->the_null_scb->address; scb_data[SCBfuncEntryPtr] &= 0xFFFF0000; @@ -316,7 +316,7 @@ _dsp_create_generic_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u scb->parent_scb_ptr = parent_scb; scb->task_entry = task_entry; - + /* update parent SCB */ if (scb->parent_scb_ptr) { #if 0 @@ -356,31 +356,31 @@ _dsp_create_generic_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u return scb; } -static struct dsp_scb_descriptor * -cs46xx_dsp_create_generic_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, - u32 dest, char * task_entry_name, - struct dsp_scb_descriptor * parent_scb, +static struct dsp_scb_descriptor * +cs46xx_dsp_create_generic_scb(struct snd_cs46xx *chip, char *name, u32 *scb_data, + u32 dest, char *task_entry_name, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_symbol_entry * task_entry; + struct dsp_symbol_entry *task_entry; task_entry = cs46xx_dsp_lookup_symbol (chip,task_entry_name, SYMBOL_CODE); - + if (task_entry == NULL) { snd_printk (KERN_ERR "dsp_spos: symbol %s not found\n",task_entry_name); return NULL; } - + return _dsp_create_generic_scb (chip,name,scb_data,dest,task_entry, parent_scb,scb_child_type); } -struct dsp_scb_descriptor * -cs46xx_dsp_create_timing_master_scb (struct snd_cs46xx *chip) +struct dsp_scb_descriptor * +cs46xx_dsp_create_timing_master_scb(struct snd_cs46xx *chip) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_timing_master_scb timing_master_scb = { { 0, 0, @@ -400,8 +400,8 @@ cs46xx_dsp_create_timing_master_scb (struct snd_cs46xx *chip) 0x0001,0x8000, /* fracSampAccumQm1:TMfrmsLeftInGroup */ 0x0001,0x0000, /* fracSampCorrectionQm1:TMfrmGroupLength */ 0x00060000 /* nSampPerFrmQ15 */ - }; - + }; + scb = cs46xx_dsp_create_generic_scb(chip,"TimingMasterSCBInst",(u32 *)&timing_master_scb, TIMINGMASTER_SCB_ADDR, "TIMINGMASTER",NULL,SCB_NO_PARENT); @@ -410,14 +410,14 @@ cs46xx_dsp_create_timing_master_scb (struct snd_cs46xx *chip) } -struct dsp_scb_descriptor * -cs46xx_dsp_create_codec_out_scb(struct snd_cs46xx * chip, char * codec_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_codec_out_scb(struct snd_cs46xx *chip, char *codec_name, u16 channel_disp, u16 fifo_addr, u16 child_scb_addr, - u32 dest, struct dsp_scb_descriptor * parent_scb, + u32 dest, struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_codec_output_scb codec_out_scb = { { 0, 0, @@ -439,23 +439,23 @@ cs46xx_dsp_create_codec_out_scb(struct snd_cs46xx * chip, char * codec_name, 0x0000,0x0080, /* (!AC97!) COexpVolChangeRate:COscaleShiftCount */ 0,child_scb_addr /* COreserved - need child scb to work with rom code */ }; - - + + scb = cs46xx_dsp_create_generic_scb(chip,codec_name,(u32 *)&codec_out_scb, dest,"S16_CODECOUTPUTTASK",parent_scb, scb_child_type); - + return scb; } -struct dsp_scb_descriptor * -cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx *chip, char *codec_name, u16 channel_disp, u16 fifo_addr, u16 sample_buffer_addr, - u32 dest, struct dsp_scb_descriptor * parent_scb, + u32 dest, struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; + struct dsp_scb_descriptor *scb; struct dsp_codec_input_scb codec_input_scb = { { 0, 0, @@ -469,7 +469,7 @@ cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name, 0, 0 }, - + #if 0 /* cs4620 */ SyncIOSCB,NULL_SCB_ADDR #else @@ -479,13 +479,13 @@ cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name, RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64, /* strmRsConfig */ sample_buffer_addr << 0x10, /* strmBufPtr; defined as a dword ptr, used as a byte ptr */ - channel_disp,fifo_addr, /* (!AC97!) leftChanBaseINaddr=AC97primary + channel_disp,fifo_addr, /* (!AC97!) leftChanBaseINaddr=AC97primary link input slot 3 :rightChanINdisp=""slot 4 */ - 0x0000,0x0000, /* (!AC97!) ????:scaleShiftCount; no shift needed + 0x0000,0x0000, /* (!AC97!) ????:scaleShiftCount; no shift needed because AC97 is already 20 bits */ 0x80008000 /* ??clw cwcgame.scb has 0 */ }; - + scb = cs46xx_dsp_create_generic_scb(chip,codec_name,(u32 *)&codec_input_scb, dest,"S16_CODECINPUTTASK",parent_scb, scb_child_type); @@ -493,18 +493,18 @@ cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name, } -static struct dsp_scb_descriptor * -cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx * chip, char * scb_name, +static struct dsp_scb_descriptor * +cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx *chip, char *scb_name, u16 sample_buffer_addr, u32 dest, int virtual_channel, u32 playback_hw_addr, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * scb; - + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *scb; + struct dsp_generic_scb pcm_reader_scb = { - + /* Play DMA Task xfers data from host buffer to SP buffer init/runtime variables: @@ -524,7 +524,7 @@ cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx * chip, char * scb_name, DMA_RQ_C2_AC_MONO_TO_STEREO 0x00002000L DMA_RQ_C2_AC_ENDIAN_CONVERT 0x00004000L DMA_RQ_C2_AC_SIGNED_CONVERT 0x00008000L - + HostBuffAddr: Host Buffer Physical Byte Address - SCB loc:3rd dword, Mask: 0xFFFFFFFFL aligned to dword boundary */ @@ -551,7 +551,7 @@ cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx * chip, char * scb_name, 0, 0, 0, - 0 + 0 }, /* Sublist pointer & next stream control block (SCB) link. */ NULL_SCB_ADDR,NULL_SCB_ADDR, @@ -568,7 +568,7 @@ cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx * chip, char * scb_name, /* Stream sample pointer & MAC-unit mode for this stream */ (sample_buffer_addr << 0x10), /* Fractional increment per output sample in the input sample buffer */ - 0, + 0, { /* Standard stereo volume control default muted */ @@ -580,34 +580,34 @@ cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx * chip, char * scb_name, if (ins->null_algorithm == NULL) { ins->null_algorithm = cs46xx_dsp_lookup_symbol (chip,"NULLALGORITHM", SYMBOL_CODE); - + if (ins->null_algorithm == NULL) { snd_printk (KERN_ERR "dsp_spos: symbol NULLALGORITHM not found\n"); return NULL; - } + } } scb = _dsp_create_generic_scb(chip,scb_name,(u32 *)&pcm_reader_scb, dest,ins->null_algorithm,parent_scb, scb_child_type); - + return scb; } #define GOF_PER_SEC 200 -struct dsp_scb_descriptor * -cs46xx_dsp_create_src_task_scb(struct snd_cs46xx * chip, char * scb_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_src_task_scb(struct snd_cs46xx *chip, char *scb_name, int rate, u16 src_buffer_addr, u16 src_delay_buffer_addr, u32 dest, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type, int pass_through) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * scb; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *scb; unsigned int tmp1, tmp2; unsigned int phiIncr; unsigned int correctionPerGOF, correctionPerSec; @@ -650,35 +650,35 @@ cs46xx_dsp_create_src_task_scb(struct snd_cs46xx * chip, char * scb_name, 0x0000,0x0000, src_buffer_addr,1, correctionPerGOF,correctionPerSec, - RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32, - 0x0000,src_delay_buffer_addr, - 0x0, + RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32, + 0x0000,src_delay_buffer_addr, + 0x0, 0x080,(src_delay_buffer_addr + (24 * 4)), 0,0, /* next_scb, sub_list_ptr */ 0,0, /* entry, this_spb */ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8, src_buffer_addr << 0x10, phiIncr, - { + { 0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left, 0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left } }; - + if (ins->s16_up == NULL) { ins->s16_up = cs46xx_dsp_lookup_symbol (chip,"S16_UPSRC", SYMBOL_CODE); - + if (ins->s16_up == NULL) { snd_printk (KERN_ERR "dsp_spos: symbol S16_UPSRC not found\n"); return NULL; - } + } } - + /* clear buffers */ _dsp_clear_sample_buffer (chip,src_buffer_addr,8); _dsp_clear_sample_buffer (chip,src_delay_buffer_addr,32); - + if (pass_through) { /* wont work with any other rate than the native DSP rate */ @@ -700,13 +700,13 @@ cs46xx_dsp_create_src_task_scb(struct snd_cs46xx * chip, char * scb_name, } #if 0 /* not used */ -struct dsp_scb_descriptor * -cs46xx_dsp_create_filter_scb(struct snd_cs46xx * chip, char * scb_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_filter_scb(struct snd_cs46xx *chip, char *scb_name, u16 buffer_addr, u32 dest, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_filter_scb filter_scb = { .a0_right = 0x41a9, .a0_left = 0x41a9, @@ -714,7 +714,7 @@ cs46xx_dsp_create_filter_scb(struct snd_cs46xx * chip, char * scb_name, .a1_left = 0xb8e4, .a2_right = 0x3e55, .a2_left = 0x3e55, - + .filter_unused3 = 0x0000, .filter_unused2 = 0x0000, @@ -750,14 +750,14 @@ cs46xx_dsp_create_filter_scb(struct snd_cs46xx * chip, char * scb_name, } #endif /* not used */ -struct dsp_scb_descriptor * -cs46xx_dsp_create_mix_only_scb(struct snd_cs46xx * chip, char * scb_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_mix_only_scb(struct snd_cs46xx *chip, char *scb_name, u16 mix_buffer_addr, u32 dest, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_mix_only_scb master_mix_scb = { /* 0 */ { 0, /* 1 */ 0, @@ -774,7 +774,7 @@ cs46xx_dsp_create_mix_only_scb(struct snd_cs46xx * chip, char * scb_name, /* 9 */ 0,0, /* A */ 0,0, /* B */ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32, - /* C */ (mix_buffer_addr + (16 * 4)) << 0x10, + /* C */ (mix_buffer_addr + (16 * 4)) << 0x10, /* D */ 0, { /* E */ 0x8000,0x8000, @@ -790,92 +790,92 @@ cs46xx_dsp_create_mix_only_scb(struct snd_cs46xx * chip, char * scb_name, } -struct dsp_scb_descriptor * -cs46xx_dsp_create_mix_to_ostream_scb(struct snd_cs46xx * chip, char * scb_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_mix_to_ostream_scb(struct snd_cs46xx *chip, char *scb_name, u16 mix_buffer_addr, u16 writeback_spb, u32 dest, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; + struct dsp_scb_descriptor *scb; struct dsp_mix2_ostream_scb mix2_ostream_scb = { /* Basic (non scatter/gather) DMA requestor (4 ints) */ - { + { DMA_RQ_C1_SOURCE_MOD64 + DMA_RQ_C1_DEST_ON_HOST + DMA_RQ_C1_DEST_MOD1024 + - DMA_RQ_C1_WRITEBACK_SRC_FLAG + + DMA_RQ_C1_WRITEBACK_SRC_FLAG + DMA_RQ_C1_WRITEBACK_DEST_FLAG + - 15, - + 15, + DMA_RQ_C2_AC_NONE + - DMA_RQ_C2_SIGNAL_DEST_PINGPONG + - - CS46XX_DSP_CAPTURE_CHANNEL, - DMA_RQ_SD_SP_SAMPLE_ADDR + - mix_buffer_addr, - 0x0 + DMA_RQ_C2_SIGNAL_DEST_PINGPONG + + + CS46XX_DSP_CAPTURE_CHANNEL, + DMA_RQ_SD_SP_SAMPLE_ADDR + + mix_buffer_addr, + 0x0 }, - + { 0, 0, 0, 0, 0, }, 0,0, 0,writeback_spb, - - RSCONFIG_DMA_ENABLE + - (19 << RSCONFIG_MAX_DMA_SIZE_SHIFT) + - + + RSCONFIG_DMA_ENABLE + + (19 << RSCONFIG_MAX_DMA_SIZE_SHIFT) + + ((dest >> 4) << RSCONFIG_STREAM_NUM_SHIFT) + - RSCONFIG_DMA_TO_HOST + + RSCONFIG_DMA_TO_HOST + RSCONFIG_SAMPLE_16STEREO + - RSCONFIG_MODULO_64, + RSCONFIG_MODULO_64, (mix_buffer_addr + (32 * 4)) << 0x10, - 1,0, + 1,0, 0x0001,0x0080, 0xFFFF,0 }; scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&mix2_ostream_scb, - + dest,"S16_MIX_TO_OSTREAM",parent_scb, scb_child_type); - + return scb; } -struct dsp_scb_descriptor * -cs46xx_dsp_create_vari_decimate_scb(struct snd_cs46xx * chip,char * scb_name, +struct dsp_scb_descriptor * +cs46xx_dsp_create_vari_decimate_scb(struct snd_cs46xx *chip,char *scb_name, u16 vari_buffer_addr0, u16 vari_buffer_addr1, u32 dest, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_vari_decimate_scb vari_decimate_scb = { 0x0028,0x00c8, 0x5555,0x0000, 0x0000,0x0000, vari_buffer_addr0,vari_buffer_addr1, - + 0x0028,0x00c8, - RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_256, - - 0xFF800000, + RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_256, + + 0xFF800000, 0, - 0x0080,vari_buffer_addr1 + (25 * 4), - - 0,0, + 0x0080,vari_buffer_addr1 + (25 * 4), + + 0,0, 0,0, RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8, - vari_buffer_addr0 << 0x10, - 0x04000000, + vari_buffer_addr0 << 0x10, + 0x04000000, { - 0x8000,0x8000, + 0x8000,0x8000, 0xFFFF,0xFFFF } }; @@ -883,19 +883,19 @@ cs46xx_dsp_create_vari_decimate_scb(struct snd_cs46xx * chip,char * scb_name, scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&vari_decimate_scb, dest,"VARIDECIMATE",parent_scb, scb_child_type); - + return scb; } -static struct dsp_scb_descriptor * -cs46xx_dsp_create_pcm_serial_input_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest, +static struct dsp_scb_descriptor * +cs46xx_dsp_create_pcm_serial_input_scb(struct snd_cs46xx *chip, char *scb_name, u32 dest, struct dsp_scb_descriptor * input_scb, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; + struct dsp_scb_descriptor *scb; struct dsp_pcm_serial_input_scb pcm_serial_input_scb = { @@ -931,15 +931,15 @@ cs46xx_dsp_create_pcm_serial_input_scb(struct snd_cs46xx * chip, char * scb_name } -static struct dsp_scb_descriptor * -cs46xx_dsp_create_asynch_fg_tx_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest, +static struct dsp_scb_descriptor * +cs46xx_dsp_create_asynch_fg_tx_scb(struct snd_cs46xx *chip, char *scb_name, u32 dest, u16 hfg_scb_address, u16 asynch_buffer_address, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; + struct dsp_scb_descriptor *scb; struct dsp_asynch_fg_tx_scb asynch_fg_tx_scb = { 0xfc00,0x03ff, /* Prototype sample buffer size of 256 dwords */ @@ -949,27 +949,27 @@ cs46xx_dsp_create_asynch_fg_tx_scb(struct snd_cs46xx * chip, char * scb_name, u3 0,0, /* Initialize current Delta and Consumer ptr adjustment count */ 0, /* Initialize accumulated Phi to 0 */ 0,0x2aab, /* Const 1/3 */ - + { 0, /* Define the unused elements */ 0, 0 }, - + 0,0, 0,dest + AFGTxAccumPhi, - + RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_256, /* Stereo, 256 dword */ (asynch_buffer_address) << 0x10, /* This should be automagically synchronized to the producer pointer */ - + /* There is no correct initial value, it will depend upon the detected rate etc */ 0x18000000, /* Phi increment for approx 32k operation */ 0x8000,0x8000, /* Volume controls are unused at this time */ 0x8000,0x8000 }; - + scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_tx_scb, dest,"ASYNCHFGTXCODE",parent_scb, scb_child_type); @@ -978,15 +978,15 @@ cs46xx_dsp_create_asynch_fg_tx_scb(struct snd_cs46xx * chip, char * scb_name, u3 } -struct dsp_scb_descriptor * -cs46xx_dsp_create_asynch_fg_rx_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest, +struct dsp_scb_descriptor * +cs46xx_dsp_create_asynch_fg_rx_scb(struct snd_cs46xx *chip, char *scb_name, u32 dest, u16 hfg_scb_address, u16 asynch_buffer_address, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * scb; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *scb; struct dsp_asynch_fg_rx_scb asynch_fg_rx_scb = { 0xfe00,0x01ff, /* Prototype sample buffer size of 128 dwords */ @@ -1001,18 +1001,18 @@ cs46xx_dsp_create_asynch_fg_rx_scb(struct snd_cs46xx * chip, char * scb_name, u3 0, 0 }, - + 0,0, 0,dest, - + RSCONFIG_MODULO_128 | RSCONFIG_SAMPLE_16STEREO, /* Stereo, 128 dword */ - ( (asynch_buffer_address + (16 * 4)) << 0x10), /* This should be automagically + ( (asynch_buffer_address + (16 * 4)) << 0x10), /* This should be automagically synchrinized to the producer pointer */ - + /* There is no correct initial value, it will depend upon the detected rate etc */ - 0x18000000, + 0x18000000, /* Set IEC958 input volume */ 0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left, @@ -1028,16 +1028,16 @@ cs46xx_dsp_create_asynch_fg_rx_scb(struct snd_cs46xx * chip, char * scb_name, u3 #if 0 /* not used */ -struct dsp_scb_descriptor * -cs46xx_dsp_create_output_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest, +struct dsp_scb_descriptor * +cs46xx_dsp_create_output_snoop_scb(struct snd_cs46xx *chip, char *scb_name, u32 dest, u16 snoop_buffer_address, struct dsp_scb_descriptor * snoop_scb, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_output_snoop_scb output_snoop_scb = { { 0, /* not used. Zero */ 0, @@ -1051,17 +1051,17 @@ cs46xx_dsp_create_output_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u3 0, 0 }, - + 0,0, 0,0, - + RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64, - snoop_buffer_address << 0x10, + snoop_buffer_address << 0x10, 0,0, 0, 0,snoop_scb->address }; - + scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&output_snoop_scb, dest,"OUTPUTSNOOP",parent_scb, scb_child_type); @@ -1070,13 +1070,13 @@ cs46xx_dsp_create_output_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u3 #endif /* not used */ -struct dsp_scb_descriptor * -cs46xx_dsp_create_spio_write_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest, - struct dsp_scb_descriptor * parent_scb, +struct dsp_scb_descriptor * +cs46xx_dsp_create_spio_write_scb(struct snd_cs46xx *chip, char *scb_name, u32 dest, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_spio_write_scb spio_write_scb = { 0,0, /* SPIOWAddress2:SPIOWAddress1; */ 0, /* SPIOWData1; */ @@ -1086,11 +1086,11 @@ cs46xx_dsp_create_spio_write_scb(struct snd_cs46xx * chip, char * scb_name, u32 0, /* SPIOWData4; */ 0,0, /* SPIOWDataPtr:Unused1; */ { 0,0 }, /* Unused2[2]; */ - + 0,0, /* SPIOWChildPtr:SPIOWSiblingPtr; */ 0,0, /* SPIOWThisPtr:SPIOWEntryPoint; */ - - { + + { 0, 0, 0, @@ -1107,14 +1107,14 @@ cs46xx_dsp_create_spio_write_scb(struct snd_cs46xx * chip, char * scb_name, u32 } struct dsp_scb_descriptor * -cs46xx_dsp_create_magic_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest, +cs46xx_dsp_create_magic_snoop_scb(struct snd_cs46xx *chip, char *scb_name, u32 dest, u16 snoop_buffer_address, struct dsp_scb_descriptor * snoop_scb, - struct dsp_scb_descriptor * parent_scb, + struct dsp_scb_descriptor *parent_scb, int scb_child_type) { - struct dsp_scb_descriptor * scb; - + struct dsp_scb_descriptor *scb; + struct dsp_magic_snoop_task magic_snoop_scb = { /* 0 */ 0, /* i0 */ /* 1 */ 0, /* i1 */ @@ -1143,10 +1143,10 @@ cs46xx_dsp_create_magic_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u32 } static struct dsp_scb_descriptor * -find_next_free_scb (struct snd_cs46xx * chip, struct dsp_scb_descriptor * from) +find_next_free_scb(struct snd_cs46xx *chip, struct dsp_scb_descriptor *from) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * scb = from; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *scb = from; while (scb->next_scb_ptr != ins->the_null_scb) { if (snd_BUG_ON(!scb->next_scb_ptr)) @@ -1198,9 +1198,9 @@ static u32 src_output_buffer_addr[DSP_MAX_SRC_NR] = { 0x2BA0, 0x2BC0, 0x2BE0, - 0x2D00, - 0x2D20, - 0x2D40, + 0x2D00, + 0x2D20, + 0x2D40, 0x2D60, 0x2D80, 0x2DA0, @@ -1228,16 +1228,16 @@ static u32 src_delay_buffer_addr[DSP_MAX_SRC_NR] = { }; struct dsp_pcm_channel_descriptor * -cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, - u32 sample_rate, void * private_data, +cs46xx_dsp_create_pcm_channel(struct snd_cs46xx *chip, + u32 sample_rate, void * private_data, u32 hw_dma_addr, int pcm_channel_id) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * src_scb = NULL, * pcm_scb, * mixer_scb = NULL; - struct dsp_scb_descriptor * src_parent_scb = NULL; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *src_scb = NULL, *pcm_scb, *mixer_scb = NULL; + struct dsp_scb_descriptor *src_parent_scb = NULL; - /* struct dsp_scb_descriptor * pcm_parent_scb; */ + /* struct dsp_scb_descriptor *pcm_parent_scb; */ char scb_name[DSP_MAX_SCB_NAME]; int i, pcm_index = -1, insert_point, src_index = -1, pass_through = 0; unsigned long flags; @@ -1278,19 +1278,19 @@ cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, if (!sample_rate) sample_rate = 44100; /* search for a already created SRC SCB with the same sample rate */ - for (i = 0; i < DSP_MAX_PCM_CHANNELS && + for (i = 0; i < DSP_MAX_PCM_CHANNELS && (pcm_index == -1 || src_scb == NULL); ++i) { - /* virtual channel reserved + /* virtual channel reserved for capture */ if (i == CS46XX_DSP_CAPTURE_CHANNEL) continue; if (ins->pcm_channels[i].active) { - if (!src_scb && + if (!src_scb && ins->pcm_channels[i].sample_rate == sample_rate && ins->pcm_channels[i].mixer_scb == mixer_scb) { src_scb = ins->pcm_channels[i].src_scb; - ins->pcm_channels[i].src_scb->ref_count ++; + ins->pcm_channels[i].src_scb->ref_count++; src_index = ins->pcm_channels[i].src_slot; } } else if (pcm_index == -1) { @@ -1330,7 +1330,7 @@ cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, } snprintf (scb_name,DSP_MAX_SCB_NAME,"SrcTask_SCB%d",src_index); - + snd_printdd( "dsp_spos: creating SRC \"%s\"\n",scb_name); src_scb = cs46xx_dsp_create_src_task_scb(chip,scb_name, sample_rate, @@ -1349,10 +1349,10 @@ cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, /* cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate); */ - ins->nsrc_scb ++; - } - - + ins->nsrc_scb++; + } + + snprintf (scb_name,DSP_MAX_SCB_NAME,"PCMReader_SCB%d",pcm_index); snd_printdd( "dsp_spos: creating PCM \"%s\" (%d)\n",scb_name, @@ -1365,14 +1365,14 @@ cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, pcm_index, /* virtual channel 0-31 */ hw_dma_addr, /* pcm hw addr */ NULL, /* parent SCB ptr */ - 0 /* insert point */ - ); + 0 /* insert point */ + ); if (!pcm_scb) { snd_printk (KERN_ERR "dsp_spos: failed to create PCMreaderSCB\n"); return NULL; } - + spin_lock_irqsave(&chip->reg_lock, flags); ins->pcm_channels[pcm_index].sample_rate = sample_rate; ins->pcm_channels[pcm_index].pcm_reader_scb = pcm_scb; @@ -1383,14 +1383,14 @@ cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, ins->pcm_channels[pcm_index].active = 1; ins->pcm_channels[pcm_index].pcm_slot = pcm_index; ins->pcm_channels[pcm_index].mixer_scb = mixer_scb; - ins->npcm_channels ++; + ins->npcm_channels++; spin_unlock_irqrestore(&chip->reg_lock, flags); return (ins->pcm_channels + pcm_index); } -int cs46xx_dsp_pcm_channel_set_period (struct snd_cs46xx * chip, - struct dsp_pcm_channel_descriptor * pcm_channel, +int cs46xx_dsp_pcm_channel_set_period(struct snd_cs46xx *chip, + struct dsp_pcm_channel_descriptor *pcm_channel, int period_size) { u32 temp = snd_cs46xx_peek (chip,pcm_channel->pcm_reader_scb->address << 2); @@ -1414,10 +1414,10 @@ int cs46xx_dsp_pcm_channel_set_period (struct snd_cs46xx * chip, break; case 64: temp |= DMA_RQ_C1_SOURCE_MOD32; - break; + break; case 32: temp |= DMA_RQ_C1_SOURCE_MOD16; - break; + break; default: snd_printdd ("period size (%d) not supported by HW\n", period_size); return -EINVAL; @@ -1428,7 +1428,7 @@ int cs46xx_dsp_pcm_channel_set_period (struct snd_cs46xx * chip, return 0; } -int cs46xx_dsp_pcm_ostream_set_period (struct snd_cs46xx * chip, +int cs46xx_dsp_pcm_ostream_set_period(struct snd_cs46xx *chip, int period_size) { u32 temp = snd_cs46xx_peek (chip,WRITEBACK_SCB_ADDR << 2); @@ -1452,10 +1452,10 @@ int cs46xx_dsp_pcm_ostream_set_period (struct snd_cs46xx * chip, break; case 64: temp |= DMA_RQ_C1_DEST_MOD32; - break; + break; case 32: temp |= DMA_RQ_C1_DEST_MOD16; - break; + break; default: snd_printdd ("period size (%d) not supported by HW\n", period_size); return -EINVAL; @@ -1466,10 +1466,10 @@ int cs46xx_dsp_pcm_ostream_set_period (struct snd_cs46xx * chip, return 0; } -void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip, - struct dsp_pcm_channel_descriptor * pcm_channel) +void cs46xx_dsp_destroy_pcm_channel(struct snd_cs46xx *chip, + struct dsp_pcm_channel_descriptor *pcm_channel) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; unsigned long flags; if (snd_BUG_ON(!pcm_channel->active || @@ -1499,8 +1499,8 @@ void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip, } } -int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip, - struct dsp_pcm_channel_descriptor * pcm_channel) +int cs46xx_dsp_pcm_unlink(struct snd_cs46xx *chip, + struct dsp_pcm_channel_descriptor *pcm_channel) { unsigned long flags; @@ -1525,12 +1525,12 @@ int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip, return 0; } -int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, - struct dsp_pcm_channel_descriptor * pcm_channel) +int cs46xx_dsp_pcm_link(struct snd_cs46xx *chip, + struct dsp_pcm_channel_descriptor *pcm_channel) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * parent_scb; - struct dsp_scb_descriptor * src_scb = pcm_channel->src_scb; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *parent_scb; + struct dsp_scb_descriptor *src_scb = pcm_channel->src_scb; unsigned long flags; spin_lock(&pcm_channel->src_scb->lock); @@ -1568,12 +1568,12 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, } struct dsp_scb_descriptor * -cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * source, - u16 addr, char * scb_name) +cs46xx_add_record_source(struct snd_cs46xx *chip, struct dsp_scb_descriptor * source, + u16 addr, char *scb_name) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * parent; - struct dsp_scb_descriptor * pcm_input; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *parent; + struct dsp_scb_descriptor *pcm_input; int insert_point; if (snd_BUG_ON(!ins->record_mixer_scb)) @@ -1594,7 +1594,7 @@ cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * s return pcm_input; } -int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) +int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor *src) { if (snd_BUG_ON(!src->parent_scb_ptr)) return -EINVAL; @@ -1607,10 +1607,10 @@ int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) return 0; } -int cs46xx_src_link(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) +int cs46xx_src_link(struct snd_cs46xx *chip, struct dsp_scb_descriptor *src) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; - struct dsp_scb_descriptor * parent_scb; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; + struct dsp_scb_descriptor *parent_scb; if (snd_BUG_ON(src->parent_scb_ptr)) return -EINVAL; @@ -1629,20 +1629,20 @@ int cs46xx_src_link(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) /* update entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,parent_scb); - + return 0; } -int cs46xx_dsp_enable_spdif_out (struct snd_cs46xx *chip) +int cs46xx_dsp_enable_spdif_out(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; - if ( ! (ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) ) { + if (!(ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED)) { cs46xx_dsp_enable_spdif_hw (chip); } /* dont touch anything if SPDIF is open */ - if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) { + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) { /* when cs46xx_iec958_post_close(...) is called it will call this function if necessary depending on this bit */ @@ -1659,8 +1659,8 @@ int cs46xx_dsp_enable_spdif_out (struct snd_cs46xx *chip) /* reset output snooper sample buffer pointer */ snd_cs46xx_poke (chip, (ins->ref_snoop_scb->address + 2) << 2, - (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10 ); - + (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10); + /* The asynch. transfer task */ ins->asynch_tx_scb = cs46xx_dsp_create_asynch_fg_tx_scb(chip,"AsynchFGTxSCB",ASYNCTX_SCB_ADDR, SPDIFO_SCB_INST, @@ -1674,8 +1674,8 @@ int cs46xx_dsp_enable_spdif_out (struct snd_cs46xx *chip) ins->ref_snoop_scb, ins->asynch_tx_scb, SCB_ON_PARENT_SUBLIST_SCB); - - + + if (!ins->spdif_pcm_input_scb) return -ENOMEM; /* monitor state */ @@ -1684,12 +1684,12 @@ int cs46xx_dsp_enable_spdif_out (struct snd_cs46xx *chip) return 0; } -int cs46xx_dsp_disable_spdif_out (struct snd_cs46xx *chip) +int cs46xx_dsp_disable_spdif_out(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; /* dont touch anything if SPDIF is open */ - if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) { + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) { ins->spdif_status_out &= ~DSP_SPDIF_STATUS_OUTPUT_ENABLED; return -EBUSY; } @@ -1723,18 +1723,18 @@ int cs46xx_dsp_disable_spdif_out (struct snd_cs46xx *chip) int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; - if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { + if (ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED) { /* remove AsynchFGTxSCB and and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ ins->spdif_status_out |= DSP_SPDIF_STATUS_OUTPUT_ENABLED; } - + /* if not enabled already */ - if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) ) { + if (!(ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED)) { cs46xx_dsp_enable_spdif_hw (chip); } @@ -1754,9 +1754,9 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) return 0; } -int cs46xx_iec958_post_close (struct snd_cs46xx *chip) +int cs46xx_iec958_post_close(struct snd_cs46xx *chip) { - struct dsp_spos_instance * ins = chip->dsp_spos_instance; + struct dsp_spos_instance *ins = chip->dsp_spos_instance; if (snd_BUG_ON(!ins->asynch_tx_scb)) return -EINVAL; @@ -1765,7 +1765,7 @@ int cs46xx_iec958_post_close (struct snd_cs46xx *chip) /* restore settings */ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default); - + /* deallocate stuff */ if (ins->spdif_pcm_input_scb != NULL) { cs46xx_dsp_remove_scb (chip,ins->spdif_pcm_input_scb); @@ -1779,9 +1779,9 @@ int cs46xx_iec958_post_close (struct snd_cs46xx *chip) _dsp_clear_sample_buffer(chip,SPDIFO_IP_OUTPUT_BUFFER1,256); /* restore state */ - if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { + if (ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED) { cs46xx_dsp_enable_spdif_out (chip); } - + return 0; } diff --git a/sound/pci/cs46xx/imgs/cwcbinhack.h b/sound/pci/cs46xx/imgs/cwcbinhack.h index f4d9368..0587ab3 100644 --- a/sound/pci/cs46xx/imgs/cwcbinhack.h +++ b/sound/pci/cs46xx/imgs/cwcbinhack.h @@ -1,4 +1,4 @@ -/* generated by Benny +/* generated by Benny MODIFY ON YOUR OWN RISK */ #ifndef __HEADER_cwcbinhack_H__

1 0

[alsa-devel] [PATCH 04/24] sound/pci: coding style fixes: ca0106
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ca0106/ca0106.h | 32 +++--- sound/pci/ca0106/ca0106_main.c | 262 +++++++++++++++++++------------------- sound/pci/ca0106/ca0106_mixer.c | 22 ++-- sound/pci/ca0106/ca0106_proc.c | 66 +++++----- sound/pci/ca0106/ca_midi.c | 14 +- sound/pci/ca0106/ca_midi.h | 6 +- 6 files changed, 201 insertions(+), 201 deletions(-) diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 74175fc..645980f 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -5,7 +5,7 @@ * * FEATURES currently supported: * See ca0106_main.c for features. - * + * * Changelog: * Support interrupts per period. * Removed noise from Center/LFE channel when in Analog mode. @@ -174,10 +174,10 @@ /********************************************************************************************************/ /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ /********************************************************************************************************/ - + /* Initally all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ - /* One list entry: 4 bytes for DMA address, + /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. * One list entry is 8 bytes long. * One list entry for each period in the buffer. @@ -219,7 +219,7 @@ * Playback mixer in enable [27:24] (one bit per channel) * Playback mixer out enable [31:28] (one bit per channel) */ -/* The Digital out jack is shared with the Center/LFE Analogue output. +/* The Digital out jack is shared with the Center/LFE Analogue output. * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. @@ -232,7 +232,7 @@ * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output */ /* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel. - * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs. + * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs. */ #define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */ #define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */ @@ -331,7 +331,7 @@ #define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */ #define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */ #define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */ - /* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3 + /* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3 * Record source select for channel 0 [18:16] * Record source select for channel 1 [22:20] * Record source select for channel 2 [26:24] @@ -432,7 +432,7 @@ * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source. * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) - * Record mixer output enable [12:10] + * Record mixer output enable [12:10] * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) * I2S output source select [18] (0=Audio from host, 1=Audio from SRC) @@ -445,7 +445,7 @@ * Not used [27] * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) - */ + */ /* Sample rate output control register Channel=1 * I2S Input 0 volume Right [7:0] * I2S Input 0 volume Left [15:8] @@ -467,7 +467,7 @@ * AC97 output enable [5:0] * I2S output enable [19:16] * SPDIF output enable [27:24] - */ + */ #define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */ #define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */ #define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */ @@ -502,14 +502,14 @@ #define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction #define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode -#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC +#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC #define I2C_A_ADC_READ 0x00000001 //To perform a read operation #define I2C_A_ADC_START 0x00000100 //Start I2C transaction #define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort #define I2C_A_ADC_LAST 0x00000400 //I2C last transaction #define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode -#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register +#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register #define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register #define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable @@ -708,13 +708,13 @@ struct snd_ca0106 { int snd_ca0106_mixer(struct snd_ca0106 *emu); int snd_ca0106_proc_init(struct snd_ca0106 * emu); -unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, - unsigned int reg, +unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, + unsigned int reg, unsigned int chn); -void snd_ca0106_ptr_write(struct snd_ca0106 *emu, - unsigned int reg, - unsigned int chn, +void snd_ca0106_ptr_write(struct snd_ca0106 *emu, + unsigned int reg, + unsigned int chn, unsigned int data); int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 6abe8a3..6d4534a 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -18,7 +18,7 @@ * So, to record from the MIC, set the MIC Playback volume to max, * unmute the MIC and turn up the MASTER Playback volume. * So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume. - * + * * The only playback controls that currently do anything are: - * Analog Front * Analog Rear @@ -26,12 +26,12 @@ * SPDIF Front * SPDIF Rear * SPDIF Center/LFE - * + * * For capture from Mic in or Line in. * Digital/Analog ( switch must be in Analog mode for CAPTURE. ) - * + * * CAPTURE feedback into PLAYBACK - * + * * Changelog: * Support interrupts per period. * Removed noise from Center/LFE channel when in Analog mode. @@ -108,7 +108,7 @@ * DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support) * SPDIF Out control switches between Mic in and SPDIF out. * No sound out or mic input working yet. - * + * * GENERAL INFO: * Model: SB0413 * P17 Chip: CA0106-DAT @@ -202,7 +202,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { /* AudigyLS[SB0310] */ { .serial = 0x10021102, .name = "AudigyLS [SB0310]", - .ac97 = 1 } , + .ac97 = 1 } , /* Unknown AudigyLS that also says SB0310 on it */ { .serial = 0x10051102, .name = "AudigyLS [SB0310b]", @@ -279,7 +279,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { /* hardware definition */ static struct snd_pcm_hardware snd_ca0106_playback_hw = { - .info = SNDRV_PCM_INFO_MMAP | + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -289,8 +289,8 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_RATE_192000), .rate_min = 48000, .rate_max = 192000, - .channels_min = 2, //1, - .channels_max = 2, //6, + .channels_min = 2, /* 1, */ + .channels_max = 2, /* 6, */ .buffer_bytes_max = ((65536 - 64) * 8), .period_bytes_min = 64, .period_bytes_max = (65536 - 64), @@ -300,7 +300,7 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { }; static struct snd_pcm_hardware snd_ca0106_capture_hw = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), @@ -319,13 +319,13 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; -unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, - unsigned int reg, +unsigned int snd_ca0106_ptr_read(struct snd_ca0106 *emu, + unsigned int reg, unsigned int chn) { unsigned long flags; unsigned int regptr, val; - + regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); @@ -335,9 +335,9 @@ unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, return val; } -void snd_ca0106_ptr_write(struct snd_ca0106 *emu, - unsigned int reg, - unsigned int chn, +void snd_ca0106_ptr_write(struct snd_ca0106 *emu, + unsigned int reg, + unsigned int chn, unsigned int data) { unsigned int regptr; @@ -351,7 +351,7 @@ void snd_ca0106_ptr_write(struct snd_ca0106 *emu, spin_unlock_irqrestore(&emu->emu_lock, flags); } -int snd_ca0106_spi_write(struct snd_ca0106 * emu, +int snd_ca0106_spi_write(struct snd_ca0106 *emu, unsigned int data) { unsigned int reset, set; @@ -398,7 +398,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; - // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); + /* snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); */ /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -407,8 +407,8 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, for (retry = 0; retry < 10; retry++) { /* Send the data to i2c */ - //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); - //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + /* tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); */ + /* tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); */ tmp = 0; tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD); snd_ca0106_ptr_write(emu, I2C_A, 0, tmp); @@ -416,7 +416,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, /* Wait till the transaction ends */ while (1) { status = snd_ca0106_ptr_read(emu, I2C_A, 0); - //snd_printk("I2C:status=0x%x\n", status); + /* snd_printk("I2C:status=0x%x\n", status); */ timeout++; if ((status & I2C_A_ADC_START) == 0) break; @@ -424,7 +424,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, if (timeout > 1000) break; } - //Read back and see if the transaction is successful + /* Read back and see if the transaction is successful */ if ((status & I2C_A_ADC_ABORT) == 0) break; } @@ -433,7 +433,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, snd_printk(KERN_ERR "Writing to ADC failed!\n"); return -EINVAL; } - + return 0; } @@ -496,18 +496,18 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_playback_hw; channel->emu = chip; channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_ca0106_pcm_channel_interrupt; + /* printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); */ + /* channel->interrupt = snd_ca0106_pcm_channel_interrupt; */ channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -584,22 +584,22 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_capture_hw; channel->emu = chip; channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_ca0106_pcm_channel_interrupt; + /* printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); */ + /* channel->interrupt = snd_ca0106_pcm_channel_interrupt; */ channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; - //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); + /* snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); */ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err; return 0; @@ -680,37 +680,37 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) u32 reg40_set = 0; u32 reg40; /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask = 0x03030000; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ u32 reg71_set = 0; u32 reg71; int i; - - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + + /* snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); */ + /* snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); */ + /* snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); */ /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ switch (runtime->rate) { case 44100: reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; + reg71_set = 0x01010000; break; case 48000: reg40_set = 0; - reg71_set = 0; + reg71_set = 0; break; case 96000: reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; + reg71_set = 0x02020000; break; case 192000: reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; + reg71_set = 0x03030000; break; default: reg40_set = 0; - reg71_set = 0; + reg71_set = 0; break; } /* Format is a global setting */ @@ -726,7 +726,7 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) hcfg_set = 0; break; } - hcfg = inl(emu->port + HCFG) ; + hcfg = inl(emu->port + HCFG); hcfg = (hcfg & ~hcfg_mask) | hcfg_set; outl(hcfg, emu->port + HCFG); reg40 = snd_ca0106_ptr_read(emu, 0x40, 0); @@ -737,18 +737,18 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) snd_ca0106_ptr_write(emu, 0x71, 0, reg71); /* FIXME: Check emu->buffer.size before actually writing to it. */ - for(i=0; i < runtime->periods; i++) { + for (i = 0; i < runtime->periods; i++) { table_base[i*2] = runtime->dma_addr + (i * period_size_bytes); table_base[i*2+1] = period_size_bytes << 16; } - + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel)); snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19); snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0); snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr); - snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes + snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); /* buffer size in bytes */ /* FIXME test what 0 bytes does. */ - snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes + snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); /* buffer size in bytes */ snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0); snd_ca0106_ptr_write(emu, 0x07, channel, 0x0); snd_ca0106_ptr_write(emu, 0x08, channel, 0); @@ -758,7 +758,7 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | - 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); } #endif @@ -776,31 +776,31 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) u32 hcfg_set = 0x00000000; u32 hcfg; u32 over_sampling=0x2; - u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */ + u32 reg71_mask = 0x0000c000; /* Global. Set ADC rate. */ u32 reg71_set = 0; u32 reg71; - - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + + /* snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); */ + /* snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); */ + /* snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); */ /* reg71 controls ADC rate. */ switch (runtime->rate) { case 44100: reg71_set = 0x00004000; break; case 48000: - reg71_set = 0; + reg71_set = 0; break; case 96000: reg71_set = 0x00008000; over_sampling=0xa; break; case 192000: - reg71_set = 0x0000c000; + reg71_set = 0x0000c000; over_sampling=0xa; break; default: - reg71_set = 0; + reg71_set = 0; break; } /* Format is a global setting */ @@ -816,7 +816,7 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) hcfg_set = 0; break; } - hcfg = inl(emu->port + HCFG) ; + hcfg = inl(emu->port + HCFG); hcfg = (hcfg & ~hcfg_mask) | hcfg_set; outl(hcfg, emu->port + HCFG); reg71 = snd_ca0106_ptr_read(emu, 0x71, 0); @@ -827,10 +827,10 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) } - //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + /* printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); */ snd_ca0106_ptr_write(emu, 0x13, channel, 0); snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); - snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes + snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); /* buffer size in bytes */ snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0); return 0; @@ -866,13 +866,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; - //snd_printk("channel=%d\n",channel); + /* snd_printk("channel=%d\n",channel); */ epcm->running = running; basic |= (0x1<<channel); extended |= (0x10<<channel); snd_pcm_trigger_done(s, substream); } - //snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); + /* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -940,7 +940,7 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + /* printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); */ return ptr; } @@ -963,7 +963,7 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + /* printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); */ return ptr; } @@ -1030,9 +1030,9 @@ static struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_ca0106_pcm_hw_params_playback, .hw_free = snd_ca0106_pcm_hw_free_playback, - .prepare = snd_ca0106_pcm_prepare_playback, - .trigger = snd_ca0106_pcm_trigger_playback, - .pointer = snd_ca0106_pcm_pointer_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, }; static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = { @@ -1041,9 +1041,9 @@ static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_ca0106_pcm_hw_params_playback, .hw_free = snd_ca0106_pcm_hw_free_playback, - .prepare = snd_ca0106_pcm_prepare_playback, - .trigger = snd_ca0106_pcm_trigger_playback, - .pointer = snd_ca0106_pcm_pointer_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, }; static struct snd_pcm_ops snd_ca0106_playback_rear_ops = { @@ -1052,9 +1052,9 @@ static struct snd_pcm_ops snd_ca0106_playback_rear_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_ca0106_pcm_hw_params_playback, .hw_free = snd_ca0106_pcm_hw_free_playback, - .prepare = snd_ca0106_pcm_prepare_playback, - .trigger = snd_ca0106_pcm_trigger_playback, - .pointer = snd_ca0106_pcm_pointer_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, }; @@ -1077,7 +1077,7 @@ static void snd_ca0106_ac97_write(struct snd_ac97 *ac97, { struct snd_ca0106 *emu = ac97->private_data; unsigned long flags; - + spin_lock_irqsave(&emu->emu_lock, flags); outb(reg, emu->port + AC97ADDRESS); outw(val, emu->port + AC97DATA); @@ -1093,7 +1093,7 @@ static int snd_ca0106_ac97(struct snd_ca0106 *chip) .write = snd_ca0106_ac97_write, .read = snd_ca0106_ac97_read, }; - + if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) return err; pbus->no_vra = 1; /* we don't need VRA */ @@ -1107,13 +1107,13 @@ static int snd_ca0106_ac97(struct snd_ca0106 *chip) static int snd_ca0106_free(struct snd_ca0106 *chip) { if (chip->res_port != NULL) { /* avoid access to already used hardware */ - // disable interrupts + /* disable interrupts */ snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0); outl(0, chip->port + INTE); snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0); udelay(1000); - // disable audio - //outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); + /* disable audio */ + /* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */ outl(0, chip->port + HCFG); /* FIXME: We need to stop and DMA transfers here. * But as I am not sure how yet, we cannot from the dma pages. @@ -1122,13 +1122,13 @@ static int snd_ca0106_free(struct snd_ca0106 *chip) } if (chip->irq >= 0) free_irq(chip->irq, chip); - // release the data + /* release the data */ #if 1 if (chip->buffer.area) snd_dma_free_pages(&chip->buffer); #endif - // release the i/o port + /* release the i/o port */ release_and_free_resource(chip->res_port); pci_disable_device(chip->pci); @@ -1153,38 +1153,38 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) struct snd_ca0106_channel *pchannel; status = inl(chip->port + IPR); - if (! status) + if (!status) return IRQ_NONE; stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); - //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); - //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + /* snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); */ + /* snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); */ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ - for(i = 0; i < 4; i++) { + for (i = 0; i < 4; i++) { pchannel = &(chip->playback_channels[i]); if (stat76 & mask) { /* FIXME: Select the correct substream for period elapsed */ - if(pchannel->use) { + if (pchannel->use) { snd_pcm_period_elapsed(pchannel->epcm->substream); - //printk(KERN_INFO "interrupt [%d] used\n", i); + /* printk(KERN_INFO "interrupt [%d] used\n", i); */ } } - //printk(KERN_INFO "channel=%p\n",pchannel); - //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); + /* printk(KERN_INFO "channel=%p\n",pchannel); */ + /* printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); */ mask <<= 1; } mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */ - for(i = 0; i < 4; i++) { + for (i = 0; i < 4; i++) { pchannel = &(chip->capture_channels[i]); if (stat76 & mask) { /* FIXME: Select the correct substream for period elapsed */ - if(pchannel->use) { + if (pchannel->use) { snd_pcm_period_elapsed(pchannel->epcm->substream); - //printk(KERN_INFO "interrupt [%d] used\n", i); + /* printk(KERN_INFO "interrupt [%d] used\n", i); */ } } - //printk(KERN_INFO "channel=%p\n",pchannel); - //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); + /* printk(KERN_INFO "channel=%p\n",pchannel); */ + /* printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); */ mask <<= 1; } @@ -1198,7 +1198,7 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) chip->midi.interrupt_disable(&chip->midi, chip->midi.tx_enable | chip->midi.rx_enable); } - // acknowledge the interrupt if necessary + /* acknowledge the interrupt if necessary */ outl(status, chip->port+IPR); return IRQ_HANDLED; @@ -1209,12 +1209,12 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s struct snd_pcm *pcm; struct snd_pcm_substream *substream; int err; - + if (rpcm) *rpcm = NULL; if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0) return err; - + pcm->private_data = emu; switch (device) { @@ -1241,29 +1241,29 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s strcpy(pcm->name, "CA0106"); emu->pcm = pcm; - for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - substream; + for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream; substream = substream->next) { - if ((err = snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */ return err; } - for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - substream; + for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream; substream = substream->next) { - if ((err = snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), 64*1024, 64*1024)) < 0) return err; } - + if (rpcm) *rpcm = pcm; - + return 0; } @@ -1313,9 +1313,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_ca0106_dev_free, }; - + *rchip = NULL; - + if ((err = pci_enable_device(pci)) < 0) return err; if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || @@ -1324,22 +1324,22 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_disable_device(pci); return -ENXIO; } - + chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) { pci_disable_device(pci); return -ENOMEM; } - + chip->card = card; chip->pci = pci; chip->irq = -1; spin_lock_init(&chip->emu_lock); - + chip->port = pci_resource_start(pci, 0); if ((chip->res_port = request_region(chip->port, 0x20, - "snd_ca0106")) == NULL) { + "snd_ca0106")) == NULL) { snd_ca0106_free(chip); printk(KERN_ERR "cannot allocate the port\n"); return -EBUSY; @@ -1352,9 +1352,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - - /* This stores the periods table. */ - if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { + + /* This stores the periods table. */ + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { snd_ca0106_free(chip); return -ENOMEM; } @@ -1441,13 +1441,13 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006); #endif - //snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */ + /* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); OSS drivers set this. */ /* Analog or Digital output */ snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf); snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */ chip->spdif_enable = 0; /* Set digital SPDIF output off */ - //snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */ - //snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */ + /* snd_ca0106_ptr_write(chip, 0x45, 0, 0); Analogue out */ + /* snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); Digital out */ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */ @@ -1457,11 +1457,11 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676); snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410); snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676); - for(ch = 0; ch < 4; ch++) { + for (ch = 0; ch < 4; ch++) { snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030); - //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */ - //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */ + /* snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); Mute */ + /* snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); Mute */ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */ } @@ -1485,30 +1485,30 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ outl(0x0, chip->port+GPIO); - //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + /* outl(0x00f0e000, chip->port+GPIO); Analog */ outl(0x005f5301, chip->port+GPIO); /* Analog */ } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ outl(0x0, chip->port+GPIO); - //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + /* outl(0x00f0e000, chip->port+GPIO); Analog */ outl(0x005f5301, chip->port+GPIO); /* Analog */ } else { outl(0x0, chip->port+GPIO); outl(0x005f03a3, chip->port+GPIO); /* Analog */ - //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */ + /* outl(0x005f02a2, chip->port+GPIO); SPDIF */ } snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */ - //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); - //outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ - //outl(0x00000009, chip->port+HCFG); + /* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */ + /* outl(0x00001409, chip->port+HCFG); 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ + /* outl(0x00000009, chip->port+HCFG); */ outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */ if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ int size, n; size = ARRAY_SIZE(i2c_adc_init); - //snd_printk("I2C:array size=0x%x\n", size); + /* snd_printk("I2C:array size=0x%x\n", size); */ for (n=0; n < size; n++) { snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); } @@ -1517,7 +1517,7 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, chip->i2c_capture_volume[n][1]= 0xcf; } chip->i2c_capture_source=2; /* Line in */ - //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ + /* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); Enable Line-in capture. MIC in currently untested. */ } if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */ int size, n; @@ -1616,7 +1616,7 @@ static int __devinit snd_ca0106_midi(struct snd_ca0106 *chip, unsigned int chann midi->get_dev_id_port = ca0106_dev_id_port; midi->dev_id = chip; - + if ((err = ca_midi_init(chip, midi, 0, name)) < 0) return err; diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 3025ed1..9bfc92d 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -5,7 +5,7 @@ * * FEATURES currently supported: * See ca0106_main.c for features. - * + * * Changelog: * Support interrupts per period. * Removed noise from Center/LFE channel when in Analog mode. @@ -71,7 +71,7 @@ #include <sound/ac97_codec.h> #include <sound/info.h> #include <sound/tlv.h> -#include <asm/io.h> +#include <linux/io.h> #include "ca0106.h" @@ -157,7 +157,7 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, u32 mask; u32 source; - val = ucontrol->value.enumerated.item[0] ; + val = ucontrol->value.enumerated.item[0]; if (val >= 6) return -EINVAL; change = (emu->capture_source != val); @@ -207,7 +207,7 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, * update the capture volume from the cached value * for the particular source. */ - source_id = ucontrol->value.enumerated.item[0] ; + source_id = ucontrol->value.enumerated.item[0]; if (source_id >= 4) return -EINVAL; change = (emu->i2c_capture_source != source_id); @@ -273,23 +273,23 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, int change = 0; u32 tmp; - val = ucontrol->value.enumerated.item[0] ; + val = ucontrol->value.enumerated.item[0]; if (val > 1) return -EINVAL; change = (emu->capture_mic_line_in != val); if (change) { emu->capture_mic_line_in = val; if (val) { - //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ + /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); Mute input */ tmp = inl(emu->port+GPIO) & ~0x400; tmp = tmp | 0x400; outl(tmp, emu->port+GPIO); - //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); + /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */ } else { - //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ + /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); Mute input */ tmp = inl(emu->port+GPIO) & ~0x400; outl(tmp, emu->port+GPIO); - //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); + /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */ } } return change; @@ -405,7 +405,7 @@ static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol, nval = ((0xff - ucontrol->value.integer.value[0]) << 24) | ((0xff - ucontrol->value.integer.value[1]) << 16); nval |= ((0xff - ucontrol->value.integer.value[0]) << 8) | - ((0xff - ucontrol->value.integer.value[1]) ); + ((0xff - ucontrol->value.integer.value[1])); if (oval == nval) return 0; snd_ca0106_ptr_write(emu, reg, channel_id, nval); @@ -451,7 +451,7 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol, return -EINVAL; if (ogain != ngain) { if (emu->i2c_capture_source == source_id) - snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) ); + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff)); emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0]; change = 1; } diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d1..16789b4 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -5,7 +5,7 @@ * * FEATURES currently supported: * See ca0106_main.c for features. - * + * * Changelog: * Support interrupts per period. * Removed noise from Center/LFE channel when in Analog mode. @@ -71,7 +71,7 @@ #include <sound/ac97_codec.h> #include <sound/info.h> #include <sound/asoundef.h> -#include <asm/io.h> +#include <linux/io.h> #include "ca0106.h" @@ -98,7 +98,7 @@ static struct snd_ca0106_category_str snd_ca0106_con_category[] = { }; -static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 value) +static void snd_ca0106_proc_dump_iec958(struct snd_info_buffer *buffer, u32 value) { int i; u32 status[4]; @@ -106,8 +106,8 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val status[1] = (value >> 8) & 0xff; status[2] = (value >> 16) & 0xff; status[3] = (value >> 24) & 0xff; - - if (! (status[0] & IEC958_AES0_PROFESSIONAL)) { + + if (!(status[0] & IEC958_AES0_PROFESSIONAL)) { /* consumer */ snd_iprintf(buffer, "Mode: consumer\n"); snd_iprintf(buffer, "Data: "); @@ -272,7 +272,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val } } -static void snd_ca0106_proc_iec958(struct snd_info_entry *entry, +static void snd_ca0106_proc_iec958(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -282,9 +282,9 @@ static void snd_ca0106_proc_iec958(struct snd_info_entry *entry, snd_iprintf(buffer, "Status: %s, %s, %s\n", (value & 0x100000) ? "Rate Locked" : "Not Rate Locked", (value & 0x200000) ? "SPDIF Locked" : "No SPDIF Lock", - (value & 0x400000) ? "Audio Valid" : "No valid audio" ); - snd_iprintf(buffer, "Estimated sample rate: %u\n", - ((value & 0xfffff) * 48000) / 0x8000 ); + (value & 0x400000) ? "Audio Valid" : "No valid audio"); + snd_iprintf(buffer, "Estimated sample rate: %u\n", + ((value & 0xfffff) * 48000) / 0x8000); if (value & 0x200000) { snd_iprintf(buffer, "IEC958/SPDIF input status:\n"); value = snd_ca0106_ptr_read(emu, SPDIF_INPUT_STATUS, 0); @@ -294,7 +294,7 @@ static void snd_ca0106_proc_iec958(struct snd_info_entry *entry, snd_iprintf(buffer, "\n"); } -static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", &reg, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff)) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -312,7 +312,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, } } -static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -320,7 +320,7 @@ static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry, unsigned long flags; int i; snd_iprintf(buffer, "Registers:\n\n"); - for(i = 0; i < 0x20; i+=4) { + for (i = 0; i < 0x20; i += 4) { spin_lock_irqsave(&emu->emu_lock, flags); value = inl(emu->port + i); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -328,7 +328,7 @@ static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry, } } -static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -336,7 +336,7 @@ static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry, unsigned long flags; int i; snd_iprintf(buffer, "Registers:\n\n"); - for(i = 0; i < 0x20; i+=2) { + for (i = 0; i < 0x20; i += 2) { spin_lock_irqsave(&emu->emu_lock, flags); value = inw(emu->port + i); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -344,7 +344,7 @@ static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry, } } -static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -352,7 +352,7 @@ static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry, unsigned long flags; int i; snd_iprintf(buffer, "Registers:\n\n"); - for(i = 0; i < 0x20; i+=1) { + for (i = 0; i < 0x20; i++) { spin_lock_irqsave(&emu->emu_lock, flags); value = inb(emu->port + i); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -360,7 +360,7 @@ static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry, } } -static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -368,7 +368,7 @@ static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry, int i,j; snd_iprintf(buffer, "Registers\n"); - for(i = 0; i < 0x40; i++) { + for (i = 0; i < 0x40; i++) { snd_iprintf(buffer, "%02X: ",i); for (j = 0; j < 4; j++) { value = snd_ca0106_ptr_read(emu, i, j); @@ -378,7 +378,7 @@ static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry, } } -static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -386,7 +386,7 @@ static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry, int i,j; snd_iprintf(buffer, "Registers\n"); - for(i = 0x40; i < 0x80; i++) { + for (i = 0x40; i < 0x80; i++) { snd_iprintf(buffer, "%02X: ",i); for (j = 0; j < 4; j++) { value = snd_ca0106_ptr_read(emu, i, j); @@ -396,7 +396,7 @@ static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry, } } -static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, +static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -405,12 +405,12 @@ static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) ) + if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3)) snd_ca0106_ptr_write(emu, reg, channel_id, val); } } -static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry, +static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ca0106 *emu = entry->private_data; @@ -425,32 +425,32 @@ static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry, } } -int __devinit snd_ca0106_proc_init(struct snd_ca0106 * emu) +int __devinit snd_ca0106_proc_init(struct snd_ca0106 *emu) { struct snd_info_entry *entry; - - if(! snd_card_proc_new(emu->card, "iec958", &entry)) + + if (!snd_card_proc_new(emu->card, "iec958", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958); - if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { + if (!snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; entry->mode |= S_IWUSR; } - if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) + if (!snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); - if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry)) + if (!snd_card_proc_new(emu->card, "ca0106_reg8", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8); - if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { + if (!snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; entry->mode |= S_IWUSR; } - if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { + if (!snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { entry->c.text.write = snd_ca0106_proc_i2c_write; entry->private_data = emu; entry->mode |= S_IWUSR; } - if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) + if (!snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); return 0; } diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c index c788511..3056f98 100644 --- a/sound/pci/ca0106/ca_midi.c +++ b/sound/pci/ca0106/ca_midi.c @@ -1,4 +1,4 @@ -/* +/* * Copyright 10/16/2005 Tilman Kranz <tilde(a)tk-sls.de> * Creative Audio MIDI, for the CA0106 Driver * Version: 0.0.1 @@ -66,7 +66,7 @@ static void ca_midi_interrupt(struct snd_ca_midi *midi, unsigned int status) ca_midi_clear_rx(midi); } else { byte = ca_midi_read_data(midi); - if(midi->substream_input) + if (midi->substream_input) snd_rawmidi_receive(midi->substream_input, &byte, 1); @@ -124,7 +124,7 @@ static int ca_midi_input_open(struct snd_rawmidi_substream *substream) { struct snd_ca_midi *midi = substream->rmidi->private_data; unsigned long flags; - + if (snd_BUG_ON(!midi->dev_id)) return -ENXIO; spin_lock_irqsave(&midi->open_lock, flags); @@ -187,13 +187,13 @@ static int ca_midi_output_close(struct snd_rawmidi_substream *substream) if (snd_BUG_ON(!midi->dev_id)) return -ENXIO; - + spin_lock_irqsave(&midi->open_lock, flags); midi->interrupt_disable(midi,midi->tx_enable); midi->midi_mode &= ~CA_MIDI_MODE_OUTPUT; midi->substream_output = NULL; - + if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) { spin_unlock_irqrestore(&midi->open_lock, flags); ca_midi_cmd(midi, midi->reset, 0); @@ -230,7 +230,7 @@ static void ca_midi_output_trigger(struct snd_rawmidi_substream *substream, int unsigned char byte; spin_lock_irqsave(&midi->output_lock, flags); - + /* try to send some amount of bytes here before interrupts */ while (max > 0) { if (ca_midi_output_ready(midi)) { @@ -309,7 +309,7 @@ int __devinit ca_midi_init(void *dev_id, struct snd_ca_midi *midi, int device, c SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = midi; rmidi->private_free = ca_rmidi_free; - + midi->rmidi = rmidi; return 0; } diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h index 922ed3e..d285456 100644 --- a/sound/pci/ca0106/ca_midi.h +++ b/sound/pci/ca0106/ca_midi.h @@ -1,4 +1,4 @@ -/* +/* * Copyright 10/16/2005 Tilman Kranz <tilde(a)tk-sls.de> * Creative Audio MIDI, for the CA0106 Driver * Version: 0.0.1 @@ -46,8 +46,8 @@ struct snd_ca_midi { unsigned int midi_mode; int port; int tx_enable, rx_enable; - int ipr_tx, ipr_rx; - + int ipr_tx, ipr_rx; + int input_avail, output_ready; int ack, reset, enter_uart;

1 0

[alsa-devel] [PATCH 03/24] sound/pci: coding style fixes: sound/pci/a*
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ad1889.c | 116 +++++++++++++++++++++--------------------- sound/pci/ak4531_codec.c | 14 +++--- sound/pci/als300.c | 16 +++--- sound/pci/als4000.c | 70 +++++++++++++------------- sound/pci/atiixp.c | 64 ++++++++++++------------ sound/pci/atiixp_modem.c | 46 +++++++++--------- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/aw2/aw2-saa7146.c | 2 +- sound/pci/azt3328.c | 12 ++-- sound/pci/azt3328.h | 6 +- 10 files changed, 174 insertions(+), 174 deletions(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a97..5fdca0e 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -45,7 +45,7 @@ #include <sound/initval.h> #include <sound/ac97_codec.h> -#include <asm/io.h> +#include <linux/io.h> #include "ad1889.h" #include "ac97/ac97_id.h" @@ -137,7 +137,7 @@ static inline void ad1889_unmute(struct snd_ad1889 *chip) { u16 st; - st = ad1889_readw(chip, AD_DS_WADA) & + st = ad1889_readw(chip, AD_DS_WADA) & ~(AD_DS_WADA_RWAM | AD_DS_WADA_LWAM); ad1889_writew(chip, AD_DS_WADA, st); ad1889_readw(chip, AD_DS_WADA); @@ -198,13 +198,13 @@ static void ad1889_channel_reset(struct snd_ad1889 *chip, unsigned int channel) { u16 reg; - + if (channel & AD_CHAN_WAV) { /* Disable wave channel */ reg = ad1889_readw(chip, AD_DS_WSMC) & ~AD_DS_WSMC_WAEN; ad1889_writew(chip, AD_DS_WSMC, reg); chip->wave.reg = reg; - + /* disable IRQs */ reg = ad1889_readw(chip, AD_DMA_WAV); reg &= AD_DMA_IM_DIS; @@ -219,7 +219,7 @@ ad1889_channel_reset(struct snd_ad1889 *chip, unsigned int channel) /* flush */ ad1889_readw(chip, AD_DMA_WAV); } - + if (channel & AD_CHAN_ADC) { /* Disable ADC channel */ reg = ad1889_readw(chip, AD_DS_RAMC) & ~AD_DS_RAMC_ADEN; @@ -230,7 +230,7 @@ ad1889_channel_reset(struct snd_ad1889 *chip, unsigned int channel) reg &= AD_DMA_IM_DIS; reg &= ~AD_DMA_LOOP; ad1889_writew(chip, AD_DMA_ADC, reg); - + ad1889_load_adc_buffer_address(chip, 0x0); ad1889_load_adc_buffer_count(chip, 0x0); ad1889_load_adc_interrupt_count(chip, 0x0); @@ -258,8 +258,8 @@ static int snd_ad1889_ac97_ready(struct snd_ad1889 *chip) { int retry = 400; /* average needs 352 msec */ - - while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY) + + while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY) && --retry) mdelay(1); if (!retry) { @@ -272,11 +272,11 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip) return 0; } -static int +static int snd_ad1889_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(substream, + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -372,7 +372,7 @@ snd_ad1889_playback_prepare(struct snd_pcm_substream *ss) ad1889_channel_reset(chip, AD_CHAN_WAV); reg = ad1889_readw(chip, AD_DS_WSMC); - + /* Mask out 16-bit / Stereo */ reg &= ~(AD_DS_WSMC_WA16 | AD_DS_WSMC_WAST); @@ -384,13 +384,13 @@ snd_ad1889_playback_prepare(struct snd_pcm_substream *ss) /* let's make sure we don't clobber ourselves */ spin_lock_irq(&chip->lock); - + chip->wave.size = size; chip->wave.reg = reg; chip->wave.addr = rt->dma_addr; ad1889_writew(chip, AD_DS_WSMC, chip->wave.reg); - + /* Set sample rates on the codec */ ad1889_writew(chip, AD_DS_WAS, rt->rate); @@ -401,9 +401,9 @@ snd_ad1889_playback_prepare(struct snd_pcm_substream *ss) /* writes flush */ ad1889_readw(chip, AD_DS_WSMC); - + spin_unlock_irq(&chip->lock); - + ad1889_debug("prepare playback: addr = 0x%x, count = %u, " "size = %u, reg = 0x%x, rate = %u\n", chip->wave.addr, count, size, reg, rt->rate); @@ -420,7 +420,7 @@ snd_ad1889_capture_prepare(struct snd_pcm_substream *ss) u16 reg; ad1889_channel_reset(chip, AD_CHAN_ADC); - + reg = ad1889_readw(chip, AD_DS_RAMC); /* Mask out 16-bit / Stereo */ @@ -434,7 +434,7 @@ snd_ad1889_capture_prepare(struct snd_pcm_substream *ss) /* let's make sure we don't clobber ourselves */ spin_lock_irq(&chip->lock); - + chip->ramc.size = size; chip->ramc.reg = reg; chip->ramc.addr = rt->dma_addr; @@ -448,9 +448,9 @@ snd_ad1889_capture_prepare(struct snd_pcm_substream *ss) /* writes flush */ ad1889_readw(chip, AD_DS_RAMC); - + spin_unlock_irq(&chip->lock); - + ad1889_debug("prepare capture: addr = 0x%x, count = %u, " "size = %u, reg = 0x%x, rate = %u\n", chip->ramc.addr, count, size, reg, rt->rate); @@ -466,7 +466,7 @@ snd_ad1889_playback_trigger(struct snd_pcm_substream *ss, int cmd) { u16 wsmc; struct snd_ad1889 *chip = snd_pcm_substream_chip(ss); - + wsmc = ad1889_readw(chip, AD_DS_WSMC); switch (cmd) { @@ -486,9 +486,9 @@ snd_ad1889_playback_trigger(struct snd_pcm_substream *ss, int cmd) snd_BUG(); return -EINVAL; } - + chip->wave.reg = wsmc; - ad1889_writew(chip, AD_DS_WSMC, wsmc); + ad1889_writew(chip, AD_DS_WSMC, wsmc); ad1889_readw(chip, AD_DS_WSMC); /* flush */ /* reset the chip when STOP - will disable IRQs */ @@ -509,7 +509,7 @@ snd_ad1889_capture_trigger(struct snd_pcm_substream *ss, int cmd) struct snd_ad1889 *chip = snd_pcm_substream_chip(ss); ramc = ad1889_readw(chip, AD_DS_RAMC); - + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* enable DMA loop & interrupts */ @@ -524,15 +524,15 @@ snd_ad1889_capture_trigger(struct snd_pcm_substream *ss, int cmd) default: return -EINVAL; } - + chip->ramc.reg = ramc; - ad1889_writew(chip, AD_DS_RAMC, ramc); + ad1889_writew(chip, AD_DS_RAMC, ramc); ad1889_readw(chip, AD_DS_RAMC); /* flush */ - + /* reset the chip when STOP - will disable IRQs */ if (cmd == SNDRV_PCM_TRIGGER_STOP) ad1889_channel_reset(chip, AD_CHAN_ADC); - + return 0; } @@ -548,10 +548,10 @@ snd_ad1889_playback_pointer(struct snd_pcm_substream *ss) ptr = ad1889_readl(chip, AD_DMA_WAVCA); ptr -= chip->wave.addr; - + if (snd_BUG_ON(ptr >= chip->wave.size)) return 0; - + return bytes_to_frames(ss->runtime, ptr); } @@ -570,7 +570,7 @@ snd_ad1889_capture_pointer(struct snd_pcm_substream *ss) if (snd_BUG_ON(ptr >= chip->ramc.size)) return 0; - + return bytes_to_frames(ss->runtime, ptr); } @@ -582,7 +582,7 @@ static struct snd_pcm_ops snd_ad1889_playback_ops = { .hw_free = snd_ad1889_hw_free, .prepare = snd_ad1889_playback_prepare, .trigger = snd_ad1889_playback_trigger, - .pointer = snd_ad1889_playback_pointer, + .pointer = snd_ad1889_playback_pointer, }; static struct snd_pcm_ops snd_ad1889_capture_ops = { @@ -593,7 +593,7 @@ static struct snd_pcm_ops snd_ad1889_capture_ops = { .hw_free = snd_ad1889_hw_free, .prepare = snd_ad1889_capture_prepare, .trigger = snd_ad1889_capture_trigger, - .pointer = snd_ad1889_capture_pointer, + .pointer = snd_ad1889_capture_pointer, }; static irqreturn_t @@ -636,7 +636,7 @@ snd_ad1889_pcm_init(struct snd_ad1889 *chip, int device, struct snd_pcm **rpcm) if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ad1889_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ad1889_capture_ops); @@ -644,7 +644,7 @@ snd_ad1889_pcm_init(struct snd_ad1889 *chip, int device, struct snd_pcm **rpcm) pcm->private_data = chip; pcm->info_flags = 0; strcpy(pcm->name, chip->card->shortname); - + chip->pcm = pcm; chip->psubs = NULL; chip->csubs = NULL; @@ -658,10 +658,10 @@ snd_ad1889_pcm_init(struct snd_ad1889 *chip, int device, struct snd_pcm **rpcm) snd_printk(KERN_ERR PFX "buffer allocation error: %d\n", err); return err; } - + if (rpcm) *rpcm = pcm; - + return 0; } @@ -679,24 +679,24 @@ snd_ad1889_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffe (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono"); snd_iprintf(buffer, "Wave Quality: %d-bit linear\n", (reg & AD_DS_WSMC_WA16) ? 16 : 8); - + /* WARQ is at offset 12 */ tmp = (reg & AD_DS_WSMC_WARQ) ? (((reg & AD_DS_WSMC_WARQ >> 12) & 0x01) ? 12 : 18) : 4; tmp /= (reg & AD_DS_WSMC_WAST) ? 2 : 1; - + snd_iprintf(buffer, "Wave FIFO: %d %s words\n\n", tmp, (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono"); - - + + snd_iprintf(buffer, "Synthesis output: %s\n", reg & AD_DS_WSMC_SYEN ? "enabled" : "disabled"); - + /* SYRQ is at offset 4 */ tmp = (reg & AD_DS_WSMC_SYRQ) ? (((reg & AD_DS_WSMC_SYRQ >> 4) & 0x01) ? 12 : 18) : 4; tmp /= (reg & AD_DS_WSMC_WAST) ? 2 : 1; - + snd_iprintf(buffer, "Synthesis FIFO: %d %s words\n\n", tmp, (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono"); @@ -707,27 +707,27 @@ snd_ad1889_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffe (reg & AD_DS_RAMC_ADST) ? "stereo" : "mono"); snd_iprintf(buffer, "ADC Quality: %d-bit linear\n", (reg & AD_DS_RAMC_AD16) ? 16 : 8); - + /* ACRQ is at offset 4 */ tmp = (reg & AD_DS_RAMC_ACRQ) ? (((reg & AD_DS_RAMC_ACRQ >> 4) & 0x01) ? 12 : 18) : 4; tmp /= (reg & AD_DS_RAMC_ADST) ? 2 : 1; - + snd_iprintf(buffer, "ADC FIFO: %d %s words\n\n", tmp, (reg & AD_DS_RAMC_ADST) ? "stereo" : "mono"); - + snd_iprintf(buffer, "Resampler input: %s\n", reg & AD_DS_RAMC_REEN ? "enabled" : "disabled"); - + /* RERQ is at offset 12 */ tmp = (reg & AD_DS_RAMC_RERQ) ? (((reg & AD_DS_RAMC_RERQ >> 12) & 0x01) ? 12 : 18) : 4; tmp /= (reg & AD_DS_RAMC_ADST) ? 2 : 1; - + snd_iprintf(buffer, "Resampler FIFO: %d %s words\n\n", tmp, (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono"); - - + + /* doc says LSB represents -1.5dB, but the max value (-94.5dB) suggests that LSB is -3dB, which is more coherent with the logarithmic nature of the dB scale */ @@ -739,7 +739,7 @@ snd_ad1889_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffe snd_iprintf(buffer, "Right: %s, -%d dB\n", (reg & AD_DS_WADA_RWAM) ? "mute" : "unmute", ((reg & AD_DS_WADA_RWAA) >> 8) * 3); - + reg = ad1889_readw(chip, AD_DS_WAS); snd_iprintf(buffer, "Wave samplerate: %u Hz\n", reg); reg = ad1889_readw(chip, AD_DS_RES); @@ -779,7 +779,7 @@ snd_ad1889_ac97_xinit(struct snd_ad1889 *chip) /* Interface Enable */ reg |= AD_AC97_ACIC_ACIE; ad1889_writew(chip, AD_AC97_ACIC, reg); - + snd_ad1889_ac97_ready(chip); /* Audio Stream Output | Variable Sample Rate Mode */ @@ -820,7 +820,7 @@ snd_ad1889_ac97_init(struct snd_ad1889 *chip, const char *quirk_override) err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus); if (err < 0) return err; - + chip->ac97_bus->private_free = snd_ad1889_ac97_bus_free; memset(&ac97, 0, sizeof(ac97)); @@ -831,9 +831,9 @@ snd_ad1889_ac97_init(struct snd_ad1889 *chip, const char *quirk_override) err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97); if (err < 0) return err; - + snd_ac97_tune_hardware(chip->ac97, ac97_quirks, quirk_override); - + return 0; } @@ -871,14 +871,14 @@ skip_hw: } static int -snd_ad1889_dev_free(struct snd_device *device) +snd_ad1889_dev_free(struct snd_device *device) { struct snd_ad1889 *chip = device->device_data; return snd_ad1889_free(chip); } static int __devinit -snd_ad1889_init(struct snd_ad1889 *chip) +snd_ad1889_init(struct snd_ad1889 *chip) { ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */ ad1889_readw(chip, AD_DS_CCS); /* flush posted write */ @@ -938,7 +938,7 @@ snd_ad1889_create(struct snd_card *card, err = -EBUSY; goto free_and_ret; } - + pci_set_master(pci); spin_lock_init(&chip->lock); /* only now can we call ad1889_free */ @@ -1017,7 +1017,7 @@ snd_ad1889_probe(struct pci_dev *pci, err = snd_ad1889_ac97_init(chip, ac97_quirk[devno]); if (err < 0) goto free_and_ret; - + err = snd_ad1889_pcm_init(chip, 0, NULL); if (err < 0) goto free_and_ret; diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 0f819dd..0275686 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -43,13 +43,13 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453 /* * */ - + #if 0 static void snd_ak4531_dump(struct snd_ak4531 *ak4531) { int idx; - + for (idx = 0; idx < 0x19; idx++) printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]); } @@ -84,7 +84,7 @@ static int snd_ak4531_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_ uinfo->value.integer.max = mask; return 0; } - + static int snd_ak4531_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); @@ -151,7 +151,7 @@ static int snd_ak4531_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_ uinfo->value.integer.max = mask; return 0; } - + static int snd_ak4531_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); @@ -226,7 +226,7 @@ static int snd_ak4531_info_input_sw(struct snd_kcontrol *kcontrol, struct snd_ct uinfo->value.integer.max = 1; return 0; } - + static int snd_ak4531_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); @@ -469,7 +469,7 @@ void snd_ak4531_resume(struct snd_ak4531 *ak4531) * /proc interface */ -static void snd_ak4531_proc_read(struct snd_info_entry *entry, +static void snd_ak4531_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ak4531 *ak4531 = entry->private_data; @@ -486,7 +486,7 @@ snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) { struct snd_info_entry *entry; - if (! snd_card_proc_new(card, "ak4531", &entry)) + if (!snd_card_proc_new(card, "ak4531", &entry)) snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); } #endif diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8df6824..9799051 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -38,7 +38,7 @@ #include <linux/interrupt.h> #include <linux/slab.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/control.h> @@ -104,7 +104,7 @@ #define snd_als300_dbgplay(format, args...) printk(KERN_ERR format, ##args) #else #define snd_als300_dbgplay(format, args...) -#endif +#endif enum {DEVICE_ALS300, DEVICE_ALS300_PLUS}; @@ -136,7 +136,7 @@ struct snd_als300 { int chip_type; /* ALS300 or ALS300+ */ - char revision; + char revision; }; struct snd_als300_substream_data { @@ -239,7 +239,7 @@ static irqreturn_t snd_als300plus_interrupt(int irq, void *dev_id) u8 general, mpu, dram; struct snd_als300 *chip = dev_id; struct snd_als300_substream_data *data; - + general = inb(chip->port+ALS300P_IRQ_STATUS); mpu = inb(chip->port+MPU_IRQ_STATUS); dram = inb(chip->port+ALS300P_DRAM_IRQ_STATUS); @@ -461,7 +461,7 @@ static int snd_als300_playback_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; unsigned short period_bytes = snd_pcm_lib_period_bytes(substream); unsigned short buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - + snd_als300_dbgcallenter(); spin_lock_irq(&chip->reg_lock); tmp = snd_als300_gcr_read(chip->port, PLAYBACK_CONTROL); @@ -469,7 +469,7 @@ static int snd_als300_playback_prepare(struct snd_pcm_substream *substream) snd_als300_dbgplay("Period bytes: %d Buffer bytes %d\n", period_bytes, buffer_bytes); - + /* set block size */ tmp &= 0xffff0000; tmp |= period_bytes - 1; @@ -571,7 +571,7 @@ static snd_pcm_uframes_t snd_als300_pointer(struct snd_pcm_substream *substream) data = substream->runtime->private_data; period_bytes = snd_pcm_lib_period_bytes(substream); - + snd_als300_dbgcallenter(); spin_lock(&chip->reg_lock); current_ptr = (u16) snd_als300_gcr_read(chip->port, @@ -641,7 +641,7 @@ static void snd_als300_init(struct snd_als300 *chip) { unsigned long flags; u32 tmp; - + snd_als300_dbgcallenter(); spin_lock_irqsave(&chip->reg_lock, flags); chip->revision = (snd_als300_gcr_read(chip->port, MISC_CONTROL) >> 16) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index ba57005..14d58d8 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -33,15 +33,15 @@ * refer to: ALS4000a.PDF specs Ver 1.0, May 28th, 1998. * * The ALS4000 seems to be the PCI-cousin of the ALS100. It contains an - * ALS100-like SB DSP/mixer, an OPL3 synth, a MPU401 and a gameport - * interface. These subsystems can be mapped into ISA io-port space, - * using the PCI-interface. In addition, the PCI-bit provides DMA and IRQ + * ALS100-like SB DSP/mixer, an OPL3 synth, a MPU401 and a gameport + * interface. These subsystems can be mapped into ISA io-port space, + * using the PCI-interface. In addition, the PCI-bit provides DMA and IRQ * services to the subsystems. - * + * * While ALS4000 is very similar to a SoundBlaster, the differences in * DMA and capturing require more changes to the SoundBlaster than * desirable, so I made this separate driver. - * + * * The ALS4000 can do real full duplex playback/capture. * * FMDAC: @@ -65,7 +65,7 @@ * - power management? (card can do voice wakeup according to datasheet!!) */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/init.h> #include <linux/pci.h> #include <linux/slab.h> @@ -228,7 +228,7 @@ static inline void snd_als4k_gcr_write(struct snd_sb *sb, u32 val) { snd_als4k_gcr_write_addr(sb->alt_port, reg, val); -} +} static inline u32 snd_als4k_gcr_read_addr(unsigned long iobase, enum als4k_gcr_t reg) @@ -337,7 +337,7 @@ static const struct { } playback_cmd_vals[]={ /* ALS4000_FORMAT_U8_MONO */ { SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_UNS_MONO }, -/* ALS4000_FORMAT_S8_MONO */ +/* ALS4000_FORMAT_S8_MONO */ { SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_SIGN_MONO }, /* ALS4000_FORMAT_U16L_MONO */ { SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_UNS_MONO }, @@ -345,7 +345,7 @@ static const struct { { SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_SIGN_MONO }, /* ALS4000_FORMAT_U8_STEREO */ { SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_UNS_STEREO }, -/* ALS4000_FORMAT_S8_STEREO */ +/* ALS4000_FORMAT_S8_STEREO */ { SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_SIGN_STEREO }, /* ALS4000_FORMAT_U16L_STEREO */ { SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_UNS_STEREO }, @@ -359,14 +359,14 @@ enum { CMD_WIDTH8=0x04, CMD_SIGNED=0x10, CMD_MONO=0x80, CMD_STEREO=0xA0 }; static const unsigned char capture_cmd_vals[]= { CMD_WIDTH8|CMD_MONO, /* ALS4000_FORMAT_U8_MONO */ -CMD_WIDTH8|CMD_SIGNED|CMD_MONO, /* ALS4000_FORMAT_S8_MONO */ +CMD_WIDTH8|CMD_SIGNED|CMD_MONO, /* ALS4000_FORMAT_S8_MONO */ CMD_MONO, /* ALS4000_FORMAT_U16L_MONO */ CMD_SIGNED|CMD_MONO, /* ALS4000_FORMAT_S16L_MONO */ CMD_WIDTH8|CMD_STEREO, /* ALS4000_FORMAT_U8_STEREO */ -CMD_WIDTH8|CMD_SIGNED|CMD_STEREO, /* ALS4000_FORMAT_S8_STEREO */ +CMD_WIDTH8|CMD_SIGNED|CMD_STEREO, /* ALS4000_FORMAT_S8_STEREO */ CMD_STEREO, /* ALS4000_FORMAT_U16L_STEREO */ CMD_SIGNED|CMD_STEREO, /* ALS4000_FORMAT_S16L_STEREO */ -}; +}; #define capture_cmd(chip) (capture_cmd_vals[(chip)->capture_format]) static int snd_als4000_hw_params(struct snd_pcm_substream *substream, @@ -389,10 +389,10 @@ static int snd_als4000_capture_prepare(struct snd_pcm_substream *substream) unsigned count; chip->capture_format = snd_als4000_get_format(runtime); - + size = snd_pcm_lib_buffer_bytes(substream); count = snd_pcm_lib_period_bytes(substream); - + if (chip->capture_format & ALS4000_FORMAT_16BIT) count >>= 1; count--; @@ -416,14 +416,14 @@ static int snd_als4000_playback_prepare(struct snd_pcm_substream *substream) unsigned count; chip->playback_format = snd_als4000_get_format(runtime); - + size = snd_pcm_lib_buffer_bytes(substream); count = snd_pcm_lib_period_bytes(substream); - + if (chip->playback_format & ALS4000_FORMAT_16BIT) count >>= 1; count--; - + /* FIXME: from second playback on, there's a lot more clicks and pops * involved here than on first playback. Fiddling with * tons of different settings didn't help (DMA, speaker on/off, @@ -433,16 +433,16 @@ static int snd_als4000_playback_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&chip->reg_lock); snd_als4000_set_rate(chip, runtime->rate); snd_als4000_set_playback_dma(chip, runtime->dma_addr, size); - + /* SPEAKER_ON not needed, since dma_on seems to also enable speaker */ /* snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); */ snd_sbdsp_command(chip, playback_cmd(chip).dsp_cmd); snd_sbdsp_command(chip, playback_cmd(chip).format); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); - snd_sbdsp_command(chip, playback_cmd(chip).dma_off); + snd_sbdsp_command(chip, playback_cmd(chip).dma_off); spin_unlock_irq(&chip->reg_lock); - + return 0; } @@ -450,7 +450,7 @@ static int snd_als4000_capture_trigger(struct snd_pcm_substream *substream, int { struct snd_sb *chip = snd_pcm_substream_chip(substream); int result = 0; - + /* FIXME race condition in here!!! chip->mode non-atomic update gets consistently protected by reg_lock always, _except_ for this place!! @@ -509,11 +509,11 @@ static snd_pcm_uframes_t snd_als4000_capture_pointer(struct snd_pcm_substream *s struct snd_sb *chip = snd_pcm_substream_chip(substream); unsigned int result; - spin_lock(&chip->reg_lock); + spin_lock(&chip->reg_lock); result = snd_als4k_gcr_read(chip, ALS4K_GCRA4_FIFO2_CURRENT_ADDR); spin_unlock(&chip->reg_lock); result &= 0xffff; - return bytes_to_frames( substream->runtime, result ); + return bytes_to_frames(substream->runtime, result); } static snd_pcm_uframes_t snd_als4000_playback_pointer(struct snd_pcm_substream *substream) @@ -521,11 +521,11 @@ static snd_pcm_uframes_t snd_als4000_playback_pointer(struct snd_pcm_substream * struct snd_sb *chip = snd_pcm_substream_chip(substream); unsigned result; - spin_lock(&chip->reg_lock); + spin_lock(&chip->reg_lock); result = snd_als4k_gcr_read(chip, ALS4K_GCRA0_FIFO1_CURRENT_ADDR); spin_unlock(&chip->reg_lock); result &= 0xffff; - return bytes_to_frames( substream->runtime, result ); + return bytes_to_frames(substream->runtime, result); } /* FIXME: this IRQ routine doesn't really support IRQ sharing (we always @@ -563,12 +563,12 @@ static irqreturn_t snd_als4000_interrupt(int irq, void *dev_id) /* ACK the PCI block IRQ */ snd_als4k_iobase_writeb(chip->alt_port, ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU, pci_irqstatus); - + spin_lock(&chip->mixer_lock); /* SPECS_PAGE: 20 */ sb_irqstatus = snd_sbmixer_read(chip, SB_DSP4_IRQSTATUS); spin_unlock(&chip->mixer_lock); - + if (sb_irqstatus & SB_IRQTYPE_8BIT) snd_sb_ack_8bit(chip); if (sb_irqstatus & SB_IRQTYPE_16BIT) @@ -755,7 +755,7 @@ static void snd_als4000_configure(struct snd_sb *chip) snd_als4_cr_write(chip, ALS4K_CR0_SB_CONFIG, tmp & ~ALS4K_CR0_MX80_81_REG_WRITE_ENABLE); spin_unlock_irq(&chip->mixer_lock); - + spin_lock_irq(&chip->reg_lock); /* enable interrupts */ snd_als4k_gcr_write(chip, ALS4K_GCR8C_MISC_CTRL, @@ -764,7 +764,7 @@ static void snd_als4000_configure(struct snd_sb *chip) /* SPECS_PAGE: 39 */ for (i = ALS4K_GCR91_DMA0_ADDR; i <= ALS4K_GCR96_DMA3_MODE_COUNT; ++i) snd_als4k_gcr_write(chip, i, 0); - + snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL, snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL)); spin_unlock_irq(&chip->reg_lock); @@ -836,7 +836,7 @@ static inline int snd_als4000_create_gameport(struct snd_card_als4000 *acard, in static inline void snd_als4000_free_gameport(struct snd_card_als4000 *acard) { } #endif -static void snd_card_als4000_free( struct snd_card *card ) +static void snd_card_als4000_free(struct snd_card *card) { struct snd_card_als4000 *acard = card->private_data; @@ -888,8 +888,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, pci_read_config_word(pci, PCI_COMMAND, &word); pci_write_config_word(pci, PCI_COMMAND, word | PCI_COMMAND_IO); pci_set_master(pci); - - card = snd_card_new(index[dev], id[dev], THIS_MODULE, + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(*acard) /* private_data: acard */); if (card == NULL) { pci_release_regions(pci); @@ -948,7 +948,7 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, } if ((err = snd_sbmixer_new(chip)) < 0) { goto out_err; - } + } if (snd_opl3_create(card, iobase + ALS4K_IOB_10_ADLIB_ADDR0, @@ -975,7 +975,7 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, out_err: snd_card_free(card); - + out: return err; } @@ -994,7 +994,7 @@ static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state) struct snd_sb *chip = acard->chip; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - + snd_pcm_suspend_all(chip->pcm); snd_sbmixer_suspend(chip); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b..3f9a2c5 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -19,7 +19,7 @@ * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -265,7 +265,7 @@ struct atiixp { unsigned long addr; void __iomem *remap_addr; int irq; - + struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97[NUM_ATI_CODECS]; @@ -350,7 +350,7 @@ static int snd_atiixp_update_bits(struct atiixp *chip, unsigned int reg, * list. although we can change the list dynamically, in this version, * a static RING of buffer descriptors is used. * - * the ring is built in this function, and is set up to the hardware. + * the ring is built in this function, and is set up to the hardware. */ static int atiixp_build_dma_packets(struct atiixp *chip, struct atiixp_dma *dma, struct snd_pcm_substream *substream, @@ -430,7 +430,7 @@ static int snd_atiixp_acquire_codec(struct atiixp *chip) int timeout = 1000; while (atiixp_read(chip, PHYS_OUT_ADDR) & ATI_REG_PHYS_OUT_ADDR_EN) { - if (! timeout--) { + if (!timeout--) { snd_printk(KERN_WARNING "atiixp: codec acquire timeout\n"); return -EBUSY; } @@ -471,7 +471,7 @@ static void snd_atiixp_codec_write(struct atiixp *chip, unsigned short codec, unsigned short reg, unsigned short val) { unsigned int data; - + if (snd_atiixp_acquire_codec(chip) < 0) return; data = ((unsigned int)val << ATI_REG_PHYS_OUT_DATA_SHIFT) | @@ -486,7 +486,7 @@ static unsigned short snd_atiixp_ac97_read(struct snd_ac97 *ac97, { struct atiixp *chip = ac97->private_data; return snd_atiixp_codec_read(chip, ac97->num, reg); - + } static void snd_atiixp_ac97_write(struct snd_ac97 *ac97, unsigned short reg, @@ -512,9 +512,9 @@ static int snd_atiixp_aclink_reset(struct atiixp *chip) atiixp_read(chip, CMD); udelay(10); atiixp_update(chip, CMD, ATI_REG_CMD_AC_SOFT_RESET, 0); - + timeout = 10; - while (! (atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) { + while (!(atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) { /* do a hard reset */ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET, ATI_REG_CMD_AC_SYNC); @@ -537,8 +537,8 @@ static int snd_atiixp_aclink_reset(struct atiixp *chip) #ifdef CONFIG_PM static int snd_atiixp_aclink_down(struct atiixp *chip) { - // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */ - // return -EBUSY; + /* if (atiixp_read(chip, MODEM_MIRROR) & 0x1) modem running, too? */ + /* return -EBUSY; */ atiixp_update(chip, CMD, ATI_REG_CMD_POWERDOWN | ATI_REG_CMD_AC_RESET, ATI_REG_CMD_POWERDOWN); @@ -582,7 +582,7 @@ static int __devinit snd_atiixp_codec_detect(struct atiixp *chip) if (ac97_codec == -1) ac97_codec = ac97_probing_bugs(chip->pci); if (ac97_codec >= 0) { - chip->codec_not_ready_bits |= + chip->codec_not_ready_bits |= CODEC_CHECK_BITS ^ (1 << (ac97_codec + 10)); return 0; } @@ -684,7 +684,7 @@ static snd_pcm_uframes_t snd_atiixp_pcm_pointer(struct snd_pcm_substream *substr */ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) { - if (! dma->substream || ! dma->running) + if (!dma->substream || !dma->running) return; snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); @@ -695,7 +695,7 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) */ static void snd_atiixp_update_dma(struct atiixp *chip, struct atiixp_dma *dma) { - if (! dma->substream || ! dma->running) + if (!dma->substream || !dma->running) return; snd_pcm_period_elapsed(dma->substream); } @@ -747,7 +747,7 @@ static int snd_atiixp_pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = -EINVAL; break; } - if (! err) { + if (!err) { snd_atiixp_check_bus_busy(chip); if (cmd == SNDRV_PCM_TRIGGER_STOP) { dma->ops->flush_dma(chip); @@ -805,7 +805,7 @@ static void atiixp_in_enable_transfer(struct atiixp *chip, int on) { if (on) { unsigned int data = atiixp_read(chip, CMD); - if (! (data & ATI_REG_CMD_RECEIVE_EN)) { + if (!(data & ATI_REG_CMD_RECEIVE_EN)) { data |= ATI_REG_CMD_RECEIVE_EN; #if 0 /* FIXME: this causes the endless loop */ /* wait until slot 3/4 are finished */ @@ -852,10 +852,10 @@ static void atiixp_spdif_flush_dma(struct atiixp *chip) /* DMA off, transfer on */ atiixp_spdif_enable_dma(chip, 0); atiixp_spdif_enable_transfer(chip, 1); - + timeout = 100; do { - if (! (atiixp_read(chip, SPDF_DMA_DT_SIZE) & ATI_REG_DMA_FIFO_USED)) + if (!(atiixp_read(chip, SPDF_DMA_DT_SIZE) & ATI_REG_DMA_FIFO_USED)) break; udelay(1); } while (timeout-- > 0); @@ -930,7 +930,7 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) */ atiixp_update(chip, 6CH_REORDER, ATI_REG_6CH_REORDER_EN, substream->runtime->channels >= 6 ? ATI_REG_6CH_REORDER_EN: 0); - + spin_unlock_irq(&chip->reg_lock); return 0; } @@ -1226,7 +1226,7 @@ static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { .enable_transfer = atiixp_out_enable_transfer, .flush_dma = atiixp_out_flush_dma, }; - + static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { .type = ATI_DMA_CAPTURE, .llp_offset = ATI_REG_IN_DMA_LINKPTR, @@ -1235,7 +1235,7 @@ static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { .enable_transfer = atiixp_in_enable_transfer, .flush_dma = atiixp_in_flush_dma, }; - + static struct atiixp_dma_ops snd_atiixp_spdif_dma_ops = { .type = ATI_DMA_SPDIF, .llp_offset = ATI_REG_SPDF_DMA_LINKPTR, @@ -1244,7 +1244,7 @@ static struct atiixp_dma_ops snd_atiixp_spdif_dma_ops = { .enable_transfer = atiixp_spdif_enable_transfer, .flush_dma = atiixp_spdif_flush_dma, }; - + static int __devinit snd_atiixp_pcm_new(struct atiixp *chip) { @@ -1255,7 +1255,7 @@ static int __devinit snd_atiixp_pcm_new(struct atiixp *chip) /* initialize constants */ chip->dmas[ATI_DMA_PLAYBACK].ops = &snd_atiixp_playback_dma_ops; chip->dmas[ATI_DMA_CAPTURE].ops = &snd_atiixp_capture_dma_ops; - if (! chip->spdif_over_aclink) + if (!chip->spdif_over_aclink) chip->dmas[ATI_DMA_SPDIF].ops = &snd_atiixp_spdif_dma_ops; /* assign AC97 pcm */ @@ -1293,9 +1293,9 @@ static int __devinit snd_atiixp_pcm_new(struct atiixp *chip) 64*1024, 128*1024); /* no SPDIF support on codec? */ - if (chip->pcms[ATI_PCM_SPDIF] && ! chip->pcms[ATI_PCM_SPDIF]->rates) + if (chip->pcms[ATI_PCM_SPDIF] && !chip->pcms[ATI_PCM_SPDIF]->rates) return 0; - + /* FIXME: non-48k sample rate doesn't work on my test machine with AD1888 */ if (chip->pcms[ATI_PCM_SPDIF]) chip->pcms[ATI_PCM_SPDIF]->rates = SNDRV_PCM_RATE_48000; @@ -1340,7 +1340,7 @@ static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id) status = atiixp_read(chip, ISR); - if (! status) + if (!status) return IRQ_NONE; /* process audio DMA */ @@ -1352,7 +1352,7 @@ static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id) snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_CAPTURE]); else if (status & ATI_REG_ISR_IN_STATUS) snd_atiixp_update_dma(chip, &chip->dmas[ATI_DMA_CAPTURE]); - if (! chip->spdif_over_aclink) { + if (!chip->spdif_over_aclink) { if (status & ATI_REG_ISR_SPDF_XRUN) snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_SPDIF]); else if (status & ATI_REG_ISR_SPDF_STATUS) @@ -1430,7 +1430,7 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock, ac97.pci = chip->pci; ac97.num = i; ac97.scaps = AC97_SCAP_SKIP_MODEM | AC97_SCAP_POWER_SAVE; - if (! chip->spdif_over_aclink) + if (!chip->spdif_over_aclink) ac97.scaps |= AC97_SCAP_NO_SPDIF; if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97[i])) < 0) { chip->ac97[i] = NULL; /* to be sure */ @@ -1440,7 +1440,7 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock, codec_count++; } - if (! codec_count) { + if (!codec_count) { snd_printk(KERN_ERR "atiixp: no codec available\n"); return -ENODEV; } @@ -1541,7 +1541,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "atiixp", &entry)) + if (!snd_card_proc_new(chip->card, "atiixp", &entry)) snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read); } #else /* !CONFIG_PROC_FS */ @@ -1559,7 +1559,7 @@ static int snd_atiixp_free(struct atiixp *chip) goto __hw_end; snd_atiixp_chip_stop(chip); - __hw_end: +__hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); if (chip->remap_addr) @@ -1665,7 +1665,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, if ((err = snd_atiixp_pcm_new(chip)) < 0) goto __error; - + snd_atiixp_proc_init(chip); snd_atiixp_chip_start(chip); @@ -1682,7 +1682,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); return 0; - __error: +__error: snd_card_free(card); return err; } diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f10630..b198f4c 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -19,7 +19,7 @@ * */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -240,7 +240,7 @@ struct atiixp_modem { unsigned long addr; void __iomem *remap_addr; int irq; - + struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97[NUM_ATI_CODECS]; @@ -319,7 +319,7 @@ static int snd_atiixp_update_bits(struct atiixp_modem *chip, unsigned int reg, * list. although we can change the list dynamically, in this version, * a static RING of buffer descriptors is used. * - * the ring is built in this function, and is set up to the hardware. + * the ring is built in this function, and is set up to the hardware. */ static int atiixp_build_dma_packets(struct atiixp_modem *chip, struct atiixp_dma *dma, @@ -399,7 +399,7 @@ static int snd_atiixp_acquire_codec(struct atiixp_modem *chip) int timeout = 1000; while (atiixp_read(chip, PHYS_OUT_ADDR) & ATI_REG_PHYS_OUT_ADDR_EN) { - if (! timeout--) { + if (!timeout--) { snd_printk(KERN_WARNING "atiixp-modem: codec acquire timeout\n"); return -EBUSY; } @@ -443,7 +443,7 @@ static void snd_atiixp_codec_write(struct atiixp_modem *chip, unsigned short reg, unsigned short val) { unsigned int data; - + if (snd_atiixp_acquire_codec(chip) < 0) return; data = ((unsigned int)val << ATI_REG_PHYS_OUT_DATA_SHIFT) | @@ -458,7 +458,7 @@ static unsigned short snd_atiixp_ac97_read(struct snd_ac97 *ac97, { struct atiixp_modem *chip = ac97->private_data; return snd_atiixp_codec_read(chip, ac97->num, reg); - + } static void snd_atiixp_ac97_write(struct snd_ac97 *ac97, unsigned short reg, @@ -489,9 +489,9 @@ static int snd_atiixp_aclink_reset(struct atiixp_modem *chip) atiixp_read(chip, CMD); udelay(10); atiixp_update(chip, CMD, ATI_REG_CMD_AC_SOFT_RESET, 0); - + timeout = 10; - while (! (atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) { + while (!(atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) { /* do a hard reset */ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET, ATI_REG_CMD_AC_SYNC); @@ -514,8 +514,8 @@ static int snd_atiixp_aclink_reset(struct atiixp_modem *chip) #ifdef CONFIG_PM static int snd_atiixp_aclink_down(struct atiixp_modem *chip) { - // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */ - // return -EBUSY; + /* if (atiixp_read(chip, MODEM_MIRROR) & 0x1) modem running, too? */ + /* return -EBUSY; */ atiixp_update(chip, CMD, ATI_REG_CMD_POWERDOWN | ATI_REG_CMD_AC_RESET, ATI_REG_CMD_POWERDOWN); @@ -570,7 +570,7 @@ static int snd_atiixp_chip_start(struct atiixp_modem *chip) /* set up spdif, enable burst mode */ reg = atiixp_read(chip, CMD); reg |= ATI_REG_CMD_BURST_EN; - if(!(reg & ATI_REG_CMD_MODEM_PRESENT)) + if (!(reg & ATI_REG_CMD_MODEM_PRESENT)) reg |= ATI_REG_CMD_MODEM_PRESENT; atiixp_write(chip, CMD, reg); @@ -635,7 +635,7 @@ static snd_pcm_uframes_t snd_atiixp_pcm_pointer(struct snd_pcm_substream *substr static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, struct atiixp_dma *dma) { - if (! dma->substream || ! dma->running) + if (!dma->substream || !dma->running) return; snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); @@ -647,7 +647,7 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, static void snd_atiixp_update_dma(struct atiixp_modem *chip, struct atiixp_dma *dma) { - if (! dma->substream || ! dma->running) + if (!dma->substream || !dma->running) return; snd_pcm_period_elapsed(dma->substream); } @@ -692,7 +692,7 @@ static int snd_atiixp_pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = -EINVAL; break; } - if (! err) { + if (!err) { snd_atiixp_check_bus_busy(chip); if (cmd == SNDRV_PCM_TRIGGER_STOP) { dma->ops->flush_dma(chip); @@ -750,7 +750,7 @@ static void atiixp_in_enable_transfer(struct atiixp_modem *chip, int on) { if (on) { unsigned int data = atiixp_read(chip, CMD); - if (! (data & ATI_REG_CMD_MODEM_RECEIVE_EN)) { + if (!(data & ATI_REG_CMD_MODEM_RECEIVE_EN)) { data |= ATI_REG_CMD_MODEM_RECEIVE_EN; atiixp_write(chip, CMD, data); } @@ -811,7 +811,7 @@ static int snd_atiixp_pcm_hw_params(struct snd_pcm_substream *substream, /* set up modem rate */ for (i = 0; i < NUM_ATI_CODECS; i++) { - if (! chip->ac97[i]) + if (!chip->ac97[i]) continue; snd_ac97_write(chip->ac97[i], AC97_LINE1_RATE, params_rate(hw_params)); snd_ac97_write(chip->ac97[i], AC97_LINE1_LEVEL, 0); @@ -978,7 +978,7 @@ static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = { .enable_transfer = atiixp_out_enable_transfer, .flush_dma = atiixp_out_flush_dma, }; - + static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = { .type = ATI_DMA_CAPTURE, .llp_offset = ATI_REG_MODEM_IN_DMA_LINKPTR, @@ -1027,7 +1027,7 @@ static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id) status = atiixp_read(chip, ISR); - if (! status) + if (!status) return IRQ_NONE; /* process audio DMA */ @@ -1102,7 +1102,7 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock) codec_count++; } - if (! codec_count) { + if (!codec_count) { snd_printk(KERN_ERR "atiixp-modem: no codec available\n"); return -ENODEV; } @@ -1184,7 +1184,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp_modem *chip) { struct snd_info_entry *entry; - if (! snd_card_proc_new(chip->card, "atiixp-modem", &entry)) + if (!snd_card_proc_new(chip->card, "atiixp-modem", &entry)) snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read); } #else @@ -1202,7 +1202,7 @@ static int snd_atiixp_free(struct atiixp_modem *chip) goto __hw_end; snd_atiixp_chip_stop(chip); - __hw_end: +__hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); if (chip->remap_addr) @@ -1306,7 +1306,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, if ((err = snd_atiixp_pcm_new(chip)) < 0) goto __error; - + snd_atiixp_proc_init(chip); snd_atiixp_chip_start(chip); @@ -1320,7 +1320,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); return 0; - __error: +__error: snd_card_free(card); return err; } diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 3f00ddf..efb0802 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -26,7 +26,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/delay.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index 6a3891a..3be6ac2 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -29,7 +29,7 @@ #include <linux/interrupt.h> #include <linux/delay.h> #include <asm/system.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 333007c..72cca10 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -159,7 +159,7 @@ * - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport. */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/init.h> #include <linux/pci.h> #include <linux/delay.h> @@ -1360,7 +1360,7 @@ snd_azf3328_playback_pointer(struct snd_pcm_substream *substream) /* calculate offset */ result -= bufptr; - frmres = bytes_to_frames( substream->runtime, result); + frmres = bytes_to_frames(substream->runtime, result); snd_azf3328_dbgplay("PLAY @ 0x%8lx, frames %8ld\n", result, frmres); return frmres; } @@ -1381,7 +1381,7 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) /* calculate offset */ result -= bufptr; - frmres = bytes_to_frames( substream->runtime, result); + frmres = bytes_to_frames(substream->runtime, result); snd_azf3328_dbgplay("REC @ 0x%8lx, frames %8ld\n", result, frmres); return frmres; } @@ -2063,14 +2063,14 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) snd_azf3328_game_inb(chip, 5) ); - for (tmp = 0; tmp < 0x07; tmp += 1) + for (tmp = 0; tmp < 0x07; tmp++) snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp)); - for (tmp = 0; tmp <= 0x07; tmp += 1) + for (tmp = 0; tmp <= 0x07; tmp++) snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n", tmp, inb(0x200 + tmp), inb(0x208 + tmp)); - for (tmp = 0; tmp <= 0x01; tmp += 1) + for (tmp = 0; tmp <= 0x01; tmp++) snd_azf3328_dbgmisc( "0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, " "mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n", diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 974e051..3038efb 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -119,7 +119,7 @@ enum azf_freq_t { /* general */ #define IDX_IO_42H 0x42 /* PU:0x0001 */ -/** DirectX timer, main interrupt area (FIXME: and something else?) **/ +/** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ /* timer countdown value; triggers IRQ when timer is finished */ #define TIMER_VALUE_MASK 0x000fffffUL @@ -204,7 +204,7 @@ enum azf_freq_t { /*** Gameport area port indices ***/ -/* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ +/* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ #define AZF_IO_SIZE_GAME 0x08 #define AZF_IO_SIZE_GAME_PM 0x06 @@ -333,7 +333,7 @@ enum { #define MIXER_ADVCTL2_MIX 0x0200 /* Mono output select 0=Mix, 1=Mic; Win driver: "MonoSelectSource"?? */ #define MIXER_ADVCTL2_3D 0x2000 /* 3D Enhancement 1=on */ #define MIXER_ADVCTL2_POP 0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */ - + #define IDX_MIXER_SOMETHING30H 0x30 /* used, but unknown??? */ /* driver internal flags */

1 0

[alsa-devel] [PATCH 02/24] sound/pci: coding style fixes: au88x0
by Alexander Beregalov 09 Sep '08

09 Sep '08

- <asm/io.h> -> <linux/io.h> - remove trailing whitespaces - convert comments Only compile tested. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/au88x0/au88x0.c | 46 ++++---- sound/pci/au88x0/au88x0.h | 4 +- sound/pci/au88x0/au88x0_a3d.c | 46 ++++---- sound/pci/au88x0/au88x0_core.c | 254 +++++++++++++++++++------------------- sound/pci/au88x0/au88x0_eq.c | 24 ++-- sound/pci/au88x0/au88x0_mixer.c | 2 +- sound/pci/au88x0/au88x0_mpu401.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 46 ++++---- sound/pci/au88x0/au88x0_synth.c | 46 ++++---- sound/pci/au88x0/au88x0_xtalk.c | 26 ++-- sound/pci/au88x0/au88x0_xtalk.h | 2 +- 11 files changed, 249 insertions(+), 249 deletions(-) diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e4..440be0b 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -125,7 +125,7 @@ static int snd_vortex_dev_free(struct snd_device *device) vortex_gameport_unregister(vortex); vortex_core_shutdown(vortex); - // Take down PCI interface. + /* Take down PCI interface. */ free_irq(vortex->irq, vortex); iounmap(vortex->mmio); pci_release_regions(vortex->pci_dev); @@ -148,7 +148,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) *rchip = NULL; - // check PCI availability (DMA). + /* check PCI availability (DMA). */ if ((err = pci_enable_device(pci)) < 0) return err; if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || @@ -166,7 +166,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) chip->card = card; - // initialize the stuff + /* initialize the stuff */ chip->pci_dev = pci; chip->io = pci_resource_start(pci, 0); chip->vendor = pci->vendor; @@ -174,8 +174,8 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) chip->card = card; chip->irq = -1; - // (1) PCI resource allocation - // Get MMIO area + /* (1) PCI resource allocation */ + /* Get MMIO area */ // if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; @@ -205,9 +205,9 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) chip->irq = pci->irq; pci_set_master(pci); - // End of PCI setup. + /* End of PCI setup. */ - // Register alsa root device. + /* Register alsa root device. */ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { goto alloc_out; } @@ -228,7 +228,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) pci_release_regions(chip->pci_dev); regions_out: pci_disable_device(chip->pci_dev); - //FIXME: this not the right place to unregister the gameport + /* FIXME: this not the right place to unregister the gameport */ vortex_gameport_unregister(chip); kfree(chip); return err; @@ -243,64 +243,64 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) vortex_t *chip; int err; - // (1) + /* (1) */ if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { dev++; return -ENOENT; } - // (2) + /* (2) */ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; - // (3) + /* (3) */ if ((err = snd_vortex_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; } snd_vortex_workaround(pci, pcifix[dev]); - // Card details needed in snd_vortex_midi + /* Card details needed in snd_vortex_midi */ strcpy(card->driver, CARD_NAME_SHORT); sprintf(card->shortname, "Aureal Vortex %s", CARD_NAME_SHORT); sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->io, chip->irq); - // (4) Alloc components. - // ADB pcm. + /* (4) Alloc components. */ + /* ADB pcm. */ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) { snd_card_free(card); return err; } #ifndef CHIP_AU8820 - // ADB SPDIF + /* ADB SPDIF */ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_SPDIF, 1)) < 0) { snd_card_free(card); return err; } - // A3D + /* A3D */ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_A3D, NR_A3D)) < 0) { snd_card_free(card); return err; } #endif /* - // ADB I2S + //ADB I2S if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_I2S, 1)) < 0) { snd_card_free(card); return err; } */ #ifndef CHIP_AU8810 - // WT pcm. + /* WT pcm. */ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_WT, NR_WT)) < 0) { snd_card_free(card); return err; } #endif - // snd_ac97_mixer and Vortex mixer. + /* snd_ac97_mixer and Vortex mixer. */ if ((err = snd_vortex_mixer(chip)) < 0) { snd_card_free(card); return err; @@ -329,12 +329,12 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) } #endif - // (5) + /* (5) */ if ((err = pci_read_config_word(pci, PCI_DEVICE_ID, &(chip->device))) < 0) { snd_card_free(card); return err; - } + } if ((err = pci_read_config_word(pci, PCI_VENDOR_ID, &(chip->vendor))) < 0) { snd_card_free(card); @@ -354,12 +354,12 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) } #endif - // (6) + /* (6) */ if ((err = snd_card_register(card)) < 0) { snd_card_free(card); return err; } - // (7) + /* (7) */ pci_set_drvdata(pci, card); dev++; vortex_connect_default(chip, 1); diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index cf46bba..02e949d 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -13,13 +13,13 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ - + #ifndef __SOUND_AU88X0_H #define __SOUND_AU88X0_H #ifdef __KERNEL__ #include <linux/pci.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/rawmidi.h> diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 649849e..526fb5f 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -49,7 +49,7 @@ static void a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack, short *GTrack, short *CTrack) { - // stub! + /* stub! */ } #endif @@ -219,7 +219,7 @@ static void a3dsrc_GetHrtfState(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) { vortex_t *vortex = (vortex_t *) (a->vortex); int i; - // FIXME: verify this! + /* FIXME: verify this! */ for (i = 0; i < HRTF_SZ; i++) aa[i] = hwread(vortex->mmio, @@ -245,10 +245,10 @@ static void a3dsrc_GetHrtfOutput(a3dsrc_t * a, short *left, short *right) #endif -/* Interaural Time Difference. - * "The other main clue that humans use to locate sounds, is called - * Interaural Time Difference (ITD). The differences in distance from - * the sound source to a listeners ears means that the sound will +/* Interaural Time Difference. + * "The other main clue that humans use to locate sounds, is called + * Interaural Time Difference (ITD). The differences in distance from + * the sound source to a listeners ears means that the sound will * reach one ear slightly before the other....", found somewhere with google.*/ static void a3dsrc_SetItdTarget(a3dsrc_t * a, short litd, short ritd) { @@ -265,7 +265,7 @@ static void a3dsrc_SetItdTarget(a3dsrc_t * a, short litd, short ritd) hwwrite(vortex->mmio, a3d_addrB(a->slice, a->source, A3D_B_ITDTarget), (ritd << 0x10) | litd); - //hwwrite(vortex->mmio, addr(0x191DF+5, this04, this08), (ritd<<0x10)|litd); + /* hwwrite(vortex->mmio, addr(0x191DF+5, this04, this08), (ritd<<0x10)|litd); */ } static void a3dsrc_SetItdCurrent(a3dsrc_t * a, short litd, short ritd) @@ -283,7 +283,7 @@ static void a3dsrc_SetItdCurrent(a3dsrc_t * a, short litd, short ritd) hwwrite(vortex->mmio, a3d_addrB(a->slice, a->source, A3D_B_ITDCurrent), (ritd << 0x10) | litd); - //hwwrite(vortex->mmio, addr(0x191DF+1, this04, this08), (ritd<<0x10)|litd); + /* hwwrite(vortex->mmio, addr(0x191DF+1, this04, this08), (ritd<<0x10)|litd); */ } static void a3dsrc_SetItdDline(a3dsrc_t * a, a3d_ItdDline_t const dline) @@ -392,7 +392,7 @@ static void a3dsrc_SetA3DSampleRate(a3dsrc_t * a, int sr) esp0 = (((esp0 & 0x7fffffff) | 0xB8000000) & 0x7) | ((sr & 0x1f) << 3); hwwrite(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd), esp0); - //hwwrite(vortex->mmio, 0x19C38 + (this08<<0xd), esp0); + /* hwwrite(vortex->mmio, 0x19C38 + (this08<<0xd), esp0); */ } static void a3dsrc_EnableA3D(a3dsrc_t * a) @@ -400,7 +400,7 @@ static void a3dsrc_EnableA3D(a3dsrc_t * a) vortex_t *vortex = (vortex_t *) (a->vortex); hwwrite(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd), 0xF0000001); - //hwwrite(vortex->mmio, 0x19C38 + (this08<<0xd), 0xF0000001); + /* hwwrite(vortex->mmio, 0x19C38 + (this08<<0xd), 0xF0000001); */ } static void a3dsrc_DisableA3D(a3dsrc_t * a) @@ -428,7 +428,7 @@ static void a3dsrc_GetA3DSampleRate(a3dsrc_t * a, int *sr) vortex_t *vortex = (vortex_t *) (a->vortex); *sr = ((hwread(vortex->mmio, A3D_SLICE_Control + (a->slice << 0xd)) >> 3) & 0x1f); - //*sr = ((hwread(vortex->mmio, 0x19C38 + (this08<<0xd))>>3)&0x1f); + /**sr = ((hwread(vortex->mmio, 0x19C38 + (this08<<0xd))>>3)&0x1f); */ } static void a3dsrc_GetA3DControlReg(a3dsrc_t * a, unsigned long *ctrl) @@ -463,7 +463,7 @@ static void a3dsrc_ZeroSliceIO(a3dsrc_t * a) static void a3dsrc_ZeroState(a3dsrc_t * a) { - //printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source); + /* printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source); */ a3dsrc_SetAtmosState(a, 0, 0, 0, 0); a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros); @@ -500,7 +500,7 @@ static void a3dsrc_ZeroStateA3D(a3dsrc_t * a) for (i = 0; i < 4; i++) { a->slice = i; a3dsrc_ZeroSliceIO(a); - //a3dsrc_ZeroState(a); + /* a3dsrc_ZeroState(a); */ } a->source = var2; a->slice = var; @@ -524,8 +524,8 @@ static void a3dsrc_ProgramPipe(a3dsrc_t * a) a3dsrc_SetHrtfTarget(a, A3dHrirImpulse, A3dHrirImpulse); /* Test: Sounds saturated. */ - //a3dsrc_SetHrtfCurrent(a, A3dHrirSatTest, A3dHrirSatTest); - //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); + /* a3dsrc_SetHrtfCurrent(a, A3dHrirSatTest, A3dHrirSatTest); */ + /* a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); */ } /* VDB = Vortex audio Dataflow Bus */ @@ -534,7 +534,7 @@ static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) { vortex_t *vortex = (vortex_t *) (a->vortex); - // ((aa >> 2) << 8) - (aa >> 2) + /* ((aa >> 2) << 8) - (aa >> 2) */ hwwrite(vortex->mmio, a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0); hwwrite(vortex->mmio, @@ -552,7 +552,7 @@ static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice) { a3dsrc_t *a3dsrc = &(v->a3d[source + (slice * 4)]); - //a3dsrc_t *a3dsrc = &(v->a3d[source + (slice*4)]); + /* a3dsrc_t *a3dsrc = &(v->a3d[source + (slice*4)]); */ a3dsrc->vortex = (void *)v; a3dsrc->source = source; /* source */ @@ -616,12 +616,12 @@ static void vortex_Vort3D_disable(vortex_t * v) static void vortex_Vort3D_connect(vortex_t * v, int en) { int i; - + // Disable AU8810 routes, since they seem to be wrong (in au8810.h). #ifdef CHIP_AU8810 return; #endif - + #if 1 /* Alloc Xtalk mixin resources */ v->mixxtlk[0] = @@ -642,7 +642,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) /* Connect A3D -> XTALK */ for (i = 0; i < 4; i++) { - // 2 outputs per each A3D slice. + /* 2 outputs per each A3D slice. */ vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2), ADB_XTALKIN(i)); vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2) + 1, ADB_XTALKIN(5 + i)); } @@ -689,8 +689,8 @@ static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en) ItdTCDefault, GainTCDefault, CoefTCDefault); /* Remark: zero gain is muted. */ - //a3dsrc_SetGainTarget(a,0,0); - //a3dsrc_SetGainCurrent(a,0,0); + /* a3dsrc_SetGainTarget(a,0,0); */ + /* a3dsrc_SetGainCurrent(a,0,0); */ a3dsrc_EnableA3D(a); } else { a3dsrc_DisableA3D(a); @@ -764,7 +764,7 @@ snd_vortex_a3d_filter_info(struct snd_kcontrol *kcontrol, static int snd_vortex_a3d_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - //a3dsrc_t *a = kcontrol->private_data; + /* a3dsrc_t *a = kcontrol->private_data; */ /* No read yet. Would this be really useable/needed ? */ return 0; diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index b070e57..9009d38 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -16,7 +16,7 @@ /* Vortex core low level functions. - + Author: Manuel Jander (mjander(a)users.sourceforge.cl) These functions are mainly the result of translations made from the original disassembly of the au88x0 binary drivers, @@ -73,7 +73,7 @@ 06-06-2003 Buffer shifter bugfix. Mixer volume fix. 07-12-2003 A3D routing finally fixed. Believed to be OK. 25-03-2004 Many thanks to Claudia, for such valuable bug reports. - + */ #include "au88x0.h" @@ -112,7 +112,7 @@ static int vortex_mix_getvolume(vortex_t * vortex, unsigned char mix) { int a; a = hwread(vortex->mmio, VORTEX_MIX_VOL_A + (mix << 2)) & 0xff; - //FP2LinearFrac(a); + /* FP2LinearFrac(a); */ return (a); } @@ -144,7 +144,7 @@ static void vortex_mix_rampvolume(vortex_t * vortex, int mix) { int ch; char a; - // This function is intended for ramping down only (see vortex_disableinput()). + /* This function is intended for ramping down only (see vortex_disableinput()). */ for (ch = 0; ch < 0x20; ch++) { if (((1 << ch) & rampchs[mix]) == 0) continue; @@ -260,8 +260,8 @@ vortex_mix_disableinput(vortex_t * vortex, unsigned char mix, int channel, { if (ramp) { rampchs[mix] |= (1 << channel); - // Register callback. - //vortex_mix_startrampvolume(vortex); + /* Register callback. */ + /* vortex_mix_startrampvolume(vortex); */ vortex_mix_killinput(vortex, mix, channel); } else vortex_mix_killinput(vortex, mix, channel); @@ -283,7 +283,7 @@ vortex_mixer_addWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) while (temp & 0x10) { prev = VORTEX_MIXER_RTBASE + ((temp & 0xf) << 2); temp = hwread(vortex->mmio, prev); - //printk(KERN_INFO "vortex: mixAddWTD: while addr=%x, val=%x\n", prev, temp); + /* printk(KERN_INFO "vortex: mixAddWTD: while addr=%x, val=%x\n", prev, temp); */ if ((++lifeboat) > 0xf) { printk(KERN_ERR "vortex_mixer_addWTD: lifeboat overflow\n"); @@ -299,7 +299,7 @@ static int vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) { int esp14 = -1, esp18, eax, ebx, edx, ebp, esi = 0; - //int esp1f=edi(while)=src, esp10=ch; + /* int esp1f=edi(while)=src, esp10=ch; */ eax = hwread(vortex->mmio, VORTEX_MIXER_SR); if (((1 << ch) & eax) == 0) { @@ -313,15 +313,15 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) if (mix == ebx) { ebx = VORTEX_MIXER_RTBASE + (mix << 2); edx = hwread(vortex->mmio, ebx); - //7b60 + /* 7b60 */ hwwrite(vortex->mmio, ebp, edx); hwwrite(vortex->mmio, ebx, 0); } else { - //7ad3 + /* 7ad3 */ edx = hwread(vortex->mmio, VORTEX_MIXER_RTBASE + (ebx << 2)); - //printk(KERN_INFO "vortex: mixdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src); + /* printk(KERN_INFO "vortex: mixdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src); */ while ((edx & 0xf) != mix) { if ((esi) > 0xf) { printk(KERN_ERR @@ -334,21 +334,21 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) edx = hwread(vortex->mmio, VORTEX_MIXER_RTBASE + ebp); - //printk(KERN_INFO "vortex: mixdelWTD: while addr=%x, val=%x\n", ebp, edx); + /* printk(KERN_INFO "vortex: mixdelWTD: while addr=%x, val=%x\n", ebp, edx); */ esi++; } - //7b30 + /* 7b30 */ ebp = ebx << 2; if (edx & 0x10) { /* Delete entry in between others */ ebx = VORTEX_MIXER_RTBASE + ((edx & 0xf) << 2); edx = hwread(vortex->mmio, ebx); - //7b60 + /* 7b60 */ hwwrite(vortex->mmio, VORTEX_MIXER_RTBASE + ebp, edx); hwwrite(vortex->mmio, ebx, 0); - //printk(KERN_INFO "vortex mixdelWTD between addr= 0x%x, val= 0x%x\n", ebp, edx); + /* printk(KERN_INFO "vortex mixdelWTD between addr= 0x%x, val= 0x%x\n", ebp, edx); */ } else { /* Delete last entry */ - //7b83 + /* 7b83 */ if (esp14 == -1) hwwrite(vortex->mmio, VORTEX_MIXER_CHNBASE + @@ -358,7 +358,7 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) hwwrite(vortex->mmio, VORTEX_MIXER_RTBASE + (esp14 << 2), ebx); - //printk(KERN_INFO "vortex mixdelWTD last addr= 0x%x, val= 0x%x\n", esp14, ebx); + /* printk(KERN_INFO "vortex mixdelWTD last addr= 0x%x, val= 0x%x\n", esp14, ebx); */ } hwwrite(vortex->mmio, VORTEX_MIXER_RTBASE + ebp, 0); @@ -366,8 +366,8 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) } } } else { - //printk(KERN_INFO "removed last mix\n"); - //7be0 + /* printk(KERN_INFO "removed last mix\n"); */ + /* 7be0 */ vortex_mixer_dis_sr(vortex, ch); hwwrite(vortex->mmio, ebp, 0); } @@ -379,7 +379,7 @@ static void vortex_mixer_init(vortex_t * vortex) u32 addr; int x; - // FIXME: get rid of this crap. + /* FIXME: get rid of this crap. */ memset(mchannels, 0, NR_MIXOUT * sizeof(int)); memset(rampchs, 0, NR_MIXOUT * sizeof(int)); @@ -572,12 +572,12 @@ static void vortex_src_setupchannel(vortex_t * card, unsigned char src, unsigned int cr, unsigned int b, int sweep, int d, int dirplay, int sl, unsigned int tr, int thsource) { - // noplayback: d=2,4,7,0xa,0xb when using first 2 src's. - // c: enables pitch sweep. - // looks like g is c related. Maybe g is a sweep parameter ? - // g = cvr - // dirplay: 0 = recording, 1 = playback - // d = src hw index. + /* noplayback: d=2,4,7,0xa,0xb when using first 2 src's. */ + /* c: enables pitch sweep. */ + /* looks like g is c related. Maybe g is a sweep parameter ? */ + /* g = cvr */ + /* dirplay: 0 = recording, 1 = playback */ + /* d = src hw index. */ int esi, ebp = 0, esp10; @@ -634,7 +634,7 @@ static void vortex_src_setupchannel(vortex_t * card, unsigned char src, hwwrite(card->mmio, VORTEX_SRC_U2 + (src << 2), (tr << 0x11) | (dirplay << 0x10) | (ebp << 0x8) | esp10); /* 0x30f00 e=g=1 esp10=0 ebp=f */ - //printk(KERN_INFO "vortex: SRC %d, d=0x%x, esi=0x%x, esp10=0x%x, ebp=0x%x\n", src, d, esi, esp10, ebp); + /* printk(KERN_INFO "vortex: SRC %d, d=0x%x, esi=0x%x, esp10=0x%x, ebp=0x%x\n", src, d, esi, esp10, ebp); */ } static void vortex_srcblock_init(vortex_t * vortex) @@ -647,15 +647,15 @@ static void vortex_srcblock_init(vortex_t * vortex) vortex_src_init(&vortex_src[x], x); } */ - //addr = 0xcc3c; - //addr = 0x26c3c; + /* addr = 0xcc3c; */ + /* addr = 0x26c3c; */ addr = VORTEX_SRC_RTBASE + 0x3c; for (x = 0xf; x >= 0; x--) { hwwrite(vortex->mmio, addr, 0); addr -= 4; } - //addr = 0xcc94; - //addr = 0x26c94; + /* addr = 0xcc94; */ + /* addr = 0x26c94; */ addr = VORTEX_SRC_CHNBASE + 0x54; for (x = 0x15; x >= 0; x--) { hwwrite(vortex->mmio, addr, 0); @@ -667,7 +667,7 @@ static int vortex_src_addWTD(vortex_t * vortex, unsigned char src, unsigned char ch) { int temp, lifeboat = 0, prev; - // esp13 = src + /* esp13 = src */ temp = hwread(vortex->mmio, VORTEX_SRCBLOCK_SR); if ((temp & (1 << ch)) == 0) { @@ -677,12 +677,12 @@ vortex_src_addWTD(vortex_t * vortex, unsigned char src, unsigned char ch) } prev = VORTEX_SRC_CHNBASE + (ch << 2); /*ebp */ temp = hwread(vortex->mmio, prev); - //while (temp & NR_SRC) { + /* while (temp & NR_SRC) { */ while (temp & 0x10) { prev = VORTEX_SRC_RTBASE + ((temp & 0xf) << 2); /*esp12 */ - //prev = VORTEX_SRC_RTBASE + ((temp & (NR_SRC-1)) << 2); /*esp12*/ + /* prev = VORTEX_SRC_RTBASE + ((temp & (NR_SRC-1)) << 2); esp12 */ temp = hwread(vortex->mmio, prev); - //printk(KERN_INFO "vortex: srcAddWTD: while addr=%x, val=%x\n", prev, temp); + /* printk(KERN_INFO "vortex: srcAddWTD: while addr=%x, val=%x\n", prev, temp); */ if ((++lifeboat) > 0xf) { printk(KERN_ERR "vortex_src_addWTD: lifeboat overflow\n"); @@ -690,7 +690,7 @@ vortex_src_addWTD(vortex_t * vortex, unsigned char src, unsigned char ch) } } hwwrite(vortex->mmio, VORTEX_SRC_RTBASE + ((temp & 0xf) << 2), src); - //hwwrite(vortex->mmio, prev, (temp & (NR_SRC-1)) | NR_SRC); + /* hwwrite(vortex->mmio, prev, (temp & (NR_SRC-1)) | NR_SRC); */ hwwrite(vortex->mmio, prev, (temp & 0xf) | 0x10); return 1; } @@ -699,7 +699,7 @@ static int vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) { int esp14 = -1, esp18, eax, ebx, edx, ebp, esi = 0; - //int esp1f=edi(while)=src, esp10=ch; + /* int esp1f=edi(while)=src, esp10=ch; */ eax = hwread(vortex->mmio, VORTEX_SRCBLOCK_SR); if (((1 << ch) & eax) == 0) { @@ -713,15 +713,15 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) if (src == ebx) { ebx = VORTEX_SRC_RTBASE + (src << 2); edx = hwread(vortex->mmio, ebx); - //7b60 + /* 7b60 */ hwwrite(vortex->mmio, ebp, edx); hwwrite(vortex->mmio, ebx, 0); } else { - //7ad3 + /* 7ad3 */ edx = hwread(vortex->mmio, VORTEX_SRC_RTBASE + (ebx << 2)); - //printk(KERN_INFO "vortex: srcdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src); + /* printk(KERN_INFO "vortex: srcdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src); */ while ((edx & 0xf) != src) { if ((esi) > 0xf) { printk @@ -734,21 +734,21 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) edx = hwread(vortex->mmio, VORTEX_SRC_RTBASE + ebp); - //printk(KERN_INFO "vortex: srcdelWTD: while addr=%x, val=%x\n", ebp, edx); + /* printk(KERN_INFO "vortex: srcdelWTD: while addr=%x, val=%x\n", ebp, edx); */ esi++; } - //7b30 + /* 7b30 */ ebp = ebx << 2; if (edx & 0x10) { /* Delete entry in between others */ ebx = VORTEX_SRC_RTBASE + ((edx & 0xf) << 2); edx = hwread(vortex->mmio, ebx); - //7b60 + /* 7b60 */ hwwrite(vortex->mmio, VORTEX_SRC_RTBASE + ebp, edx); hwwrite(vortex->mmio, ebx, 0); - //printk(KERN_INFO "vortex srcdelWTD between addr= 0x%x, val= 0x%x\n", ebp, edx); + /* printk(KERN_INFO "vortex srcdelWTD between addr= 0x%x, val= 0x%x\n", ebp, edx); */ } else { /* Delete last entry */ - //7b83 + /* 7b83 */ if (esp14 == -1) hwwrite(vortex->mmio, VORTEX_SRC_CHNBASE + @@ -758,7 +758,7 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) hwwrite(vortex->mmio, VORTEX_SRC_RTBASE + (esp14 << 2), ebx); - //printk(KERN_INFO"vortex srcdelWTD last addr= 0x%x, val= 0x%x\n", esp14, ebx); + /* printk(KERN_INFO"vortex srcdelWTD last addr= 0x%x, val= 0x%x\n", esp14, ebx); */ } hwwrite(vortex->mmio, VORTEX_SRC_RTBASE + ebp, 0); @@ -766,14 +766,14 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) } } } else { - //7be0 + /* 7be0 */ vortex_src_dis_sr(vortex, ch); hwwrite(vortex->mmio, ebp, 0); } return 1; } - /*FIFO*/ + /*FIFO*/ static void vortex_fifo_clearadbdata(vortex_t * vortex, int fifo, int x) @@ -809,7 +809,7 @@ vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int b, int priority, int empty, int valid, int f) { int temp, lifeboat = 0; - //int this_8[NR_ADB] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; /* position */ + /* int this_8[NR_ADB] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; position */ int this_4 = 0x2; /* f seems priority related. * CAsp4AdbDma::SetPriority is the only place that calls SetAdbCtrl with f set to 1 @@ -826,12 +826,12 @@ vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int b, int priority, } while (temp & FIFO_RDONLY); - // AU8830 semes to take some special care about fifo content (data). - // But i'm just to lazy to translate that :) + /* AU8830 semes to take some special care about fifo content (data). */ + /* But i'm just to lazy to translate that :) */ if (valid) { if ((temp & FIFO_VALID) == 0) { - //this_8[fifo] = 0; - vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE); // this_4 + /* this_8[fifo] = 0; */ + vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE); /* this_4 */ #ifdef CHIP_AU8820 temp = (this_4 & 0x1f) << 0xb; #else @@ -923,7 +923,7 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, if (valid) { if ((temp & FIFO_VALID) == 0) { - vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE); // this_4 + vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE); /* this_4 */ #ifdef CHIP_AU8820 temp = (this_4 & 0x1f) << 0xb; #else @@ -966,7 +966,7 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp); hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2)); -/* +/* do { temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2)); if (lifeboat++ > 0xbb8) { @@ -974,8 +974,8 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, break; } } while ((temp & FIFO_RDONLY)&&(temp & FIFO_VALID)&&(temp != 0xFFFFFFFF)); - - + + if (valid) { if (temp & FIFO_VALID) { temp = 0x40000; @@ -1001,8 +1001,8 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, temp |= 0x00400000; hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp); temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2)); - //((temp >> 6) & 0x3f) - + //((temp >> 6) & 0x3f) + priority = 0; if (((temp & 0x0fc0) ^ ((temp >> 6) & 0x0fc0)) & 0FFFFFFC0) vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE); @@ -1014,7 +1014,7 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, temp = (temp & 0xffffffdf) | ((empty & 1) << 5); hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp); } - + */ /* @@ -1062,7 +1062,7 @@ static void vortex_fifo_init(vortex_t * vortex) #endif /* trigger... */ #ifdef CHIP_AU8820 - hwwrite(vortex->mmio, 0xf8c0, 0xd03); //0x0843 0xd6b + hwwrite(vortex->mmio, 0xf8c0, 0xd03); /* 0x0843 0xd6b */ #else #ifdef CHIP_AU8830 hwwrite(vortex->mmio, 0x17000, 0x61); /* wt a */ @@ -1089,7 +1089,7 @@ static void vortex_adbdma_setfirstbuffer(vortex_t * vortex, int adbdma) static void vortex_adbdma_setstartbuffer(vortex_t * vortex, int adbdma, int sb) { stream_t *dma = &vortex->dma_adb[adbdma]; - //hwwrite(vortex->mmio, VORTEX_ADBDMA_START + (adbdma << 2), sb << (((NR_ADB-1)-((adbdma&0xf)*2)))); + /* hwwrite(vortex->mmio, VORTEX_ADBDMA_START + (adbdma << 2), sb << (((NR_ADB-1)-((adbdma&0xf)*2)))); */ hwwrite(vortex->mmio, VORTEX_ADBDMA_START + (adbdma << 2), sb << ((0xf - (adbdma & 0xf)) * 2)); dma->period_real = dma->period_virt = sb; @@ -1135,7 +1135,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, snd_pcm_sgbuf_get_addr(dma->substream, 0)); break; } - //printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1); + /* printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1); */ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0); hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1); @@ -1195,7 +1195,7 @@ static int vortex_adbdma_bufshift(vortex_t * vortex, int adbdma) pp = dma->period_real + i; if (pp >= 4) pp -= 4; - //hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE+(((adbdma << 2)+pp) << 2), dma->table[p].addr); + /* hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE+(((adbdma << 2)+pp) << 2), dma->table[p].addr); */ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (((adbdma << 2) + pp) << 2), snd_pcm_sgbuf_get_addr(dma->substream, @@ -1222,7 +1222,7 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { int p, pp, i; /* refresh hw page table */ - for (i=0 ; i < 4 && i < dma->nr_periods; i++) { + for (i = 0; i < 4 && i < dma->nr_periods; i++) { /* p: audio buffer page index */ p = dma->period_virt + i; if (p >= dma->nr_periods) @@ -1258,7 +1258,7 @@ static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma) { - int this_8 = 0 /*empty */ , this_4 = 0 /*priority */ ; + int this_8 = 0, /*empty */ this_4 = 0; /*priority */ stream_t *dma = &vortex->dma_adb[adbdma]; switch (dma->fifo_status) { @@ -1325,7 +1325,7 @@ static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma) dma->fifo_status = FIFO_PAUSE; } -#if 0 // Using pause instead +#if 0 /* Using pause instead */ static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; @@ -1346,7 +1346,7 @@ static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma) #ifndef CHIP_AU8810 static void vortex_wtdma_setfirstbuffer(vortex_t * vortex, int wtdma) { - //int this_7c=dma_ctrl; + /* int this_7c=dma_ctrl; */ stream_t *dma = &vortex->dma_wt[wtdma]; hwwrite(vortex->mmio, VORTEX_WTDMA_CTRL + (wtdma << 2), dma->dma_ctrl); @@ -1355,7 +1355,7 @@ static void vortex_wtdma_setfirstbuffer(vortex_t * vortex, int wtdma) static void vortex_wtdma_setstartbuffer(vortex_t * vortex, int wtdma, int sb) { stream_t *dma = &vortex->dma_wt[wtdma]; - //hwwrite(vortex->mmio, VORTEX_WTDMA_START + (wtdma << 2), sb << ((0x1f-(wtdma&0xf)*2))); + /* hwwrite(vortex->mmio, VORTEX_WTDMA_START + (wtdma << 2), sb << ((0x1f-(wtdma&0xf)*2))); */ hwwrite(vortex->mmio, VORTEX_WTDMA_START + (wtdma << 2), sb << ((0xf - (wtdma & 0xf)) * 2)); dma->period_real = dma->period_virt = sb; @@ -1410,7 +1410,7 @@ vortex_wtdma_setmode(vortex_t * vortex, int wtdma, int ie, int fmt, int d, { stream_t *dma = &vortex->dma_wt[wtdma]; - //dma->this_08 = e; + /* dma->this_08 = e; */ dma->dma_unknown = d; dma->dma_ctrl = 0; dma->dma_ctrl = @@ -1501,7 +1501,7 @@ static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) int temp; temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)); - //temp = (temp & POS_MASK) + (((temp>>WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK)*(dma->cfg0&POS_MASK)); + /* temp = (temp & POS_MASK) + (((temp>>WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK)*(dma->cfg0&POS_MASK)); */ temp = (temp & POS_MASK) + ((dma->period_virt) * (dma->period_bytes)); return temp; } @@ -1846,7 +1846,7 @@ vortex_connection_mixin_mix(vortex_t * vortex, int en, unsigned char mixin, { if (en) { vortex_mix_enableinput(vortex, mix, mixin); - vortex_mix_setinputvolumebyte(vortex, mix, mixin, MIX_DEFIGAIN); // added to original code. + vortex_mix_setinputvolumebyte(vortex, mix, mixin, MIX_DEFIGAIN); /* added to original code. */ } else vortex_mix_disableinput(vortex, mix, mixin, a); } @@ -1883,7 +1883,7 @@ vortex_connection_mix_adb(vortex_t * vortex, int en, unsigned char ch, unsigned char mix, unsigned char dest) { vortex_route(vortex, en, ch, ADB_MIXOUT(mix), dest); - vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); // added to original code. + vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); /* added to original code. */ } // mixer to src. @@ -1892,7 +1892,7 @@ vortex_connection_mix_src(vortex_t * vortex, int en, unsigned char ch, unsigned char mix, unsigned char src) { vortex_route(vortex, en, ch, ADB_MIXOUT(mix), ADB_SRCIN(src)); - vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); // added to original code. + vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); /* added to original code. */ } #if 0 @@ -1943,7 +1943,7 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[]) vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1)); #else #if 1 - // Connect front channels through EQ. + /* Connect front channels through EQ. */ vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_EQIN(0)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_EQIN(1)); /* Lower volume, since EQ has some gain. */ @@ -1959,10 +1959,10 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[]) ADB_CODECOUT(0 + 4)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[3], ADB_CODECOUT(1 + 4)); - //printk("SDAC detected "); + /* printk("SDAC detected "); */ } #else - // Use plain direct output to codec. + /* Use plain direct output to codec. */ vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1)); #endif @@ -1989,7 +1989,7 @@ vortex_connect_codecrec(vortex_t * vortex, int en, unsigned char mixin0, static int resnum[VORTEX_RESOURCE_LAST] = { NR_ADB, NR_SRC, NR_MIXIN, NR_MIXOUT, NR_A3D }; /* - Checkout/Checkin resource of given type. + Checkout/Checkin resource of given type. resmap: resource map to be used. If NULL means that we want to allocate a DMA resource (root of all other resources of a dma channel). out: Mean checkout if != 0. Else mean Checkin resource. @@ -2013,7 +2013,7 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) resmap[restype] |= (1 << i); else vortex->dma_adb[i].resources[restype] |= (1 << i); - //printk("vortex: ResManager: type %d out %d\n", restype, i); + /* printk("vortex: ResManager: type %d out %d\n", restype, i); */ return i; } } @@ -2024,7 +2024,7 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) for (i = 0; i < qty; i++) { if (resmap[restype] & (1 << i)) { resmap[restype] &= ~(1 << i); - //printk("vortex: ResManager: type %d in %d\n",restype, i); + /* printk("vortex: ResManager: type %d in %d\n",restype, i); */ return i; } } @@ -2039,7 +2039,7 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type); static void vortex_connect_default(vortex_t * vortex, int en) { - // Connect AC97 codec. + /* Connect AC97 codec. */ vortex->mixplayb[0] = vortex_adb_checkinout(vortex, vortex->fixed_res, en, VORTEX_RESOURCE_MIXOUT); vortex->mixplayb[1] = vortex_adb_checkinout(vortex, vortex->fixed_res, en, @@ -2058,7 +2058,7 @@ static void vortex_connect_default(vortex_t * vortex, int en) VORTEX_RESOURCE_MIXIN); vortex_connect_codecrec(vortex, en, MIX_CAPT(0), MIX_CAPT(1)); - // Connect SPDIF + /* Connect SPDIF */ #ifndef CHIP_AU8820 vortex->mixspdif[0] = vortex_adb_checkinout(vortex, vortex->fixed_res, en, VORTEX_RESOURCE_MIXOUT); @@ -2069,20 +2069,20 @@ static void vortex_connect_default(vortex_t * vortex, int en) vortex_connection_mix_adb(vortex, en, 0x14, vortex->mixspdif[1], ADB_SPDIFOUT(1)); #endif - // Connect WT + /* Connect WT */ #ifndef CHIP_AU8810 vortex_wt_connect(vortex, en); #endif - // A3D (crosstalk canceler and A3D slices). AU8810 disabled for now. + /* A3D (crosstalk canceler and A3D slices). AU8810 disabled for now. */ #ifndef CHIP_AU8820 vortex_Vort3D_connect(vortex, en); #endif - // Connect I2S + /* Connect I2S */ + + /* Connect DSP interface for SQ3500 turbo (not here i think...) */ - // Connect DSP interface for SQ3500 turbo (not here i think...) + /* Connect AC98 modem codec */ - // Connect AC98 modem codec - } /* @@ -2099,7 +2099,7 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) { stream_t *stream; int i, en; - + if ((nr_ch == 3) || ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2))) return -EBUSY; @@ -2203,13 +2203,13 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) #ifndef CHIP_AU8820 if (stream->type == VORTEX_PCM_A3D) { vortex_connection_adbdma_src(vortex, en, - src[nr_ch - 1], + src[nr_ch - 1], dma, src[i]); vortex_route(vortex, en, 0x11, ADB_SRCOUT(src[i]), ADB_A3DIN(a3d)); /* XTalk test. */ - //vortex_route(vortex, en, 0x11, dma, ADB_XTALKIN(i?9:4)); - //vortex_route(vortex, en, 0x11, ADB_SRCOUT(src[i]), ADB_XTALKIN(i?4:9)); + /* vortex_route(vortex, en, 0x11, dma, ADB_XTALKIN(i?9:4)); */ + /* vortex_route(vortex, en, 0x11, ADB_SRCOUT(src[i]), ADB_XTALKIN(i?4:9)); */ } if (stream->type == VORTEX_PCM_SPDIF) vortex_route(vortex, en, 0x14, @@ -2331,7 +2331,7 @@ vortex_adb_setsrc(vortex_t * vortex, int adbdma, unsigned int rate, int dir) stream_t *stream = &(vortex->dma_adb[adbdma]); int i, cvrt; - /* dir=1:play ; dir=0:rec */ + /* dir=1:play; dir=0:rec */ if (dir) cvrt = SRC_RATIO(rate, 48000); else @@ -2348,7 +2348,7 @@ vortex_adb_setsrc(vortex_t * vortex, int adbdma, unsigned int rate, int dir) static void vortex_settimer(vortex_t * vortex, int period) { - //set the timer period to <period> 48000ths of a second. + /* set the timer period to <period> 48000ths of a second. */ hwwrite(vortex->mmio, VORTEX_IRQ_STAT, period); } @@ -2368,7 +2368,7 @@ static void vortex_disable_timer_int(vortex_t * card) #endif static void vortex_enable_int(vortex_t * card) { - // CAsp4ISR__EnableVortexInt_void_ + /* CAsp4ISR__EnableVortexInt_void_ */ hwwrite(card->mmio, VORTEX_CTRL, hwread(card->mmio, VORTEX_CTRL) | CTRL_IRQ_ENABLE); hwwrite(card->mmio, VORTEX_IRQ_CTRL, @@ -2387,26 +2387,26 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) int i, handled; u32 source; - //check if the interrupt is ours. + /* check if the interrupt is ours. */ if (!(hwread(vortex->mmio, VORTEX_STAT) & 0x1)) return IRQ_NONE; - // This is the Interrupt Enable flag we set before (consistency check). + /* This is the Interrupt Enable flag we set before (consistency check). */ if (!(hwread(vortex->mmio, VORTEX_CTRL) & CTRL_IRQ_ENABLE)) return IRQ_NONE; source = hwread(vortex->mmio, VORTEX_IRQ_SOURCE); - // Reset IRQ flags. + /* Reset IRQ flags. */ hwwrite(vortex->mmio, VORTEX_IRQ_SOURCE, source); hwread(vortex->mmio, VORTEX_IRQ_SOURCE); - // Is at least one IRQ flag set? + /* Is at least one IRQ flag set? */ if (source == 0) { printk(KERN_ERR "vortex: missing irq source\n"); return IRQ_NONE; } handled = 0; - // Attend every interrupt source. + /* Attend every interrupt source. */ if (unlikely(source & IRQ_ERR_MASK)) { if (source & IRQ_FATAL) { printk(KERN_ERR "vortex: IRQ fatal error\n"); @@ -2430,7 +2430,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) spin_lock(&vortex->lock); for (i = 0; i < NR_ADB; i++) { if (vortex->dma_adb[i].fifo_status == FIFO_START) { - if (vortex_adbdma_bufshift(vortex, i)) ; + if (vortex_adbdma_bufshift(vortex, i)); spin_unlock(&vortex->lock); snd_pcm_period_elapsed(vortex->dma_adb[i]. substream); @@ -2440,7 +2440,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) #ifndef CHIP_AU8810 for (i = 0; i < NR_WT; i++) { if (vortex->dma_wt[i].fifo_status == FIFO_START) { - if (vortex_wtdma_bufshift(vortex, i)) ; + if (vortex_wtdma_bufshift(vortex, i)); spin_unlock(&vortex->lock); snd_pcm_period_elapsed(vortex->dma_wt[i]. substream); @@ -2451,7 +2451,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) spin_unlock(&vortex->lock); handled = 1; } - //Acknowledge the Timer interrupt + /* Acknowledge the Timer interrupt */ if (source & IRQ_TIMER) { hwread(vortex->mmio, VORTEX_IRQ_STAT); handled = 1; @@ -2510,7 +2510,7 @@ static void vortex_codec_init(vortex_t * vortex) } static void -vortex_codec_write(struct snd_ac97 * codec, unsigned short addr, unsigned short data) +vortex_codec_write(struct snd_ac97 *codec, unsigned short addr, unsigned short data) { vortex_t *card = (vortex_t *) codec->private_data; @@ -2529,13 +2529,13 @@ vortex_codec_write(struct snd_ac97 * codec, unsigned short addr, unsigned short ((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK) | ((data << VORTEX_CODEC_DATSHIFT) & VORTEX_CODEC_DATMASK) | VORTEX_CODEC_WRITE | - (codec->num << VORTEX_CODEC_ID_SHIFT) ); + (codec->num << VORTEX_CODEC_ID_SHIFT)); /* Flush Caches. */ hwread(card->mmio, VORTEX_CODEC_IO); } -static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short addr) +static unsigned short vortex_codec_read(struct snd_ac97 *codec, unsigned short addr) { vortex_t *card = (vortex_t *) codec->private_data; @@ -2552,7 +2552,7 @@ static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short } /* set up read address */ read_addr = ((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK) | - (codec->num << VORTEX_CODEC_ID_SHIFT) ; + (codec->num << VORTEX_CODEC_ID_SHIFT); hwwrite(card->mmio, VORTEX_CODEC_IO, read_addr); /* wait for address */ @@ -2579,10 +2579,10 @@ static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode) /* CAsp4Spdif::InitializeSpdifHardware(void) */ hwwrite(vortex->mmio, VORTEX_SPDIF_FLAGS, hwread(vortex->mmio, VORTEX_SPDIF_FLAGS) & 0xfff3fffd); - //for (i=0x291D4; i<0x29200; i+=4) + /* for (i = 0x291D4; i < 0x29200; i += 4) */ for (i = 0; i < 11; i++) hwwrite(vortex->mmio, VORTEX_SPDIF_CFG1 + (i << 2), 0); - //hwwrite(vortex->mmio, 0x29190, hwread(vortex->mmio, 0x29190) | 0xc0000); + /* hwwrite(vortex->mmio, 0x29190, hwread(vortex->mmio, 0x29190) | 0xc0000); */ hwwrite(vortex->mmio, VORTEX_CODEC_EN, hwread(vortex->mmio, VORTEX_CODEC_EN) | EN_SPDIF); @@ -2644,8 +2644,8 @@ static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode) break; } - /* looks like the next 2 lines transfer a 16-bit value into 2 8-bit - registers. seems to be for the standard IEC/SPDIF initialization + /* looks like the next 2 lines transfer a 16-bit value into 2 8-bit + registers. seems to be for the standard IEC/SPDIF initialization stuff */ hwwrite(vortex->mmio, VORTEX_SPDIF_CFG0, this_38 & 0xffff); hwwrite(vortex->mmio, VORTEX_SPDIF_CFG1, this_38 >> 0x10); @@ -2677,7 +2677,7 @@ static int __devinit vortex_core_init(vortex_t * vortex) /* Init audio engine. */ vortex_adbdma_init(vortex); - hwwrite(vortex->mmio, VORTEX_ENGINE_CTRL, 0x0); //, 0xc83c7e58, 0xc5f93e58 + hwwrite(vortex->mmio, VORTEX_ENGINE_CTRL, 0x0); /*, 0xc83c7e58, 0xc5f93e58 */ vortex_adb_init(vortex); /* Init processing blocks. */ vortex_fifo_init(vortex); @@ -2691,16 +2691,16 @@ static int __devinit vortex_core_init(vortex_t * vortex) #ifndef CHIP_AU8810 vortex_wt_init(vortex); #endif - // Moved to au88x0.c - //vortex_connect_default(vortex, 1); + /* Moved to au88x0.c */ + /* vortex_connect_default(vortex, 1); */ vortex_settimer(vortex, 0x90); - // Enable Interrupts. - // vortex_enable_int() must be first !! - // hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, 0); - // vortex_enable_int(vortex); - //vortex_enable_timer_int(vortex); - //vortex_disable_timer_int(vortex); + /* Enable Interrupts. */ + /* vortex_enable_int() must be first !! */ + /* hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, 0); */ + /* vortex_enable_int(vortex); */ + /* vortex_enable_timer_int(vortex); */ + /* vortex_disable_timer_int(vortex); */ printk(KERN_INFO "done.\n"); spin_lock_init(&vortex->lock); @@ -2716,7 +2716,7 @@ static int vortex_core_shutdown(vortex_t * vortex) vortex_eq_free(vortex); vortex_Vort3D_disable(vortex); #endif - //vortex_disable_timer_int(vortex); + /* vortex_disable_timer_int(vortex); */ vortex_disable_int(vortex); vortex_connect_default(vortex, 0); /* Reset all DMA fifos. */ @@ -2725,8 +2725,8 @@ static int vortex_core_shutdown(vortex_t * vortex) vortex_adb_init(vortex); /* Disable MPU401 */ - //hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, hwread(vortex->mmio, VORTEX_IRQ_CTRL) & ~IRQ_MIDI); - //hwwrite(vortex->mmio, VORTEX_CTRL, hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_EN); + /* hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, hwread(vortex->mmio, VORTEX_IRQ_CTRL) & ~IRQ_MIDI); */ + /* hwwrite(vortex->mmio, VORTEX_CTRL, hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_EN); */ hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, 0); hwwrite(vortex->mmio, VORTEX_CTRL, 0); @@ -2795,17 +2795,17 @@ vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod) switch (encod) { case 0: if (bits == 0x10) - a = 8; // 16 bit + a = 8; /* 16 bit */ break; case 1: if (bits == 8) - a = 1; // 8 bit + a = 1; /* 8 bit */ break; case 2: - a = 2; // U_LAW + a = 2; /* U_LAW */ break; case 3: - a = 3; // A_LAW + a = 3; /* A_LAW */ break; } switch (nch) { diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index 38602b8..5b911b9 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -4,7 +4,7 @@ * * Sun Jun 8 18:19:19 2003 * 2003 Manuel Jander (mjander(a)users.sourceforge.net) - * + * * 02 July 2003: First time something works :) * November 2003: A3D Bypass code completed but untested. * @@ -32,7 +32,7 @@ /* The Aureal Hardware EQ is found on AU8810 and AU8830 chips only. - it has 4 inputs (2 for general mix, 2 for A3D) and 2 outputs (supposed + it has 4 inputs (2 for general mix, 2 for A3D) and 2 outputs (supposed to be routed to the codec). */ @@ -444,9 +444,9 @@ static void vortex_EqHw_ZeroState(vortex_t * vortex) vortex_EqHw_SetRightGainsTarget(vortex, eq_gains_zero); vortex_EqHw_SetBypassGain(vortex, 0, 0); - //vortex_EqHw_SetCurrBypassGain(vortex, 0, 0); + /* vortex_EqHw_SetCurrBypassGain(vortex, 0, 0); */ vortex_EqHw_SetA3DBypassGain(vortex, 0, 0); - //vortex_EqHw_SetCurrA3DBypassGain(vortex, 0, 0); + /* vortex_EqHw_SetCurrA3DBypassGain(vortex, 0, 0); */ vortex_EqHw_SetLeftStates(vortex, eq_states_zero, asEqOutStateZeros); vortex_EqHw_SetRightStates(vortex, eq_states_zero, asEqOutStateZeros); vortex_EqHw_SetLevels(vortex, (u16 *) eq_levels); @@ -570,7 +570,7 @@ vortex_Eqlzr_GetAllBands(vortex_t * vortex, u16 * gains, s32 *cnt) return 1; si++; } - while (eq->this10 > si) ; + while (eq->this10 > si); *cnt = si * 2; return 0; } @@ -597,7 +597,7 @@ vortex_Eqlzr_SetAllBands(vortex_t * vortex, u16 gains[], s32 count) for (i = 0; i < count; i++) { eq->this130[i] = gains[i]; } - + if (eq->this54) return 0; return vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex); @@ -643,7 +643,7 @@ static void vortex_Eqlzr_ShutDownA3d(vortex_t * vortex) static void vortex_Eqlzr_SetBypass(vortex_t * vortex, u32 bp) { eqlzr_t *eq = &(vortex->eq); - + if ((eq->this28) && (bp == 0)) { /* EQ enabled */ vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex); @@ -691,7 +691,7 @@ static void vortex_Eqlzr_init(vortex_t * vortex) eqlzr_t *eq = &(vortex->eq); /* Object constructor */ - //eq->this04 = 0; + /* eq->this04 = 0; */ eq->this08 = 0; /* Bypass gain with EQ in use. */ eq->this0a = 0x5999; eq->this0c = 0x5999; /* Bypass gain with EQ disabled. */ @@ -736,7 +736,7 @@ snd_vortex_eqtoggle_get(struct snd_kcontrol *kcontrol, { vortex_t *vortex = snd_kcontrol_chip(kcontrol); eqlzr_t *eq = &(vortex->eq); - //int i = kcontrol->private_value; + /* int i = kcontrol->private_value; */ ucontrol->value.integer.value[0] = eq->this54 ? 0 : 1; @@ -749,7 +749,7 @@ snd_vortex_eqtoggle_put(struct snd_kcontrol *kcontrol, { vortex_t *vortex = snd_kcontrol_chip(kcontrol); eqlzr_t *eq = &(vortex->eq); - //int i = kcontrol->private_value; + /* int i = kcontrol->private_value; */ eq->this54 = ucontrol->value.integer.value[0] ? 0 : 1; vortex_Eqlzr_SetBypass(vortex, eq->this54); @@ -900,7 +900,7 @@ static int __devinit vortex_eq_init(vortex_t * vortex) kcontrol->private_value = i; if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0) return err; - //vortex->eqctrl[i] = kcontrol; + /* vortex->eqctrl[i] = kcontrol; */ } /* EQ band levels */ if ((kcontrol = snd_ctl_new1(&vortex_levels_kcontrol, vortex)) == NULL) @@ -916,7 +916,7 @@ static int vortex_eq_free(vortex_t * vortex) /* //FIXME: segfault because vortex->eqctrl[i] == 4 int i; - for (i=0; i<10; i++) { + for (i = 0; i < 10; i++) { if (vortex->eqctrl[i]) snd_ctl_remove(vortex->card, vortex->eqctrl[i]); } diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c index c92f493..cd3daf8 100644 --- a/sound/pci/au88x0/au88x0_mixer.c +++ b/sound/pci/au88x0/au88x0_mixer.c @@ -23,7 +23,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex) if ((err = snd_ac97_bus(vortex->card, 0, &ops, NULL, &pbus)) < 0) return err; memset(&ac97, 0, sizeof(ac97)); - // Intialize AC97 codec stuff. + /* Intialize AC97 codec stuff. */ ac97.private_data = vortex; ac97.scaps = AC97_SCAP_NO_SPDIF; err = snd_ac97_mixer(pbus, &ac97, &vortex->codec); diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 0dc8d25..4d9154f 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -72,7 +72,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) /* Check if anything is OK. */ temp = hwread(vortex->mmio, VORTEX_MIDI_DATA); - if (temp != MPU401_ACK /*0xfe */ ) { + if (temp != MPU401_ACK /*0xfe */) { printk(KERN_ERR "midi port doesn't acknowledge!\n"); return -ENODEV; } diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b9d2f20..149769f 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -13,12 +13,12 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ - + /* * Vortex PCM ALSA driver. * * Supports ADB and WT DMA. Unfortunately, WT channels do not run yet. - * It remains stuck,and DMA transfers do not happen. + * It remains stuck,and DMA transfers do not happen. */ #include <sound/asoundef.h> #include <linux/time.h> @@ -103,7 +103,7 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_wt = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, // SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, /* SNDRV_PCM_RATE_48000, */ .rate_min = 8000, .rate_max = 48000, .channels_min = 1, @@ -121,7 +121,7 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) vortex_t *vortex = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; - + /* Force equal size periods */ if ((err = snd_pcm_hw_constraint_integer(runtime, @@ -170,10 +170,10 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) /* close callback */ static int snd_vortex_pcm_close(struct snd_pcm_substream *substream) { - //vortex_t *chip = snd_pcm_substream_chip(substream); + /* vortex_t *chip = snd_pcm_substream_chip(substream); */ stream_t *stream = (stream_t *) substream->runtime->private_data; - // the hardware-specific codes will be here + /* the hardware-specific codes will be here */ if (stream != NULL) { stream->substream = NULL; stream->nr_ch = 0; @@ -191,7 +191,7 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, stream_t *stream = (stream_t *) (substream->runtime->private_data); int err; - // Alloc buffer memory. + /* Alloc buffer memory. */ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) { @@ -203,7 +203,7 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, params_period_bytes(hw_params), params_channels(hw_params)); */ spin_lock_irq(&chip->lock); - // Make audio routes and config buffer DMA. + /* Make audio routes and config buffer DMA. */ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { int dma, type = VORTEX_PCM_TYPE(substream->pcm); /* Dealloc any routes. */ @@ -253,7 +253,7 @@ static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream) stream_t *stream = (stream_t *) (substream->runtime->private_data); spin_lock_irq(&chip->lock); - // Delete audio routes. + /* Delete audio routes. */ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { if (stream != NULL) vortex_adb_allocroute(chip, stream->dma, @@ -280,7 +280,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) stream_t *stream = (stream_t *) substream->runtime->private_data; int dma = stream->dma, fmt, dir; - // set up the hardware with the current configuration. + /* set up the hardware with the current configuration. */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dir = 1; else @@ -297,7 +297,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) #ifndef CHIP_AU8810 else { vortex_wtdma_setmode(chip, dma, 1, fmt, 0, 0); - // FIXME: Set rate (i guess using vortex_wt_writereg() somehow). + /* FIXME: Set rate (i guess using vortex_wt_writereg() somehow). */ vortex_wtdma_setstartbuffer(chip, dma, 0); } #endif @@ -315,8 +315,8 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - // do something to start the PCM engine - //printk(KERN_INFO "vortex: start %d\n", dma); + /* do something to start the PCM engine */ + /* printk(KERN_INFO "vortex: start %d\n", dma); */ stream->fifo_enabled = 1; if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { vortex_adbdma_resetup(chip, dma); @@ -330,12 +330,12 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) #endif break; case SNDRV_PCM_TRIGGER_STOP: - // do something to stop the PCM engine - //printk(KERN_INFO "vortex: stop %d\n", dma); + /* do something to stop the PCM engine */ + /* printk(KERN_INFO "vortex: stop %d\n", dma); */ stream->fifo_enabled = 0; if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) vortex_adbdma_pausefifo(chip, dma); - //vortex_adbdma_stopfifo(chip, dma); + /* vortex_adbdma_stopfifo(chip, dma); */ #ifndef CHIP_AU8810 else { printk(KERN_INFO "vortex: wt stop %d\n", dma); @@ -344,7 +344,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) #endif break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - //printk(KERN_INFO "vortex: pause %d\n", dma); + /* printk(KERN_INFO "vortex: pause %d\n", dma); */ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) vortex_adbdma_pausefifo(chip, dma); #ifndef CHIP_AU8810 @@ -353,7 +353,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) #endif break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - //printk(KERN_INFO "vortex: resume %d\n", dma); + /* printk(KERN_INFO "vortex: resume %d\n", dma); */ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) vortex_adbdma_resumefifo(chip, dma); #ifndef CHIP_AU8810 @@ -384,7 +384,7 @@ static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substr else current_ptr = vortex_wtdma_getlinearpos(chip, dma); #endif - //printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr); + /* printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr); */ spin_unlock(&chip->lock); return (bytes_to_frames(substream->runtime, current_ptr)); } @@ -498,7 +498,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) if (!chip || idx < 0 || idx >= VORTEX_PCM_LAST) return -ENODEV; - /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the + /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the * same dma engine. WT uses it own separate dma engine whcih cant capture. */ if (idx == VORTEX_PCM_ADB) nr_capt = nr; @@ -510,7 +510,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; strcpy(pcm->name, vortex_pcm_name[idx]); chip->pcm[idx] = pcm; - // This is an evil hack, but it saves a lot of duplicated code. + /* This is an evil hack, but it saves a lot of duplicated code. */ VORTEX_PCM_TYPE(pcm) = idx; pcm->private_data = chip; /* set operators */ @@ -519,9 +519,9 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) if (idx == VORTEX_PCM_ADB) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vortex_playback_ops); - + /* pre-allocation of Scatter-Gather buffers */ - + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci_dev), 0x10000, 0x10000); diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 978b856..1fcd430 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -41,10 +41,10 @@ static void vortex_wt_setstereo(vortex_t * vortex, u32 wt, u32 stereo) { int temp; - //temp = hwread(vortex->mmio, 0x80 + ((wt >> 0x5)<< 0xf) + (((wt & 0x1f) >> 1) << 2)); + /* temp = hwread(vortex->mmio, 0x80 + ((wt >> 0x5)<< 0xf) + (((wt & 0x1f) >> 1) << 2)); */ temp = hwread(vortex->mmio, WT_STEREO(wt)); temp = (temp & 0xfe) | (stereo & 1); - //hwwrite(vortex->mmio, 0x80 + ((wt >> 0x5)<< 0xf) + (((wt & 0x1f) >> 1) << 2), temp); + /* hwwrite(vortex->mmio, 0x80 + ((wt >> 0x5)<< 0xf) + (((wt & 0x1f) >> 1) << 2), temp); */ hwwrite(vortex->mmio, WT_STEREO(wt), temp); } @@ -68,22 +68,22 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch) wt_voice_t *voice = &(vortex->wt_voice[wt]); int temp; - //FIXME: WT audio routing. + /* FIXME: WT audio routing. */ if (nr_ch) { vortex_fifo_wtinitialize(vortex, wt, 1); vortex_fifo_setwtvalid(vortex, wt, 1); vortex_wt_setstereo(vortex, wt, nr_ch - 1); } else vortex_fifo_setwtvalid(vortex, wt, 0); - + /* Set mixdown mode. */ vortex_wt_setdsout(vortex, wt, 1); /* Set other parameter registers. */ hwwrite(vortex->mmio, WT_SRAMP(0), 0x880000); - //hwwrite(vortex->mmio, WT_GMODE(0), 0xffffffff); + /* hwwrite(vortex->mmio, WT_GMODE(0), 0xffffffff); */ #ifdef CHIP_AU8830 hwwrite(vortex->mmio, WT_SRAMP(1), 0x880000); - //hwwrite(vortex->mmio, WT_GMODE(1), 0xffffffff); + /* hwwrite(vortex->mmio, WT_GMODE(1), 0xffffffff); */ #endif hwwrite(vortex->mmio, WT_PARM(wt, 0), 0); hwwrite(vortex->mmio, WT_PARM(wt, 1), 0); @@ -91,7 +91,7 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch) temp = hwread(vortex->mmio, WT_PARM(wt, 3)); printk(KERN_DEBUG "vortex: WT PARM3: %x\n", temp); - //hwwrite(vortex->mmio, WT_PARM(wt, 3), temp); + /* hwwrite(vortex->mmio, WT_PARM(wt, 3), temp); */ hwwrite(vortex->mmio, WT_DELAY(wt, 0), 0); hwwrite(vortex->mmio, WT_DELAY(wt, 1), 0); @@ -151,7 +151,7 @@ static void vortex_wt_connect(vortex_t * vortex, int en) #if 0 static int vortex_wt_GetReg(vortex_t * vortex, char reg, int wt) { - //int eax, esi; + /* int eax, esi; */ if (reg == 4) { return hwread(vortex->mmio, WT_PARM(wt, 3)); @@ -175,7 +175,7 @@ vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt, return 0; if ((reg - 0x20) > 0) { - if ((reg - 0x21) != 0) + if ((reg - 0x21) != 0) return 0; eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x208; // param 2 } else { @@ -186,7 +186,7 @@ vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt, return 1; } -/*public: static void __thiscall CWTHal::SetReg(unsigned char,int,unsigned long) */ +/*public: static void __thiscall CWTHal::SetReg(unsigned char, int, unsigned long) */ #endif static int vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, @@ -213,38 +213,38 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, switch (reg) { /* Voice specific parameters */ case 0: /* running */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val); */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; break; case 1: /* param 0 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val); */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; break; case 2: /* param 1 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val); */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; break; case 3: /* param 2 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val); */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; break; case 4: /* param 3 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val); */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; break; case 6: /* mute */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val); */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; break; case 0xb: { /* delay */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val); */ hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); @@ -272,7 +272,7 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, return 0; break; } - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + /* printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); */ hwwrite(vortex->mmio, ecx, val); return 1; } @@ -289,7 +289,7 @@ static void vortex_wt_init(vortex_t * vortex) var10 &= 0xfffffffe; var10 &= 0xfffffbff; var10 |= 0x1800; - // var10 = 0x1AA2 + /* var10 = 0x1AA2 */ var4 = 0x10000000; varc = 0x00830000; var8 = 0x00830000; @@ -355,7 +355,7 @@ static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr) wt_voice_t *voice = &(vortex->wt_voice[wt]); u32 eax, edx; - //FIXME: 64 bit operation. + /* FIXME: 64 bit operation. */ eax = ((sr << 0xf) * 0x57619F1) & 0xffffffff; edx = (((sr << 0xf) * 0x57619F1)) >> 0x20; @@ -384,9 +384,9 @@ static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr) voice->parm0 &= 0xffff0001; voice->parm0 |= (eax & 0x7fff) << 1; voice->parm1 = voice->parm0 | 1; - // Wt: this_1D4 - //AuWt::WriteReg((ulong)(this_1DC<<4)+0x200, (ulong)this_1E4); - //AuWt::WriteReg((ulong)(this_1DC<<4)+0x204, (ulong)this_1E8); + /* Wt: this_1D4 */ + /* AuWt::WriteReg((ulong)(this_1DC<<4)+0x200, (ulong)this_1E4); */ + /* AuWt::WriteReg((ulong)(this_1DC<<4)+0x204, (ulong)this_1E8); */ hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0); hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1); } diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index b4151e2..c8acf62 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -64,7 +64,7 @@ static xtalk_gains_t const asXtalkGains1Chan = { static xtalk_gains_t const asXtalkGainsAllChan = { 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0, 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0 - //0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff + /* 0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff */ }; static xtalk_gains_t const asXtalkGainsZeros; @@ -672,7 +672,7 @@ static void vortex_XtalkHw_ZeroIO(vortex_t * vortex) static void vortex_XtalkHw_ZeroState(vortex_t * vortex) { - vortex_XtalkHw_ZeroIO(vortex); // inlined + vortex_XtalkHw_ZeroIO(vortex); /* inlined */ vortex_XtalkHw_SetLeftEQ(vortex, 0, 0, asXtalkCoefsZeros); vortex_XtalkHw_SetRightEQ(vortex, 0, 0, asXtalkCoefsZeros); @@ -680,14 +680,14 @@ static void vortex_XtalkHw_ZeroState(vortex_t * vortex) vortex_XtalkHw_SetLeftXT(vortex, 0, 0, asXtalkCoefsZeros); vortex_XtalkHw_SetRightXT(vortex, 0, 0, asXtalkCoefsZeros); - vortex_XtalkHw_SetGains(vortex, asXtalkGainsZeros); // inlined + vortex_XtalkHw_SetGains(vortex, asXtalkGainsZeros); /* inlined */ - vortex_XtalkHw_SetDelay(vortex, 0, 0); // inlined + vortex_XtalkHw_SetDelay(vortex, 0, 0); /* inlined */ - vortex_XtalkHw_SetLeftDline(vortex, alXtalkDlineZeros); // inlined - vortex_XtalkHw_SetRightDline(vortex, alXtalkDlineZeros); // inlined - vortex_XtalkHw_SetLeftDline(vortex, alXtalkDlineZeros); // inlined - vortex_XtalkHw_SetRightDline(vortex, alXtalkDlineZeros); // inlined + vortex_XtalkHw_SetLeftDline(vortex, alXtalkDlineZeros); /* inlined */ + vortex_XtalkHw_SetRightDline(vortex, alXtalkDlineZeros); /* inlined */ + vortex_XtalkHw_SetLeftDline(vortex, alXtalkDlineZeros); /* inlined */ + vortex_XtalkHw_SetRightDline(vortex, alXtalkDlineZeros); /* inlined */ vortex_XtalkHw_SetLeftEQStates(vortex, asXtalkInStateZeros, asXtalkOutStateZeros); @@ -707,7 +707,7 @@ static void vortex_XtalkHw_ProgramPipe(vortex_t * vortex) vortex_XtalkHw_SetLeftXT(vortex, 0, 0, asXtalkCoefsZeros); vortex_XtalkHw_SetRightXT(vortex, 0, 0, asXtalkCoefsZeros); - vortex_XtalkHw_SetDelay(vortex, 0, 0); // inlined + vortex_XtalkHw_SetDelay(vortex, 0, 0); /* inlined */ } static void vortex_XtalkHw_ProgramXtalkWide(vortex_t * vortex) @@ -724,7 +724,7 @@ static void vortex_XtalkHw_ProgramXtalkWide(vortex_t * vortex) sXtalkWideShiftLeftXt, asXtalkWideCoefsLeftXt); - vortex_XtalkHw_SetDelay(vortex, wXtalkWideRightDelay, wXtalkWideLeftDelay); // inlined + vortex_XtalkHw_SetDelay(vortex, wXtalkWideRightDelay, wXtalkWideLeftDelay); /* inlined */ } static void vortex_XtalkHw_ProgramXtalkNarrow(vortex_t * vortex) @@ -743,13 +743,13 @@ static void vortex_XtalkHw_ProgramXtalkNarrow(vortex_t * vortex) sXtalkNarrowShiftLeftXt, asXtalkNarrowCoefsLeftXt); - vortex_XtalkHw_SetDelay(vortex, wXtalkNarrowRightDelay, wXtalkNarrowLeftDelay); // inlined + vortex_XtalkHw_SetDelay(vortex, wXtalkNarrowRightDelay, wXtalkNarrowLeftDelay); /* inlined */ } static void vortex_XtalkHw_ProgramDiamondXtalk(vortex_t * vortex) { - //sDiamondKLeftEq,sDiamondKRightXt,asDiamondCoefsLeftEq + /* sDiamondKLeftEq,sDiamondKRightXt,asDiamondCoefsLeftEq */ vortex_XtalkHw_SetLeftEQ(vortex, sDiamondKLeftEq, sDiamondShiftLeftEq, asDiamondCoefsLeftEq); vortex_XtalkHw_SetRightEQ(vortex, sDiamondKRightEq, @@ -759,7 +759,7 @@ static void vortex_XtalkHw_ProgramDiamondXtalk(vortex_t * vortex) vortex_XtalkHw_SetRightXT(vortex, sDiamondKLeftXt, sDiamondShiftLeftXt, asDiamondCoefsLeftXt); - vortex_XtalkHw_SetDelay(vortex, wDiamondRightDelay, wDiamondLeftDelay); // inlined + vortex_XtalkHw_SetDelay(vortex, wDiamondRightDelay, wDiamondLeftDelay); /* inlined */ } static void vortex_XtalkHw_init(vortex_t * vortex) diff --git a/sound/pci/au88x0/au88x0_xtalk.h b/sound/pci/au88x0/au88x0_xtalk.h index 7f4534b..ead8453 100644 --- a/sound/pci/au88x0/au88x0_xtalk.h +++ b/sound/pci/au88x0/au88x0_xtalk.h @@ -22,7 +22,7 @@ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ -/* The crosstalk canceler supports 5 stereo input channels. The result is +/* The crosstalk canceler supports 5 stereo input channels. The result is available at one single output route pair (stereo). */ #ifndef _AU88X0_CXTALK_H

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[alsa-devel] [PATCH 0/5] ARM: OMAP2: Add support for GUmstix Overo (rev 3)
by sakoman＠gmail.com 09 Sep '08

09 Sep '08

This patch series adds support for the Gumstix Overo board. This is the third submission and incorporates all previous feedback from both linux-omap and alsa-devel. Careful readers will also notice that this version includes nand support rather than onenand.

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[alsa-devel] [PATCH] ice1712 drivers coding style fixes (part 2)
by Vedran Miletić 09 Sep '08

09 Sep '08

I started with Takashi's sound-unstable tree as a base: Fixed: * ice1724 compilation error (one { too much) * checkpatch errors along the files sound/pci/ice1712/ak4xxx.c total: 1 errors, 3 warnings, 0 checks, 196 lines checked sound/pci/ice1712/amp.h total: 0 errors, 0 warnings, 1 checks, 51 lines checked sound/pci/ice1712/aureon.c total: 0 errors, 121 warnings, 3 checks, 2284 lines checked sound/pci/ice1712/aureon.h total: 0 errors, 1 warnings, 0 checks, 65 lines checked sound/pci/ice1712/delta.c total: 17 errors, 44 warnings, 1 checks, 817 lines checked sound/pci/ice1712/delta.h total: 0 errors, 8 warnings, 0 checks, 153 lines checked sound/pci/ice1712/envy24ht.h total: 0 errors, 21 warnings, 0 checks, 221 lines checked sound/pci/ice1712/ews.c total: 27 errors, 64 warnings, 2 checks, 1087 lines checked sound/pci/ice1712/ews.h total: 0 errors, 7 warnings, 0 checks, 86 lines checked sound/pci/ice1712/hoontech.c total: 13 errors, 10 warnings, 1 checks, 360 lines checked sound/pci/ice1712/hoontech.h total: 2 errors, 7 warnings, 0 checks, 77 lines checked sound/pci/ice1712/ice1712.c total: 14 errors, 162 warnings, 3 checks, 2799 lines checked sound/pci/ice1712/ice1712.h total: 0 errors, 22 warnings, 3 checks, 511 lines checked sound/pci/ice1712/ice1724.c total: 11 errors, 62 warnings, 8 checks, 2657 lines checked sound/pci/ice1712/juli.c total: 0 errors, 0 warnings, 1 checks, 687 lines checked sound/pci/ice1712/maya44.c total: 6 errors, 0 warnings, 16 checks, 761 lines checked sound/pci/ice1712/pontis.c total: 0 errors, 40 warnings, 3 checks, 836 lines checked sound/pci/ice1712/prodigy192.c total: 12 errors, 21 warnings, 0 checks, 817 lines checked sound/pci/ice1712/prodigy_hifi.c total: 0 errors, 0 warnings, 1 checks, 1215 lines checked sound/pci/ice1712/snd-ice1712.mod.c total: 0 errors, 1 warnings, 0 checks, 113 lines checked sound/pci/ice1712/snd-ice1724.mod.c total: 0 errors, 1 warnings, 0 checks, 126 lines checked sound/pci/ice1712/vt1732.c total: 51 errors, 3 warnings, 5 checks, 2759 lines checked sound/pci/ice1712/vt1732.h total: 9 errors, 0 warnings, 2 checks, 452 lines checked It compiles. At first run ice1724 complained about something info-related (it's likely that it's not related to this patch at all) and generated like 20 or 30 lines of warnings and errors along the lines 890 - 900. On second run there was no errors or warnings anymore. -- Vedran Miletić

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Re: [alsa-devel] [PATCH 3/5] ALSA: ASoC: Add support for TWL4030 audio codec (rev 4)
by Steve Sakoman 09 Sep '08

09 Sep '08

From: Steve Sakoman <steve(a)sakoman.com> Signed-off-by: Steve Sakoman <steve(a)sakoman.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 653 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 197 +++++++++++++ 4 files changed, 857 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl4030.c create mode 100644 sound/soc/codecs/twl4030.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..2f00e1e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -50,3 +50,8 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_TWL4030 + tristate + depends on SND_SOC && TWL4030_CORE + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..a519ced 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-twl4030-objs := twl4030.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -23,3 +24,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c new file mode 100644 index 0000000..ee2f0d3 --- /dev/null +++ b/sound/soc/codecs/twl4030.c @@ -0,0 +1,653 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman, <steve(a)sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "twl4030.h" + +/* + * twl4030 register cache & default register settings + */ +static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { + 0x00, /* this register not used */ + 0x93, /* REG_CODEC_MODE (0x1) */ + 0xc3, /* REG_OPTION (0x2) */ + 0x00, /* REG_UNKNOWN (0x3) */ + 0x00, /* REG_MICBIAS_CTL (0x4) */ + 0x24, /* REG_ANAMICL (0x5) */ + 0x04, /* REG_ANAMICR (0x6) */ + 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x00, /* REG_ADCMICSEL (0x8) */ + 0x00, /* REG_DIGMIXING (0x9) */ + 0x0c, /* REG_ATXL1PGA (0xA) */ + 0x0c, /* REG_ATXR1PGA (0xB) */ + 0x00, /* REG_AVTXL2PGA (0xC) */ + 0x00, /* REG_AVTXR2PGA (0xD) */ + 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_VOICE_IF (0xF) */ + 0x00, /* REG_ARXR1PGA (0x10) */ + 0x00, /* REG_ARXL1PGA (0x11) */ + 0x6c, /* REG_ARXR2PGA (0x12) */ + 0x6c, /* REG_ARXL2PGA (0x13) */ + 0x00, /* REG_VRXPGA (0x14) */ + 0x00, /* REG_VSTPGA (0x15) */ + 0x00, /* REG_VRX2ARXPGA (0x16) */ + 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_ARX2VTXPGA (0x18) */ + 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ + 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ + 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x00, /* REG_ATX2ARXPGA (0x1D) */ + 0x00, /* REG_BT_IF (0x1E) */ + 0x00, /* REG_BTPGA (0x1F) */ + 0x00, /* REG_BTSTPGA (0x20) */ + 0x00, /* REG_EAR_CTL (0x21) */ + 0x24, /* REG_HS_SEL (0x22) */ + 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_POPN_SET (0x24) */ + 0x00, /* REG_PREDL_CTL (0x25) */ + 0x00, /* REG_PREDR_CTL (0x26) */ + 0x00, /* REG_PRECKL_CTL (0x27) */ + 0x00, /* REG_PRECKR_CTL (0x28) */ + 0x00, /* REG_HFL_CTL (0x29) */ + 0x00, /* REG_HFR_CTL (0x2A) */ + 0x00, /* REG_ALC_CTL (0x2B) */ + 0x00, /* REG_ALC_SET1 (0x2C) */ + 0x00, /* REG_ALC_SET2 (0x2D) */ + 0x00, /* REG_BOOST_CTL (0x2E) */ + 0x01, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_DTMF_FREQSEL (0x30) */ + 0x00, /* REG_DTMF_TONEXT1H (0x31) */ + 0x00, /* REG_DTMF_TONEXT1L (0x32) */ + 0x00, /* REG_DTMF_TONEXT2H (0x33) */ + 0x00, /* REG_DTMF_TONEXT2L (0x34) */ + 0x00, /* REG_DTMF_TONOFF (0x35) */ + 0x00, /* REG_DTMF_WANONOFF (0x36) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ + 0x16, /* REG_APLL_CTL (0x3A) */ + 0x00, /* REG_DTMF_CTL (0x3B) */ + 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ + 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ + 0x00, /* REG_MISC_SET_1 (0x3E) */ + 0x00, /* REG_PCMBTMUX (0x3F) */ + 0x00, /* not used (0x40) */ + 0x00, /* not used (0x41) */ + 0x00, /* not used (0x42) */ + 0x00, /* REG_RX_PATH_SEL (0x43) */ + 0x00, /* REG_VDL_APGA_CTL (0x44) */ + 0x00, /* REG_VIBRA_CTL (0x45) */ + 0x00, /* REG_VIBRA_SET (0x46) */ + 0x00, /* REG_VIBRA_PWM_SET (0x47) */ + 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x00, /* REG_MISC_SET_2 (0x49) */ +}; + +/* + * read twl4030 register cache + */ +static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + return cache[reg]; +} + +/* + * write twl4030 register cache + */ +static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL4030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl4030 register space + */ +static int twl4030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + twl4030_write_reg_cache(codec, reg, value); + return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl4030_clear_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode & ~TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_set_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode | TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_init_chip(struct snd_soc_codec *codec) +{ + int i; + + /* clear CODECPDZ prior to setting register defaults */ + twl4030_clear_codecpdz(codec); + + /* set all audio section registers to reasonable defaults */ + for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) + twl4030_write(codec, i, twl4030_reg[i]); + +} + +static const struct snd_kcontrol_new twl4030_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 127, 0), + SOC_DOUBLE_R("Capture Volume", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 127, 0), +}; + +/* add non dapm controls */ +static int twl4030_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&twl4030_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* outputs */ + {"OUTL", NULL, "DACL"}, + {"OUTR", NULL, "DACR"}, + + /* inputs */ + {"ADCL", NULL, "INL"}, + {"ADCR", NULL, "INR"}, +}; + +static int twl4030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets, + ARRAY_SIZE(twl4030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static void twl4030_power_up(struct snd_soc_codec *codec) +{ + u8 anamicl, regmisc1, byte, popn, hsgain; + int i = 0; + + /* set CODECPDZ to turn on codec */ + twl4030_set_codecpdz(codec); + + /* initiate offset cancellation */ + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_write(codec, TWL4030_REG_ANAMICL, + anamicl | TWL4030_CNCL_OFFSET_START); + + /* wait for offset cancellation to complete */ + do { + /* this takes a little while, so don't slam i2c */ + udelay(2000); + twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_ANAMICL); + } while ((i++ < 100) && + ((byte & TWL4030_CNCL_OFFSET_START) == + TWL4030_CNCL_OFFSET_START)); + + /* anti-pop when changing analog gain */ + regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); + + /* toggle CODECPDZ as per TRM */ + twl4030_clear_codecpdz(codec); + twl4030_set_codecpdz(codec); + + /* program anti-pop with bias ramp delay */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= TWL4030_RAMP_DELAY; + popn |= TWL4030_RAMP_DELAY_645MS; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + popn |= TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* enable output stage and gain setting */ + hsgain = TWL4030_HSR_GAIN_0DB | TWL4030_HSL_GAIN_0DB; + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); + + /* enable anti-pop ramp */ + popn |= TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); +} + +static void twl4030_power_down(struct snd_soc_codec *codec) +{ + u8 popn, hsgain; + + /* disable anti-pop ramp */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= ~TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* disable output stage and gain setting */ + hsgain = TWL4030_HSR_GAIN_PWR_DOWN | TWL4030_HSL_GAIN_PWR_DOWN; + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); + + /* disable bias out */ + popn &= ~TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* power down */ + twl4030_clear_codecpdz(codec); +} + +static int twl4030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + twl4030_power_up(codec); + break; + case SND_SOC_BIAS_PREPARE: + /* TODO: develop a twl4030_prepare function */ + break; + case SND_SOC_BIAS_STANDBY: + /* TODO: develop a twl4030_standby function */ + twl4030_power_down(codec); + break; + case SND_SOC_BIAS_OFF: + twl4030_power_down(codec); + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u8 mode, old_mode, format, old_format; + + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode & ~TWL4030_APLL_RATE; + + switch (params_rate(params)) { + case 8000: + mode |= TWL4030_APLL_RATE_8000; + break; + case 11025: + mode |= TWL4030_APLL_RATE_11025; + break; + case 12000: + mode |= TWL4030_APLL_RATE_12000; + break; + case 16000: + mode |= TWL4030_APLL_RATE_16000; + break; + case 22050: + mode |= TWL4030_APLL_RATE_22050; + break; + case 24000: + mode |= TWL4030_APLL_RATE_24000; + break; + case 32000: + mode |= TWL4030_APLL_RATE_32000; + break; + case 44100: + mode |= TWL4030_APLL_RATE_44100; + break; + case 48000: + mode |= TWL4030_APLL_RATE_48000; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_set_codecpdz(codec); + } + + /* sample size */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + format &= ~TWL4030_DATA_WIDTH; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format |= TWL4030_DATA_WIDTH_16S_16W; + break; + case SNDRV_PCM_FORMAT_S24_LE: + format |= TWL4030_DATA_WIDTH_32S_24W; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + params_format(params)); + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + return 0; +} + +static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 19200000: + infreq = TWL4030_APLL_INFREQ_19200KHZ; + break; + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + case 38400000: + infreq = TWL4030_APLL_INFREQ_38400KHZ; + break; + default: + printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + format &= ~(TWL4030_AIF_SLAVE_EN); + format |= TWL4030_CLK256FS_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + format &= ~(TWL4030_CLK256FS_EN); + format |= TWL4030_AIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + format &= ~TWL4030_AIF_FORMAT; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= TWL4030_AIF_FORMAT_CODEC; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + + return 0; +} + +#define TWL4030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_dai twl4030_dai = { + .name = "twl4030", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .ops = { + .hw_params = twl4030_hw_params, + }, + .dai_ops = { + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(twl4030_dai); + +static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl4030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl4030_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialize the driver + * register the mixer and dsp interfaces with the kernel + */ + +static int twl4030_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + printk(KERN_INFO "TWL4030 Audio Codec init \n"); + + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = &twl4030_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to create pcms\n"); + goto pcm_err; + } + + twl4030_init_chip(codec); + + /* power on device */ + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + twl4030_add_controls(codec); + twl4030_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *twl4030_socdev; + +static int twl4030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + twl4030_socdev = socdev; + twl4030_init(socdev); + + return 0; +} + +static int twl4030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_probe, + .remove = twl4030_remove, + .suspend = twl4030_suspend, + .resume = twl4030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + +MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); +MODULE_AUTHOR("Steve Sakoman"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h new file mode 100644 index 0000000..09865d9 --- /dev/null +++ b/sound/soc/codecs/twl4030.h @@ -0,0 +1,197 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman <steve(a)sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL4030_AUDIO_H__ +#define __TWL4030_AUDIO_H__ + +#define TWL4030_REG_CODEC_MODE 0x1 +#define TWL4030_REG_OPTION 0x2 +#define TWL4030_REG_UNKNOWN 0x3 +#define TWL4030_REG_MICBIAS_CTL 0x4 +#define TWL4030_REG_ANAMICL 0x5 +#define TWL4030_REG_ANAMICR 0x6 +#define TWL4030_REG_AVADC_CTL 0x7 +#define TWL4030_REG_ADCMICSEL 0x8 +#define TWL4030_REG_DIGMIXING 0x9 +#define TWL4030_REG_ATXL1PGA 0xA +#define TWL4030_REG_ATXR1PGA 0xB +#define TWL4030_REG_AVTXL2PGA 0xC +#define TWL4030_REG_AVTXR2PGA 0xD +#define TWL4030_REG_AUDIO_IF 0xE +#define TWL4030_REG_VOICE_IF 0xF +#define TWL4030_REG_ARXR1PGA 0x10 +#define TWL4030_REG_ARXL1PGA 0x11 +#define TWL4030_REG_ARXR2PGA 0x12 +#define TWL4030_REG_ARXL2PGA 0x13 +#define TWL4030_REG_VRXPGA 0x14 +#define TWL4030_REG_VSTPGA 0x15 +#define TWL4030_REG_VRX2ARXPGA 0x16 +#define TWL4030_REG_AVDAC_CTL 0x17 +#define TWL4030_REG_ARX2VTXPGA 0x18 +#define TWL4030_REG_ARXL1_APGA_CTL 0x19 +#define TWL4030_REG_ARXR1_APGA_CTL 0x1A +#define TWL4030_REG_ARXL2_APGA_CTL 0x1B +#define TWL4030_REG_ARXR2_APGA_CTL 0x1C +#define TWL4030_REG_ATX2ARXPGA 0x1D +#define TWL4030_REG_BT_IF 0x1E +#define TWL4030_REG_BTPGA 0x1F +#define TWL4030_REG_BTSTPGA 0x20 +#define TWL4030_REG_EAR_CTL 0x21 +#define TWL4030_REG_HS_SEL 0x22 +#define TWL4030_REG_HS_GAIN_SET 0x23 +#define TWL4030_REG_HS_POPN_SET 0x24 +#define TWL4030_REG_PREDL_CTL 0x25 +#define TWL4030_REG_PREDR_CTL 0x26 +#define TWL4030_REG_PRECKL_CTL 0x27 +#define TWL4030_REG_PRECKR_CTL 0x28 +#define TWL4030_REG_HFL_CTL 0x29 +#define TWL4030_REG_HFR_CTL 0x2A +#define TWL4030_REG_ALC_CTL 0x2B +#define TWL4030_REG_ALC_SET1 0x2C +#define TWL4030_REG_ALC_SET2 0x2D +#define TWL4030_REG_BOOST_CTL 0x2E +#define TWL4030_REG_SOFTVOL_CTL 0x2F +#define TWL4030_REG_DTMF_FREQSEL 0x30 +#define TWL4030_REG_DTMF_TONEXT1H 0x31 +#define TWL4030_REG_DTMF_TONEXT1L 0x32 +#define TWL4030_REG_DTMF_TONEXT2H 0x33 +#define TWL4030_REG_DTMF_TONEXT2L 0x34 +#define TWL4030_REG_DTMF_TONOFF 0x35 +#define TWL4030_REG_DTMF_WANONOFF 0x36 +#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 +#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 +#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 +#define TWL4030_REG_APLL_CTL 0x3A +#define TWL4030_REG_DTMF_CTL 0x3B +#define TWL4030_REG_DTMF_PGA_CTL2 0x3C +#define TWL4030_REG_DTMF_PGA_CTL1 0x3D +#define TWL4030_REG_MISC_SET_1 0x3E +#define TWL4030_REG_PCMBTMUX 0x3F +#define TWL4030_REG_RX_PATH_SEL 0x43 +#define TWL4030_REG_VDL_APGA_CTL 0x44 +#define TWL4030_REG_VIBRA_CTL 0x45 +#define TWL4030_REG_VIBRA_SET 0x46 +#define TWL4030_REG_VIBRA_PWM_SET 0x47 +#define TWL4030_REG_ANAMIC_GAIN 0x48 +#define TWL4030_REG_MISC_SET_2 0x49 + +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) + +/* Bitfield Definitions */ + +/* TWL4030_CODEC_MODE (0x01) Fields */ + +#define TWL4030_APLL_RATE 0xF0 +#define TWL4030_APLL_RATE_8000 0x00 +#define TWL4030_APLL_RATE_11025 0x10 +#define TWL4030_APLL_RATE_12000 0x20 +#define TWL4030_APLL_RATE_16000 0x40 +#define TWL4030_APLL_RATE_22050 0x50 +#define TWL4030_APLL_RATE_24000 0x60 +#define TWL4030_APLL_RATE_32000 0x80 +#define TWL4030_APLL_RATE_44100 0x90 +#define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_SEL_16K 0x04 +#define TWL4030_CODECPDZ 0x02 +#define TWL4030_OPT_MODE 0x01 + +/* ANAMICL (0x05) Fields */ +#define TWL4030_CNCL_OFFSET_START 0x80 +#define TWL4030_OFFSET_CNCL_SEL 0x60 +#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 +#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 +#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 +#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 +#define TWL4030_MICAMPL_EN 0x10 +#define TWL4030_CKMIC_EN 0x08 +#define TWL4030_AUXL_EN 0x04 +#define TWL4030_HSMIC_EN 0x02 +#define TWL4030_MAINMIC_EN 0x01 + +/* ANAMICR (0x06) Fields */ +#define TWL4030_MICAMPR_EN 0x10 +#define TWL4030_AUXR_EN 0x04 +#define TWL4030_SUBMIC_EN 0x01 + +/* AUDIO_IF (0x0E) Fields */ + +#define TWL4030_AIF_SLAVE_EN 0x80 +#define TWL4030_DATA_WIDTH 0x60 +#define TWL4030_DATA_WIDTH_16S_16W 0x00 +#define TWL4030_DATA_WIDTH_32S_16W 0x40 +#define TWL4030_DATA_WIDTH_32S_24W 0x60 +#define TWL4030_AIF_FORMAT 0x18 +#define TWL4030_AIF_FORMAT_CODEC 0x00 +#define TWL4030_AIF_FORMAT_LEFT 0x08 +#define TWL4030_AIF_FORMAT_RIGHT 0x10 +#define TWL4030_AIF_FORMAT_TDM 0x18 +#define TWL4030_AIF_TRI_EN 0x04 +#define TWL4030_CLK256FS_EN 0x02 +#define TWL4030_AIF_EN 0x01 + +/* HS_GAIN_SET (0x23) Fields */ + +#define TWL4030_HSR_GAIN 0x0C +#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 +#define TWL4030_HSR_GAIN_0DB 0x08 +#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C +#define TWL4030_HSL_GAIN 0x03 +#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 +#define TWL4030_HSL_GAIN_0DB 0x02 +#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 + +/* HS_POPN_SET (0x24) Fields */ + +#define TWL4030_VMID_EN 0x40 +#define TWL4030_EXTMUTE 0x20 +#define TWL4030_RAMP_DELAY 0x1C +#define TWL4030_RAMP_DELAY_20MS 0x00 +#define TWL4030_RAMP_DELAY_40MS 0x04 +#define TWL4030_RAMP_DELAY_81MS 0x08 +#define TWL4030_RAMP_DELAY_161MS 0x0C +#define TWL4030_RAMP_DELAY_323MS 0x10 +#define TWL4030_RAMP_DELAY_645MS 0x14 +#define TWL4030_RAMP_DELAY_1291MS 0x18 +#define TWL4030_RAMP_DELAY_2581MS 0x1C +#define TWL4030_RAMP_EN 0x02 + +/* APLL_CTL (0x3A) Fields */ + +#define TWL4030_APLL_EN 0x10 +#define TWL4030_APLL_INFREQ 0x0F +#define TWL4030_APLL_INFREQ_19200KHZ 0x05 +#define TWL4030_APLL_INFREQ_26000KHZ 0x06 +#define TWL4030_APLL_INFREQ_38400KHZ 0x0F + +/* REG_MISC_SET_1 (0x3E) Fields */ + +#define TWL4030_CLK64_EN 0x80 +#define TWL4030_SCRAMBLE_EN 0x40 +#define TWL4030_FMLOOP_EN 0x20 +#define TWL4030_SMOOTH_ANAVOL_EN 0x02 +#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 + +extern struct snd_soc_dai twl4030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl4030; + +#endif /* End of __TWL4030_AUDIO_H__ */ -- 1.5.4.5

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[alsa-devel] [RESEND, PATCH] Fix section for sa11xx-uda1341 platform driver
by Uwe Kleine-König 08 Sep '08

08 Sep '08

Don't use __init but __devinit to define probe function. A pointer to sa11xx_uda1341_probe is passed to the core via platform_driver_register and so the function must not disappear after the module is loaded. Using __init and having HOTPLUG=y and SND_SA11XX_UDA1341=m the following probably oopses: echo -n sa11xx_uda1341.1 > /sys/bus/platform/driver/sa11xx_uda1341/unbind echo -n sa11xx_uda1341.1 > /sys/bus/platform/driver/sa11xx_uda1341/bind Signed-off-by: Uwe Kleine-König <ukleinek(a)informatik.uni-freiburg.de> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Brian Avery <b.avery(a)hp.com> Cc: alsa-devel(a)alsa-project.org Cc: Andrew Morton <akpm(a)linux-foundation.org> --- Hello, uups, I forgot to add the right cc:. Sorry. @Andrew, I have a few more patches of this type pending. Is it OK for you if they go via you? This would ease the process a bit for me. Uwe sound/arm/sa11xx-uda1341.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index b9c51bf..da25b41 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -879,7 +879,7 @@ void snd_sa11xx_uda1341_free(struct snd_card *card) audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); } -static int __init sa11xx_uda1341_probe(struct platform_device *devptr) +static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) { int err; struct snd_card *card; -- tg: (7686ad5..) t/sectionfixes/sa11xx_uda1341 (depends on: t/sectionfixes/sa11xx_uda1341) -- Uwe Kleine-König http://www.google.com/search?q=1+electron+mass%3D

1 0

[alsa-devel] How to wait for end of callback?
by William Estrada 08 Sep '08

08 Sep '08

Hi group, What API call fo I use to wait for a callback to end? Right now I use a select() and wait for stdin input or a system interrupt. Is there another way? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

Re: [alsa-devel] [PATCH1/1] hda: digital slave support for IDT codecs
by Matthew Ranostay 08 Sep '08

08 Sep '08

Added slave_dig_outs entries for several IDT codecs that have multiple SPDIF outs, and enabled these SPDIF outs in several pin configs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ee24a0d..5280eff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -235,6 +235,10 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0f, 0x10, 0x11 }; +static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { + 0x26, 0, +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -284,6 +288,10 @@ static hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; +static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { + 0x1e, 0, +}; + static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x10, 0x40, }; @@ -317,6 +325,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; +static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { + 0x22, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -418,9 +430,10 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[10] = { +static hda_nid_t stac92hd71bxx_pin_nids[11] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x1f, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1492,22 +1505,22 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), }; -static unsigned int ref92hd71bxx_pin_configs[10] = { +static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[10] = { +static unsigned int dell_m4_1_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[10] = { +static unsigned int dell_m4_2_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -3846,6 +3859,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); spec->pin_nids = stac92hd73xx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, @@ -3984,6 +3998,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; spec->dmic_nids = stac92hd83xxx_dmic_nids; @@ -4134,6 +4149,7 @@ again: case 0x111d76b5: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -4166,6 +4182,7 @@ again: default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } spec->aloopback_mask = 0x20;

1 0

[alsa-devel] [PATCH1/1] hda: digital slave support for IDT codecs
by Matthew Ranostay 08 Sep '08

08 Sep '08

Added slave_dig_outs entries for several IDT codecs that have multiple SPDIF outs, and enabled these SPDIF outs in several pin configs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 9968ee4..52b3579 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -229,6 +229,10 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0f, 0x10, 0x11 }; +static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { + 0x26, 0, +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -274,6 +278,10 @@ static hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; +static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { + 0x1e, 0, +}; + static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x10, 0x40, }; @@ -303,6 +311,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; +static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { + 0x22, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -396,9 +408,10 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[10] = { +static hda_nid_t stac92hd71bxx_pin_nids[11] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x1f, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1424,22 +1437,22 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), }; -static unsigned int ref92hd71bxx_pin_configs[10] = { +static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[10] = { +static unsigned int dell_m4_1_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[10] = { +static unsigned int dell_m4_2_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -3841,8 +3854,9 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; + codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x22, 0x25); if (!err) { if (spec->board_config < 0) { @@ -3883,6 +3897,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; spec->dmic_nids = stac92hd83xxx_dmic_nids; @@ -4033,6 +4048,7 @@ again: case 0x111d76b5: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -4065,6 +4081,7 @@ again: default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } spec->aloopback_mask = 0x20;

3 8

[alsa-devel] [PATCH1/1] hda: digital slave support for IDT codecs
by Matthew Ranostay 08 Sep '08

08 Sep '08

Added slave_dig_outs entries for several IDT codecs that have multiple SPDIF outs, and enabled these SPDIF outs in several pin configs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ee24a0d..655a87c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -235,6 +235,10 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0f, 0x10, 0x11 }; +static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { + 0x26, 0, +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -284,6 +288,10 @@ static hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; +static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { + 0x1e, 0, +}; + static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x10, 0x40, }; @@ -317,6 +325,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; +static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { + 0x22, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -418,9 +430,10 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[10] = { +static hda_nid_t stac92hd71bxx_pin_nids[11] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x1f, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1492,22 +1505,22 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), }; -static unsigned int ref92hd71bxx_pin_configs[10] = { +static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[10] = { +static unsigned int dell_m4_1_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[10] = { +static unsigned int dell_m4_2_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -3984,6 +3997,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; spec->dmic_nids = stac92hd83xxx_dmic_nids; @@ -4134,6 +4148,7 @@ again: case 0x111d76b5: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -4166,6 +4181,7 @@ again: default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } spec->aloopback_mask = 0x20;

2 1

[alsa-devel] [PATCH] hda: SPDIF mux name change
by Matthew Ranostay 08 Sep '08

08 Sep '08

Changed the mux naming scheme from "IEC9258 Mux" to "IEC958 Playback Source" to match the coding style. --- Signed-by-off: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ee24a0d..173515d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1047,7 +1047,7 @@ static struct snd_kcontrol_new stac_dmux_mixer = { static struct snd_kcontrol_new stac_smux_mixer = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Mux", + .name = "IEC958 Playback Source", /* count set later */ .info = stac92xx_smux_enum_info, .get = stac92xx_smux_enum_get,

1 0

[alsa-devel] [PATCH] fix section mismatch in the sound_core
by Krzysztof Helt 08 Sep '08

08 Sep '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The __exit cleanup_oss_soundcore() is called from the __init init_soundcore(). This causes section mismatch and breaks kernel's linking on sparc64. Remove the __exit attribute from the cleanup_oss_soundcore(). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Patch against 2.6.27-rc5-mm1. diff -urp linux-ref/sound/sound_core.c linux-2.6.26/sound/sound_core.c --- linux-ref/sound/sound_core.c 2008-09-07 18:54:22.800008536 -0400 +++ linux-2.6.26/sound/sound_core.c 2008-09-07 18:55:56.805521447 -0400 @@ -13,7 +13,7 @@ #ifdef CONFIG_SOUND_OSS_CORE static int __init init_oss_soundcore(void); -static void __exit cleanup_oss_soundcore(void); +static void cleanup_oss_soundcore(void); #else static inline int init_oss_soundcore(void) { return 0; } static inline void cleanup_oss_soundcore(void) { } @@ -573,7 +573,7 @@ int soundcore_open(struct inode *inode, MODULE_ALIAS_CHARDEV_MAJOR(SOUND_MAJOR); -static void __exit cleanup_oss_soundcore(void) +static void cleanup_oss_soundcore(void) { /* We have nothing to really do here - we know the lists must be empty */ ---------------------------------------------------------------------- Tanie polaczenia z Polska i ze swiatem Sprawdz >> http://link.interia.pl/f1f00

2 1

[alsa-devel] [PATCH] hda: SPDIF mux controls
by Matthew Ranostay 08 Sep '08

08 Sep '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dynamically create mux controls for SPDIF outs on certain IDT/Sigmatel codecs. - --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 9968ee4..4637d76 100644 - --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -173,6 +173,9 @@ struct sigmatel_spec { unsigned int num_dmics; hda_nid_t *dmux_nids; unsigned int num_dmuxes; + hda_nid_t *smux_nids; + unsigned int num_smuxes; + hda_nid_t dig_in_nid; hda_nid_t mono_nid; hda_nid_t anabeep_nid; @@ -193,6 +196,8 @@ struct sigmatel_spec { unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; + struct hda_input_mux *sinput_mux; + unsigned int cur_smux[2]; unsigned int powerdown_adcs; /* i/o switches */ @@ -209,6 +214,7 @@ struct sigmatel_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; + struct hda_input_mux private_smux; struct hda_input_mux private_mono_mux; }; @@ -251,6 +257,10 @@ static hda_nid_t stac92hd73xx_dmux_nids[2] = { 0x20, 0x21, }; +static hda_nid_t stac92hd73xx_smux_nids[2] = { + 0x22, 0x23, +}; + #define STAC92HD83XXX_NUM_DMICS 2 static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { 0x11, 0x12, 0 @@ -294,6 +304,10 @@ static hda_nid_t stac92hd71bxx_dmux_nids[1] = { 0x1c, }; +static hda_nid_t stac92hd71bxx_smux_nids[2] = { + 0x24, 0x25, +}; + static hda_nid_t stac92hd71bxx_dac_nids[1] = { 0x10, /*0x11, */ }; @@ -340,6 +354,10 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_smux_nids[1] = { + 0x21, +}; + static hda_nid_t stac927x_dac_nids[6] = { 0x02, 0x03, 0x04, 0x05, 0x06, 0 }; @@ -365,6 +383,10 @@ static hda_nid_t stac9205_dmux_nids[1] = { 0x1d, }; +static hda_nid_t stac9205_smux_nids[1] = { + 0x21, +}; + #define STAC9205_NUM_DMICS 2 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 @@ -388,7 +410,7 @@ static hda_nid_t stac922x_pin_nids[10] = { static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, - - 0x14, 0x1e, 0x22 + 0x14, 0x22, 0x23 }; static hda_nid_t stac92hd83xxx_pin_nids[14] = { @@ -443,6 +465,36 @@ static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol, spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); } +static int stac92xx_smux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->sinput_mux, uinfo); +} + +static int stac92xx_smux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_smux[smux_idx]; + return 0; +} + +static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + return snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, + spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]); +} + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -993,6 +1045,15 @@ static struct snd_kcontrol_new stac_dmux_mixer = { .put = stac92xx_dmux_enum_put, }; +static struct snd_kcontrol_new stac_smux_mixer = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Mux", + /* count set later */ + .info = stac92xx_smux_enum_info, + .get = stac92xx_smux_enum_get, + .put = stac92xx_smux_enum_put, +}; + static const char *slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", @@ -1044,6 +1105,13 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (err < 0) return err; } + if (spec->num_smuxes > 0) { + stac_smux_mixer.count = spec->num_smuxes; + err = snd_ctl_add(codec->bus->card, + snd_ctl_new1(&stac_smux_mixer, codec)); + if (err < 0) + return err; + } if (spec->multiout.dig_out_nid) { err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); @@ -2811,6 +2879,34 @@ static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) return 0; }; +static const char *stac92xx_spdif_labels[3] = { + "Digital Playback", "Analog Mux 1", "Analog Mux 2" +}; + +static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *spdif_mux = &spec->private_smux; + int i, num_cons; + hda_nid_t con_lst[ARRAY_SIZE(stac92xx_spdif_labels)]; + + num_cons = snd_hda_get_connections(codec, + spec->smux_nids[0], + con_lst, + HDA_MAX_NUM_INPUTS); + if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_spdif_labels)) + return -EINVAL; + + for (i = 0; i < num_cons; i++) { + spdif_mux->items[spdif_mux->num_items].label = + stac92xx_spdif_labels[i]; + spdif_mux->items[spdif_mux->num_items].index = i; + spdif_mux->num_items++; + } + + return 0; +} + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3114,6 +3210,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } + if (spec->num_smuxes > 0) { + err = stac92xx_auto_create_spdif_mux_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2) @@ -3130,6 +3231,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->input_mux = &spec->private_imux; if (!spec->dinput_mux) spec->dinput_mux = &spec->private_dimux; + spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; return 1; @@ -3800,10 +3902,12 @@ again: spec->adc_nids = stac92hd73xx_adc_nids; spec->dmic_nids = stac92hd73xx_dmic_nids; spec->dmux_nids = stac92hd73xx_dmux_nids; + spec->smux_nids = stac92hd73xx_smux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); + spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; @@ -3842,7 +3946,7 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; - - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x25, 0x27); if (!err) { if (spec->board_config < 0) { @@ -4081,11 +4185,13 @@ again: spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; spec->dmux_nids = stac92hd71bxx_dmux_nids; + spec->smux_nids = stac92hd71bxx_smux_nids; spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->multiout.num_dacs = 1; @@ -4254,6 +4360,8 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->smux_nids = stac927x_smux_nids; + spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids); spec->dac_list = stac927x_dac_nids; spec->multiout.dac_nids = spec->dac_nids; @@ -4375,6 +4483,8 @@ static int patch_stac9205(struct hda_codec *codec) spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids); spec->mux_nids = stac9205_mux_nids; spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); + spec->smux_nids = stac9205_smux_nids; + spec->num_smuxes = ARRAY_SIZE(stac9205_smux_nids); spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; spec->dmux_nids = stac9205_dmux_nids; -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkjBgIYACgkQ7s2wy7nhBHWxowCgiQ5j+n6XCtwsqiIuEoJgONJW x5kAoKtAa9JjHRADbOyjsxDat20Orpf7 =UxWy -----END PGP SIGNATURE-----

2 3

[alsa-devel] [PATCH0/1] hda: slave digital out support
by Matthew Ranostay 08 Sep '08

08 Sep '08

Added support for playing a stream on multiple digital outs. This is done by defining codec->slave_dig_outs as array of hda_nid_t with a null-terminated entry to set the slave SPDIF outs, in which the slave outs have cloned settings of the master out (e.g. dig_out_nid). --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index 4f32911..696d77e 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -1454,12 +1454,22 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, codec->spdif_ctls = val; if (change) { + hda_nid_t *d; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); + + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); + } } mutex_unlock(&codec->spdif_mutex); @@ -1491,10 +1501,16 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, val |= AC_DIG1_ENABLE; change = codec->spdif_ctls != val; if (change) { + hda_nid_t *d; codec->spdif_ctls = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) @@ -1643,9 +1659,14 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, mutex_lock(&codec->spdif_mutex); change = codec->spdif_in_enable != val; if (change) { + hda_nid_t *d; codec->spdif_in_enable = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -2589,15 +2610,30 @@ int snd_hda_input_mux_put(struct hda_codec *codec, static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, unsigned int stream_tag, unsigned int format) { + hda_nid_t *d; + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); /* turn on again (if needed) */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + } + } /* @@ -2621,8 +2657,12 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int format, struct snd_pcm_substream *substream) { + hda_nid_t *nid; mutex_lock(&codec->spdif_mutex); setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format); + if (codec->slave_dig_outs) + for (nid = codec->slave_dig_outs; *nid; nid++) + setup_dig_out_stream(codec, *nid, stream_tag, format); mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2689,6 +2729,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream) { hda_nid_t *nids = mout->dac_nids; + hda_nid_t *d; int chs = substream->runtime->channels; int i; @@ -2702,9 +2743,16 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, mout->dig_out_used = HDA_DIG_ANALOG_DUP; setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + setup_dig_out_stream(codec, *d, + stream_tag, format); } else { mout->dig_out_used = 0; snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_cleanup_stream(codec, *d); } } mutex_unlock(&codec->spdif_mutex); diff --git a/pci/hda/hda_codec.h b/pci/hda/hda_codec.h index 780e2ff..60468f5 100644 --- a/pci/hda/hda_codec.h +++ b/pci/hda/hda_codec.h @@ -725,6 +725,7 @@ struct hda_codec { unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ + hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_hwdep *hwdep; /* assigned hwdep device */

2 1

[alsa-devel] [GIT PULL] unify two pxa sound drivers
by Dmitry Baryshkov 08 Sep '08

08 Sep '08

Hi, Please consider this changeset. It was partially tested, but additional testing will be appreciated. pxa3xx was only compile tested. The following changes since commit 6a55617ed5d1aa62b850de2cf66f5ede2eef4825: Linus Torvalds (1): Linux v2.6.27-rc4 are available in the git repository at: git://git.infradead.org/users/dbaryshkov/zaurus-2.6.git pxa2xx-asoc-cleanup Dmitry Baryshkov (4): Permit simultaneous compilation of both PXA AC97 drivers Separate common pxa2xx-ac97 code Make pxa-ac97-lib separate module Separate common pxa2xx-pcm code Russ Dill (1): pxa2xx-lib: support building for several pxa's arch/arm/mach-pxa/include/mach/pxa-regs.h | 4 +- include/sound/pxa2xx-lib.h | 45 +++++ sound/arm/Kconfig | 9 +- sound/arm/Makefile | 4 + sound/arm/pxa2xx-ac97-lib.c | 297 +++++++++++++++++++++++++++++ sound/arm/pxa2xx-ac97.c | 247 ++---------------------- sound/arm/pxa2xx-pcm-lib.c | 277 +++++++++++++++++++++++++++ sound/arm/pxa2xx-pcm.c | 252 +----------------------- sound/arm/pxa2xx-pcm.h | 13 +- sound/soc/pxa/Kconfig | 3 + sound/soc/pxa/pxa2xx-ac97.c | 272 +------------------------- sound/soc/pxa/pxa2xx-i2s.c | 26 ++- sound/soc/pxa/pxa2xx-pcm.c | 265 +------------------------- sound/soc/pxa/pxa2xx-pcm.h | 15 -- 14 files changed, 707 insertions(+), 1022 deletions(-) create mode 100644 include/sound/pxa2xx-lib.h create mode 100644 sound/arm/pxa2xx-ac97-lib.c create mode 100644 sound/arm/pxa2xx-pcm-lib.c -- With best wishes Dmitry

5 16

Re: [alsa-devel] Asoc and alsa utils regarded
by Liam Girdwood 08 Sep '08

08 Sep '08

On Mon, 2008-09-08 at 15:02 +0530, umeshyv wrote: > Hi Sir, > Thanks for your reply.I downloaded asoc patch for au1x but some files > are missing like board setup files and exm32.h,exm1200.h. au1x is in mainline now. There is probably no need for a patch. Please use an up to date kernel. > And where did you define the frequency controller details.Can I get > the complete patch of au1x.I dont have a exact patch recently I used > another patch there psc-i2s.c is modified a bit little.I am working on > Au1200 to it tsc2301 is interfaced and Au1200 acts as master and MCLK > is generated using Au1200 frequency controller,my question is where I > am supposed define it. You will need to write a small machine driver to define all your board/machine specific details. There are quite a few examples under sound/soc/ > One more query is that for example consider the following code, > > SOC_ENUM_SINGLE(CS4251X_VOLTRANSCTL, 4, 4,cs_vol_transition) > > SOC_DOUBLE_R("Channel 4 Playback Volume", CS4251X_L4VOL, CS4251X_R4VOL, 0, 255, 1). > > > It tries to control volume transition and secondly volume > adjustment .Generally register access to codec is done using SPI or > I2C .So special functions like cs4251x_write(codec,<reg>,<value>) been > developed for register access.When using SOC_DOUBLE_R or SOC_SINGLE > etc how it access special functions to write to those specific codec > registers can get explanation. They ultimately call either i2C_write or spi_write. Please read the documentation in Documentation/sound/alsa/soc/ > Can you provide me Asoc driver for DbAu1200 board.And following is the > way i compiled libs and utils and I use buildroot to compile the libs > and utils > > Libs compilation > ------------------------------- > $PATH to buildroot > $./configure --host=mipsel-linux --prefix=/home/audio > --enable-shared=no --enable-static=yes > $make > $make install > > Utils Compilation > --------------------------------- > $export ALSA_CFLAGS=-I/home/audio/include > $export ALSA_LIBS="-L/home/audio/lib -lasound -lm -ldl -lpthread" > $./configure --host=mipsel-linux --prefix=/home/audio > --with-alsa-prefix=/home/audio/lib > --with-alsa-inc-prefix=/home/audio/include > $make > $make install > > All my installed libs and utils goes to /home/audio .Is there anything > missing in configurtion and after compilation I copy /bin,/sbin,/lib > to rootfs and then /share to /usr/share.I am going wrong > anywhere.Waiting for your valuable reply. This looks like it may be either a buildroot problem or you are missing some other buildroot alsa packages containing the conf files. Liam

1 0

[alsa-devel] Where can I find some info about the DAI?
by Sedji Gaouaou 08 Sep '08

08 Sep '08

Hi, I am trying to write a DAI. I am working with a wolfson wm8731 and an atmel at91sam9g20. Basically I had a look at the file eti_b1_wm8731.c, but I would like to use another ssc archi(atmel_ssc instead of at91_ssc). So I would like to know where should I find some info concerning the DAI. Regards, Sedji

3 4

[alsa-devel] ASoC vs non-ASoC: 2 pcm driver questions.
by Dmitry Baryshkov 08 Sep '08

08 Sep '08

Hi, While unifying pxa2xx-pcm drivers (ASoC and plain one) I've encountered the following differences in snd_pcm_open: 1. ASoC one uses snd_soc_set_runtime_hardware() function instead of just copying runtime->hw = pxa2xx_pcm_hardware; However the major difference is that snd_soc_set_runtime_hardware() doesn't initialise few fields which are set later in generic soc_pcm_open(). Can I drop this call and use the same assignment as in non-asoc driver. 2. To enforce some constraints ASoC driver uses generic snd_pcm_hw_constraint_step() instead of hand-written rule functions. Can this be also unified or there are some hidden differencies? -- With best wishes Dmitry

2 1

[alsa-devel] Asoc and alsa utils regarded
by umeshyv 08 Sep '08

08 Sep '08

Hi to all, Currently working on Asoc for Au1200.I need to know whether the framework is completed since I see some of the files missing in the all versions of the kernel and moreover I am trying to compile alsa libs and utils ,it compiled successfully but when I run at the target sh-3.2# aplay -c1 -r8000 -fs8 Infected.wav ALSA lib conf.c:3009:(snd_config_update_r) Cannot access file /home/alsa/install/share/alsa/alsa.conf ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:583: audio open error: No such file or directory Do I need to export any PATH at the target.Can anyone guide me. Thanks & Regards Umesh

2 1

[alsa-devel] Asoc and alsa utils regarded
by umeshyv 08 Sep '08

08 Sep '08

Hi to all, Currently working on Asoc for Au1200.I need to know whether the framework is completed since I see some of the files missing in the all versions of the kernel and moreover I am trying to compile alsa libs and utils ,it compiled successfully but when I run at the target sh-3.2# aplay -c1 -r8000 -fs8 Infected.wav ALSA lib conf.c:3009:(snd_config_update_r) Cannot access file /home/alsa/install/share/alsa/alsa.conf ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:583: audio open error: No such file or directory Do I need to export any PATH at the target.Can anyone guide me. Thanks & Regards Umesh

1 0

[alsa-devel] [PATCH] ice1712.c: fix coding style
by Alexander Beregalov 07 Sep '08

07 Sep '08

before: total: 113 errors, 169 warnings, 2786 lines checked total: 26 errors, 24 warnings, 504 lines checked after: total: 14 errors, 163 warnings, 2799 lines checked total: 0 errors, 24 warnings, 504 lines checked Compile tested only. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 237 +++++++++++++++++++++++-------------------- sound/pci/ice1712/ice1712.h | 52 +++++----- 2 files changed, 151 insertions(+), 138 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 05ffab6..5b44238 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -17,7 +17,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ /* NOTES: @@ -35,7 +35,7 @@ * * 2002.11.26 James Stafford <jstafford(a)ampltd.com> * Added support for VT1724 (Envy24HT) - * I have left out support for 176.4 and 192 KHz for the moment. + * I have left out support for 176.4 and 192 KHz for the moment. * I also haven't done anything with the internal S/PDIF transmitter or the MPU-401 * * 2003.02.20 Taksahi Iwai <tiwai(a)suse.de> @@ -47,7 +47,7 @@ */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -123,7 +123,7 @@ static unsigned int PRO_RATE_DEFAULT = 44100; /* * Basic I/O */ - + /* check whether the clock mode is spdif-in */ static inline int is_spdif_master(struct snd_ice1712 *ice) { @@ -135,13 +135,13 @@ static inline int is_pro_rate_locked(struct snd_ice1712 *ice) return is_spdif_master(ice) || PRO_RATE_LOCKED; } -static inline void snd_ice1712_ds_write(struct snd_ice1712 * ice, u8 channel, u8 addr, u32 data) +static inline void snd_ice1712_ds_write(struct snd_ice1712 *ice, u8 channel, u8 addr, u32 data) { outb((channel << 4) | addr, ICEDS(ice, INDEX)); outl(data, ICEDS(ice, DATA)); } -static inline u32 snd_ice1712_ds_read(struct snd_ice1712 * ice, u8 channel, u8 addr) +static inline u32 snd_ice1712_ds_read(struct snd_ice1712 *ice, u8 channel, u8 addr) { outb((channel << 4) | addr, ICEDS(ice, INDEX)); return inl(ICEDS(ice, DATA)); @@ -260,7 +260,7 @@ static unsigned short snd_ice1712_pro_ac97_read(struct snd_ac97 *ac97, static int snd_ice1712_digmix_route_ac97_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = inb(ICEMT(ice, MONITOR_ROUTECTRL)) & ICE1712_ROUTE_AC97 ? 1 : 0; return 0; } @@ -269,11 +269,12 @@ static int snd_ice1712_digmix_route_ac97_put(struct snd_kcontrol *kcontrol, stru { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char val, nval; - + spin_lock_irq(&ice->reg_lock); val = inb(ICEMT(ice, MONITOR_ROUTECTRL)); nval = val & ~ICE1712_ROUTE_AC97; - if (ucontrol->value.integer.value[0]) nval |= ICE1712_ROUTE_AC97; + if (ucontrol->value.integer.value[0]) + nval |= ICE1712_ROUTE_AC97; outb(nval, ICEMT(ice, MONITOR_ROUTECTRL)); spin_unlock_irq(&ice->reg_lock); return val != nval; @@ -329,7 +330,7 @@ static int snd_ice1712_cs8427_set_input_clock(struct snd_ice1712 *ice, int spdif unsigned char reg[2] = { 0x80 | 4, 0 }; /* CS8427 auto increment | register number 4 + data */ unsigned char val, nval; int res = 0; - + snd_i2c_lock(ice->i2c); if (snd_i2c_sendbytes(ice->cs8427, reg, 1) != 1) { snd_i2c_unlock(ice->i2c); @@ -381,9 +382,9 @@ int __devinit snd_ice1712_init_cs8427(struct snd_ice1712 *ice, int addr) { int err; - if ((err = snd_cs8427_create(ice->i2c, addr, - (ice->cs8427_timeout * HZ) / 1000, - &ice->cs8427)) < 0) { + err = snd_cs8427_create(ice->i2c, addr, + (ice->cs8427_timeout * HZ) / 1000, &ice->cs8427); + if (err < 0) { snd_printk(KERN_ERR "CS8427 initialization failed\n"); return err; } @@ -395,9 +396,9 @@ int __devinit snd_ice1712_init_cs8427(struct snd_ice1712 *ice, int addr) static void snd_ice1712_set_input_clock_source(struct snd_ice1712 *ice, int spdif_is_master) { - /* change CS8427 clock source too */ - if (ice->cs8427) - snd_ice1712_cs8427_set_input_clock(ice, spdif_is_master); + /* change CS8427 clock source too */ + if (ice->cs8427) + snd_ice1712_cs8427_set_input_clock(ice, spdif_is_master); /* notify ak4524 chip as well */ if (spdif_is_master) { unsigned int i; @@ -457,11 +458,12 @@ static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id) u16 pbkstatus; struct snd_pcm_substream *substream; pbkstatus = inw(ICEDS(ice, INTSTAT)); - //printk("pbkstatus = 0x%x\n", pbkstatus); + /* printk("pbkstatus = 0x%x\n", pbkstatus); */ for (idx = 0; idx < 6; idx++) { if ((pbkstatus & (3 << (idx * 2))) == 0) continue; - if ((substream = ice->playback_con_substream_ds[idx]) != NULL) + substream = ice->playback_con_substream_ds[idx]; + if (substream != NULL) snd_pcm_period_elapsed(substream); outw(3 << (idx * 2), ICEDS(ice, INTSTAT)); } @@ -507,7 +509,7 @@ static int snd_ice1712_playback_trigger(struct snd_pcm_substream *substream, struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); int result = 0; u32 tmp; - + spin_lock(&ice->reg_lock); tmp = snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL); if (cmd == SNDRV_PCM_TRIGGER_START) { @@ -532,7 +534,7 @@ static int snd_ice1712_playback_ds_trigger(struct snd_pcm_substream *substream, struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); int result = 0; u32 tmp; - + spin_lock(&ice->reg_lock); tmp = snd_ice1712_ds_read(ice, substream->number * 2, ICE1712_DSC_CONTROL); if (cmd == SNDRV_PCM_TRIGGER_START) { @@ -557,7 +559,7 @@ static int snd_ice1712_capture_trigger(struct snd_pcm_substream *substream, struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); int result = 0; u8 tmp; - + spin_lock(&ice->reg_lock); tmp = snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL); if (cmd == SNDRV_PCM_TRIGGER_START) { @@ -711,8 +713,7 @@ static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *s return bytes_to_frames(substream->runtime, ptr); } -static const struct snd_pcm_hardware snd_ice1712_playback = -{ +static const struct snd_pcm_hardware snd_ice1712_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -731,8 +732,7 @@ static const struct snd_pcm_hardware snd_ice1712_playback = .fifo_size = 0, }; -static const struct snd_pcm_hardware snd_ice1712_playback_ds = -{ +static const struct snd_pcm_hardware snd_ice1712_playback_ds = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -751,8 +751,7 @@ static const struct snd_pcm_hardware snd_ice1712_playback_ds = .fifo_size = 0, }; -static const struct snd_pcm_hardware snd_ice1712_capture = -{ +static const struct snd_pcm_hardware snd_ice1712_capture = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), @@ -788,7 +787,7 @@ static int snd_ice1712_playback_ds_open(struct snd_pcm_substream *substream) ice->playback_con_substream_ds[substream->number] = substream; runtime->hw = snd_ice1712_playback_ds; - spin_lock_irq(&ice->reg_lock); + spin_lock_irq(&ice->reg_lock); tmp = inw(ICEDS(ice, INTMASK)) & ~(1 << (substream->number * 2)); outw(tmp, ICEDS(ice, INTMASK)); spin_unlock_irq(&ice->reg_lock); @@ -821,7 +820,7 @@ static int snd_ice1712_playback_ds_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); u32 tmp; - spin_lock_irq(&ice->reg_lock); + spin_lock_irq(&ice->reg_lock); tmp = inw(ICEDS(ice, INTMASK)) | (3 << (substream->number * 2)); outw(tmp, ICEDS(ice, INTMASK)); spin_unlock_irq(&ice->reg_lock); @@ -870,7 +869,7 @@ static struct snd_pcm_ops snd_ice1712_capture_ops = { .pointer = snd_ice1712_capture_pointer, }; -static int __devinit snd_ice1712_pcm(struct snd_ice1712 * ice, int device, struct snd_pcm ** rpcm) +static int __devinit snd_ice1712_pcm(struct snd_ice1712 *ice, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -900,7 +899,7 @@ static int __devinit snd_ice1712_pcm(struct snd_ice1712 * ice, int device, struc return 0; } -static int __devinit snd_ice1712_pcm_ds(struct snd_ice1712 * ice, int device, struct snd_pcm ** rpcm) +static int __devinit snd_ice1712_pcm_ds(struct snd_ice1712 *ice, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -1029,14 +1028,14 @@ static void snd_ice1712_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, if (inb(ICEMT(ice, PLAYBACK_CONTROL)) & (ICE1712_CAPTURE_START_SHADOW| ICE1712_PLAYBACK_PAUSE| ICE1712_PLAYBACK_START)) { - __out: +__out: spin_unlock_irqrestore(&ice->reg_lock, flags); return; } if (!force && is_pro_rate_locked(ice)) goto __out; - old = inb(ICEMT(ice, RATE)); + old = inb(ICEMT(ice, RATE)); if (!force && old == val) goto __out; outb(val, ICEMT(ice, RATE)); @@ -1123,8 +1122,7 @@ static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substrea return bytes_to_frames(substream->runtime, ptr); } -static const struct snd_pcm_hardware snd_ice1712_playback_pro = -{ +static const struct snd_pcm_hardware snd_ice1712_playback_pro = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -1143,8 +1141,7 @@ static const struct snd_pcm_hardware snd_ice1712_playback_pro = .fifo_size = 0, }; -static const struct snd_pcm_hardware snd_ice1712_capture_pro = -{ +static const struct snd_pcm_hardware snd_ice1712_capture_pro = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -1238,7 +1235,7 @@ static struct snd_pcm_ops snd_ice1712_capture_pro_ops = { .pointer = snd_ice1712_capture_pro_pointer, }; -static int __devinit snd_ice1712_pcm_profi(struct snd_ice1712 * ice, int device, struct snd_pcm ** rpcm) +static int __devinit snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -1262,7 +1259,7 @@ static int __devinit snd_ice1712_pcm_profi(struct snd_ice1712 * ice, int device, ice->pcm_pro = pcm; if (rpcm) *rpcm = pcm; - + if (ice->cs8427) { /* assign channels to iec958 */ err = snd_cs8427_iec958_build(ice->cs8427, @@ -1272,7 +1269,8 @@ static int __devinit snd_ice1712_pcm_profi(struct snd_ice1712 * ice, int device, return err; } - if ((err = snd_ice1712_build_pro_mixer(ice)) < 0) + err = snd_ice1712_build_pro_mixer(ice); + if (err < 0) return err; return 0; } @@ -1299,7 +1297,7 @@ static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struc struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; - + spin_lock_irq(&ice->reg_lock); ucontrol->value.integer.value[0] = !((ice->pro_volumes[priv_idx] >> 15) & 1); @@ -1341,7 +1339,7 @@ static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struc struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; - + spin_lock_irq(&ice->reg_lock); ucontrol->value.integer.value[0] = (ice->pro_volumes[priv_idx] >> 0) & 127; @@ -1406,7 +1404,7 @@ static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch __devinit static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("Multi ",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, SWITCH), .info = snd_ice1712_pro_mixer_switch_info, .get = snd_ice1712_pro_mixer_switch_get, .put = snd_ice1712_pro_mixer_switch_put, @@ -1428,7 +1426,7 @@ static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume __devinit static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("Multi ",CAPTURE,VOLUME), + .name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, VOLUME), .info = snd_ice1712_pro_mixer_volume_info, .get = snd_ice1712_pro_mixer_volume_get, .put = snd_ice1712_pro_mixer_volume_put, @@ -1448,7 +1446,7 @@ static int __devinit snd_ice1712_build_pro_mixer(struct snd_ice1712 *ice) if (err < 0) return err; } - + if (ice->num_total_adcs > 0) { struct snd_kcontrol_new tmp = snd_ice1712_multi_capture_analog_switch; tmp.count = ice->num_total_adcs; @@ -1495,7 +1493,7 @@ static void snd_ice1712_mixer_free_ac97(struct snd_ac97 *ac97) ice->ac97 = NULL; } -static int __devinit snd_ice1712_ac97_mixer(struct snd_ice1712 * ice) +static int __devinit snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) { int err, bus_num = 0; struct snd_ac97_template ac97; @@ -1510,27 +1508,32 @@ static int __devinit snd_ice1712_ac97_mixer(struct snd_ice1712 * ice) }; if (ice_has_con_ac97(ice)) { - if ((err = snd_ac97_bus(ice->card, bus_num++, &con_ops, NULL, &pbus)) < 0) + err = snd_ac97_bus(ice->card, bus_num++, &con_ops, NULL, &pbus); + if (err < 0) return err; memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ice; ac97.private_free = snd_ice1712_mixer_free_ac97; - if ((err = snd_ac97_mixer(pbus, &ac97, &ice->ac97)) < 0) + err = snd_ac97_mixer(pbus, &ac97, &ice->ac97); + if (err < 0) printk(KERN_WARNING "ice1712: cannot initialize ac97 for consumer, skipped\n"); else { - if ((err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice))) < 0) + err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); + if (err < 0) return err; return 0; } } - if (! (ice->eeprom.data[ICE_EEP1_ACLINK] & ICE1712_CFG_PRO_I2S)) { - if ((err = snd_ac97_bus(ice->card, bus_num, &pro_ops, NULL, &pbus)) < 0) + if (!(ice->eeprom.data[ICE_EEP1_ACLINK] & ICE1712_CFG_PRO_I2S)) { + err = snd_ac97_bus(ice->card, bus_num, &pro_ops, NULL, &pbus); + if (err < 0) return err; memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ice; ac97.private_free = snd_ice1712_mixer_free_ac97; - if ((err = snd_ac97_mixer(pbus, &ac97, &ice->ac97)) < 0) + err = snd_ac97_mixer(pbus, &ac97, &ice->ac97); + if (err < 0) printk(KERN_WARNING "ice1712: cannot initialize pro ac97, skipped\n"); else return 0; @@ -1549,7 +1552,7 @@ static inline unsigned int eeprom_double(struct snd_ice1712 *ice, int idx) return (unsigned int)ice->eeprom.data[idx] | ((unsigned int)ice->eeprom.data[idx + 1] << 8); } -static void snd_ice1712_proc_read(struct snd_info_entry *entry, +static void snd_ice1712_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ice1712 *ice = entry->private_data; @@ -1585,15 +1588,15 @@ static void snd_ice1712_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, " SPDOUT : 0x%04x\n", (unsigned)inw(ICEMT(ice, ROUTE_SPDOUT))); snd_iprintf(buffer, " RATE : 0x%02x\n", (unsigned)inb(ICEMT(ice, RATE))); snd_iprintf(buffer, " GPIO_DATA : 0x%02x\n", (unsigned)snd_ice1712_get_gpio_data(ice)); - snd_iprintf(buffer, " GPIO_WRITE_MASK : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_WRITE_MASK)); + snd_iprintf(buffer, " GPIO_WRITE_MASK : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_WRITE_MASK)); snd_iprintf(buffer, " GPIO_DIRECTION : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_DIRECTION)); } -static void __devinit snd_ice1712_proc_init(struct snd_ice1712 * ice) +static void __devinit snd_ice1712_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "ice1712", &entry)) + if (!snd_card_proc_new(ice->card, "ice1712", &entry)) snd_info_set_text_ops(entry, ice, snd_ice1712_proc_read); } @@ -1613,7 +1616,7 @@ static int snd_ice1712_eeprom_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + memcpy(ucontrol->value.bytes.data, &ice->eeprom, sizeof(ice->eeprom)); return 0; } @@ -1641,7 +1644,7 @@ static int snd_ice1712_spdif_default_get(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); if (ice->spdif.ops.default_get) - ice->spdif.ops.default_get(ice, ucontrol); + ice->spdif.ops.default_get(ice, ucontrol); return 0; } @@ -1657,7 +1660,7 @@ static int snd_ice1712_spdif_default_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_ice1712_spdif_default __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_ice1712_spdif_info, .get = snd_ice1712_spdif_default_get, .put = snd_ice1712_spdif_default_put @@ -1709,7 +1712,7 @@ static struct snd_kcontrol_new snd_ice1712_spdif_maskc __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_ice1712_spdif_info, .get = snd_ice1712_spdif_maskc_get, }; @@ -1718,7 +1721,7 @@ static struct snd_kcontrol_new snd_ice1712_spdif_maskp __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_ice1712_spdif_info, .get = snd_ice1712_spdif_maskp_get, }; @@ -1746,7 +1749,7 @@ static struct snd_kcontrol_new snd_ice1712_spdif_stream __devinitdata = .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE), .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM), .info = snd_ice1712_spdif_info, .get = snd_ice1712_spdif_stream_get, .put = snd_ice1712_spdif_stream_put @@ -1758,7 +1761,7 @@ int snd_ice1712_gpio_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char mask = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value & (1<<24)) ? 1 : 0; - + snd_ice1712_save_gpio_status(ice); ucontrol->value.integer.value[0] = (snd_ice1712_gpio_read(ice) & mask ? 1 : 0) ^ invert; @@ -1825,7 +1828,7 @@ static int snd_ice1712_pro_internal_clock_get(struct snd_kcontrol *kcontrol, 9, 6, 3, 1, 7, 4, 0, 12, 8, 5, 2, 11, 255, 255, 255, 10 }; unsigned char val; - + spin_lock_irq(&ice->reg_lock); if (is_spdif_master(ice)) { ucontrol->value.enumerated.item[0] = 13; @@ -1867,7 +1870,7 @@ static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol, if ((oval & ICE1712_SPDIF_MASTER) != (inb(ICEMT(ice, RATE)) & ICE1712_SPDIF_MASTER)) - snd_ice1712_set_input_clock_source(ice, is_spdif_master(ice)); + snd_ice1712_set_input_clock_source(ice, is_spdif_master(ice)); return change; } @@ -1897,7 +1900,7 @@ static int snd_ice1712_pro_internal_clock_default_info(struct snd_kcontrol *kcon "64000", /* 10: 15 */ "88200", /* 11: 11 */ "96000", /* 12: 7 */ - // "IEC958 Input", /* 13: -- */ + /* "IEC958 Input", 13: -- */ }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2026,7 +2029,7 @@ static int snd_ice1712_pro_route_info(struct snd_kcontrol *kcontrol, "IEC958 In L", "IEC958 In R", /* 9-10 */ "Digital Mixer", /* 11 - optional */ }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = @@ -2070,7 +2073,7 @@ static int snd_ice1712_pro_route_analog_put(struct snd_kcontrol *kcontrol, int change, shift; int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int val, old_val, nval; - + /* update PSDOUT */ if (ucontrol->value.enumerated.item[0] >= 11) nval = idx < 2 ? 1 : 0; /* dig mixer (or pcm) */ @@ -2140,7 +2143,7 @@ static int snd_ice1712_pro_route_spdif_put(struct snd_kcontrol *kcontrol, int change, shift; int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int val, old_val, nval; - + /* update SPDOUT */ spin_lock_irq(&ice->reg_lock); val = old_val = inw(ICEMT(ice, ROUTE_SPDOUT)); @@ -2182,7 +2185,7 @@ static struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route __devinitdata static struct snd_kcontrol_new snd_ice1712_mixer_pro_spdif_route __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route", .info = snd_ice1712_pro_route_info, .get = snd_ice1712_pro_route_spdif_get, .put = snd_ice1712_pro_route_spdif_put, @@ -2204,7 +2207,7 @@ static int snd_ice1712_pro_volume_rate_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = inb(ICEMT(ice, MONITOR_RATE)); return 0; } @@ -2245,7 +2248,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx; - + spin_lock_irq(&ice->reg_lock); for (idx = 0; idx < 22; idx++) { outb(idx, ICEMT(ice, MONITOR_PEAKINDEX)); @@ -2296,12 +2299,12 @@ static int __devinit snd_ice1712_read_eeprom(struct snd_ice1712 *ice, unsigned int i, size; struct snd_ice1712_card_info * const *tbl, *c; - if (! modelname || ! *modelname) { + if (!modelname || !*modelname) { ice->eeprom.subvendor = 0; if ((inb(ICEREG(ice, I2C_CTRL)) & ICE1712_I2C_EEPROM) != 0) ice->eeprom.subvendor = (snd_ice1712_read_i2c(ice, dev, 0x00) << 0) | - (snd_ice1712_read_i2c(ice, dev, 0x01) << 8) | - (snd_ice1712_read_i2c(ice, dev, 0x02) << 16) | + (snd_ice1712_read_i2c(ice, dev, 0x01) << 8) | + (snd_ice1712_read_i2c(ice, dev, 0x02) << 16) | (snd_ice1712_read_i2c(ice, dev, 0x03) << 24); if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { @@ -2318,12 +2321,12 @@ static int __devinit snd_ice1712_read_eeprom(struct snd_ice1712 *ice, } for (tbl = card_tables; *tbl; tbl++) { for (c = *tbl; c->subvendor; c++) { - if (modelname && c->model && ! strcmp(modelname, c->model)) { + if (modelname && c->model && !strcmp(modelname, c->model)) { printk(KERN_INFO "ice1712: Using board model %s\n", c->name); ice->eeprom.subvendor = c->subvendor; } else if (c->subvendor != ice->eeprom.subvendor) continue; - if (! c->eeprom_size || ! c->eeprom_data) + if (!c->eeprom_size || !c->eeprom_data) goto found; /* if the EEPROM is given by the driver, use it */ snd_printdd("using the defined eeprom..\n"); @@ -2484,13 +2487,13 @@ static int __devinit snd_ice1712_build_controls(struct snd_ice1712 *ice) static int snd_ice1712_free(struct snd_ice1712 *ice) { - if (! ice->port) + if (!ice->port) goto __hw_end; /* mask all interrupts */ outb(0xc0, ICEMT(ice, IRQ)); outb(0xff, ICEREG(ice, IRQMASK)); /* --- */ - __hw_end: +__hw_end: if (ice->irq >= 0) free_irq(ice->irq, ice); @@ -2515,7 +2518,7 @@ static int __devinit snd_ice1712_create(struct snd_card *card, int omni, int cs8427_timeout, int dxr_enable, - struct snd_ice1712 ** r_ice1712) + struct snd_ice1712 **r_ice1712) { struct snd_ice1712 *ice; int err; @@ -2525,8 +2528,9 @@ static int __devinit snd_ice1712_create(struct snd_card *card, *r_ice1712 = NULL; - /* enable PCI device */ - if ((err = pci_enable_device(pci)) < 0) + /* enable PCI device */ + err = pci_enable_device(pci); + if (err < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || @@ -2570,7 +2574,8 @@ static int __devinit snd_ice1712_create(struct snd_card *card, snd_ice1712_proc_init(ice); synchronize_irq(pci->irq); - if ((err = pci_request_regions(pci, "ICE1712")) < 0) { + err = pci_request_regions(pci, "ICE1712"); + if (err < 0) { kfree(ice); pci_disable_device(pci); return err; @@ -2586,7 +2591,7 @@ static int __devinit snd_ice1712_create(struct snd_card *card, snd_ice1712_free(ice); return -EIO; } - + ice->irq = pci->irq; if (snd_ice1712_read_eeprom(ice, modelname) < 0) { @@ -2606,9 +2611,10 @@ static int __devinit snd_ice1712_create(struct snd_card *card, ICEREG(ice, IRQMASK)); outb(0x00, ICEMT(ice, IRQ)); - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); + if (err < 0) { snd_ice1712_free(ice); - return err; + return err; } snd_card_set_dev(card, &pci->dev); @@ -2648,10 +2654,10 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, strcpy(card->driver, "ICE1712"); strcpy(card->shortname, "ICEnsemble ICE1712"); - - if ((err = snd_ice1712_create(card, pci, model[dev], omni[dev], - cs8427_timeout[dev], dxr_enable[dev], - &ice)) < 0) { + + err = snd_ice1712_create(card, pci, model[dev], omni[dev], + cs8427_timeout[dev], dxr_enable[dev], &ice); + if (err < 0) { snd_card_free(card); return err; } @@ -2663,7 +2669,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (c->driver) /* specific driver? */ strcpy(card->driver, c->driver); if (c->chip_init) { - if ((err = c->chip_init(ice)) < 0) { + err = c->chip_init(ice); + if (err < 0) { snd_card_free(card); return err; } @@ -2675,47 +2682,52 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, c = &no_matched; __found: - if ((err = snd_ice1712_pcm_profi(ice, pcm_dev++, NULL)) < 0) { + err = snd_ice1712_pcm_profi(ice, pcm_dev++, NULL); + if (err < 0) { snd_card_free(card); return err; } - + if (ice_has_con_ac97(ice)) - if ((err = snd_ice1712_pcm(ice, pcm_dev++, NULL)) < 0) { + err = snd_ice1712_pcm(ice, pcm_dev++, NULL); + if (err < 0) { snd_card_free(card); return err; } - if ((err = snd_ice1712_ac97_mixer(ice)) < 0) { + err = snd_ice1712_ac97_mixer(ice); + if (err < 0) { snd_card_free(card); return err; } - if ((err = snd_ice1712_build_controls(ice)) < 0) { + err = snd_ice1712_build_controls(ice); + if (err < 0) { snd_card_free(card); return err; } if (c->build_controls) { - if ((err = c->build_controls(ice)) < 0) { + err = c->build_controls(ice); + if (err < 0) { snd_card_free(card); return err; } } if (ice_has_con_ac97(ice)) - if ((err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL)) < 0) { + err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL); + if (err < 0) { snd_card_free(card); return err; } - if (! c->no_mpu401) { - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG(ice, MPU1_CTRL), - (c->mpu401_1_info_flags | - MPU401_INFO_INTEGRATED), - ice->irq, 0, - &ice->rmidi[0])) < 0) { + if (!c->no_mpu401) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, + ICEREG(ice, MPU1_CTRL), + (c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED), + ice->irq, 0, &ice->rmidi[0]); + if (err < 0) { snd_card_free(card); return err; } @@ -2727,12 +2739,12 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) { /* 2nd port used */ - if ((err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, - ICEREG(ice, MPU2_CTRL), - (c->mpu401_2_info_flags | - MPU401_INFO_INTEGRATED), - ice->irq, 0, - &ice->rmidi[1])) < 0) { + err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, + ICEREG(ice, MPU2_CTRL), + (c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED), + ice->irq, 0, &ice->rmidi[1]); + + if (err < 0) { snd_card_free(card); return err; } @@ -2750,7 +2762,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, ice->port, ice->irq); - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 762fbd7..fdae6de 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -20,7 +20,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #include <sound/control.h> #include <sound/ac97_codec.h> @@ -112,7 +112,7 @@ */ #define ICEDS(ice, x) ((ice)->dmapath_port + ICE1712_DS_##x) - + #define ICE1712_DS_INTMASK 0x00 /* word - interrupt mask */ #define ICE1712_DS_INTSTAT 0x02 /* word - interrupt status */ #define ICE1712_DS_DATA 0x04 /* dword - channel data */ @@ -121,7 +121,7 @@ /* * Consumer section channel registers */ - + #define ICE1712_DSC_ADDR0 0x00 /* dword - base address 0 */ #define ICE1712_DSC_COUNT0 0x01 /* word - count 0 */ #define ICE1712_DSC_ADDR1 0x02 /* dword - base address 1 */ @@ -138,7 +138,7 @@ #define ICE1712_DSC_RATE 0x05 /* dword - rate */ #define ICE1712_DSC_VOLUME 0x06 /* word - volume control */ -/* +/* * Professional multi-track direct control registers */ @@ -214,7 +214,7 @@ /* - * + * */ struct snd_ice1712; @@ -253,12 +253,12 @@ enum { ICE_EEP1_ADC_ID2, ICE_EEP1_ADC_ID3 }; - + #define ice_has_con_ac97(ice) (!((ice)->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97)) struct snd_ak4xxx_private { - unsigned int cif: 1; /* CIF mode */ + unsigned int cif:1; /* CIF mode */ unsigned char caddr; /* C0 and C1 bits */ unsigned int data_mask; /* DATA gpio bit */ unsigned int clk_mask; /* CLK gpio bit */ @@ -306,11 +306,11 @@ struct snd_ice1712 { struct snd_pcm *pcm; struct snd_pcm *pcm_ds; struct snd_pcm *pcm_pro; - struct snd_pcm_substream *playback_con_substream; - struct snd_pcm_substream *playback_con_substream_ds[6]; - struct snd_pcm_substream *capture_con_substream; - struct snd_pcm_substream *playback_pro_substream; - struct snd_pcm_substream *capture_pro_substream; + struct snd_pcm_substream *playback_con_substream; + struct snd_pcm_substream *playback_con_substream_ds[6]; + struct snd_pcm_substream *capture_con_substream; + struct snd_pcm_substream *playback_pro_substream; + struct snd_pcm_substream *capture_pro_substream; unsigned int playback_pro_size; unsigned int capture_pro_size; unsigned int playback_con_virt_addr[6]; @@ -326,15 +326,15 @@ struct snd_ice1712 { struct snd_ice1712_eeprom eeprom; unsigned int pro_volumes[20]; - unsigned int omni: 1; /* Delta Omni I/O */ - unsigned int dxr_enable: 1; /* Terratec DXR enable for DMX6FIRE */ - unsigned int vt1724: 1; - unsigned int vt1720: 1; - unsigned int has_spdif: 1; /* VT1720/4 - has SPDIF I/O */ - unsigned int force_pdma4: 1; /* VT1720/4 - PDMA4 as non-spdif */ - unsigned int force_rdma1: 1; /* VT1720/4 - RDMA1 as non-spdif */ - unsigned int midi_output: 1; /* VT1720/4: MIDI output triggered */ - unsigned int midi_input: 1; /* VT1720/4: MIDI input triggered */ + unsigned int omni:1; /* Delta Omni I/O */ + unsigned int dxr_enable:1; /* Terratec DXR enable for DMX6FIRE */ + unsigned int vt1724:1; + unsigned int vt1720:1; + unsigned int has_spdif:1; /* VT1720/4 - has SPDIF I/O */ + unsigned int force_pdma4:1; /* VT1720/4 - PDMA4 as non-spdif */ + unsigned int force_rdma1:1; /* VT1720/4 - RDMA1 as non-spdif */ + unsigned int midi_output:1; /* VT1720/4: MIDI output triggered */ + unsigned int midi_input:1; /* VT1720/4: MIDI input triggered */ unsigned int num_total_dacs; /* total DACs */ unsigned int num_total_adcs; /* total ADCs */ unsigned int cur_rate; /* current rate */ @@ -351,7 +351,7 @@ struct snd_ice1712 { struct snd_i2c_bus *i2c; /* I2C bus */ struct snd_i2c_device *cs8427; /* CS8427 I2C device */ unsigned int cs8427_timeout; /* CS8427 reset timeout in HZ/100 */ - + struct ice1712_gpio { unsigned int direction; /* current direction bits */ unsigned int write_mask; /* current mask bits */ @@ -455,7 +455,7 @@ static inline int snd_ice1712_gpio_read_bits(struct snd_ice1712 *ice, { ice->gpio.direction &= ~mask; snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); - return (snd_ice1712_gpio_read(ice) & mask); + return snd_ice1712_gpio_read(ice) & mask; } int snd_ice1712_spdif_build_controls(struct snd_ice1712 *ice); @@ -467,13 +467,13 @@ int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice); int snd_ice1712_init_cs8427(struct snd_ice1712 *ice, int addr); -static inline void snd_ice1712_write(struct snd_ice1712 * ice, u8 addr, u8 data) +static inline void snd_ice1712_write(struct snd_ice1712 *ice, u8 addr, u8 data) { outb(addr, ICEREG(ice, INDEX)); outb(data, ICEREG(ice, DATA)); } -static inline u8 snd_ice1712_read(struct snd_ice1712 * ice, u8 addr) +static inline u8 snd_ice1712_read(struct snd_ice1712 *ice, u8 addr) { outb(addr, ICEREG(ice, INDEX)); return inb(ICEREG(ice, DATA)); @@ -491,7 +491,7 @@ struct snd_ice1712_card_info { char *driver; int (*chip_init)(struct snd_ice1712 *); int (*build_controls)(struct snd_ice1712 *); - unsigned int no_mpu401: 1; + unsigned int no_mpu401:1; unsigned int mpu401_1_info_flags; unsigned int mpu401_2_info_flags; const char *mpu401_1_name;

2 3

[alsa-devel] [PATCH] phase and juli coding style fixes (more coming up)
by Vedran Miletić 07 Sep '08

07 Sep '08

From my signoff: From b4a282ae0d54087bc152571c250afcd57fa9fa38 Mon Sep 17 00:00:00 2001 From: Vedran Miletic <rivanvx(a)gmail.com> Date: Sun, 7 Sep 2008 11:22:35 +0200 Subject: [PATCH] ALSA: ice1712/ice1724: Coding style fixes part 1 (more coming up) Inspired by Alexander Beregalov's patches for wtm and aureon.c, I decided to run checkpatch on some more files. After some work checkpatch.pl-0.23 --no-tree --file --strict <file> reports 0 errors, 0 warnings, 0 checks, n lines checked for: phase.c phase.h juli.c (1 check about unused code, maybe we should comment it) juli.h (no changes necessary) In other files I have just fixed // comments and long lines along the way (but not all of them), more coming up. Signed-off-by: Vedran Miletic <rivanvx(a)gmail.com> Patch is attached. -- Vedran Miletić

2 3

[alsa-devel] [PATCH] hda: removed unneeded hp_nid references.
by Matthew Ranostay 07 Sep '08

07 Sep '08

Removed unneeded hp_nid references for 92hd73xx codec family. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 6ad6ef3..5b715a7 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -3775,17 +3775,14 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ - spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; case 0x4: /* 8 Channel */ - spec->multiout.hp_nid = 0x18; spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; break; case 0x5: /* 10 Channel */ - spec->multiout.hp_nid = 0x19; spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; };

2 1

[alsa-devel] [PATCH] ice1724.c: fix coding style
by Alexander Beregalov 07 Sep '08

07 Sep '08

before: total: 96 errors, 66 warnings, 2612 lines checked after: total: 11 errors, 64 warnings, 2624 lines checked Compile tested only. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/ice1724.c | 180 +++++++++++++++++++++++-------------------- 1 files changed, 96 insertions(+), 84 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 7bb99df..1b3f117 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -20,9 +20,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -105,7 +105,7 @@ static unsigned int PRO_RATE_DEFAULT = 44100; /* * Basic I/O */ - + /* * default rates, default clock routines */ @@ -198,7 +198,7 @@ static void snd_vt1724_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data) static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) { outw(data, ICEREG1724(ice, GPIO_WRITE_MASK)); - if (! ice->vt1720) /* VT1720 supports only 16 GPIO bits */ + if (!ice->vt1720) /* VT1720 supports only 16 GPIO bits */ outb((data >> 16) & 0xff, ICEREG1724(ice, GPIO_WRITE_MASK_22)); inw(ICEREG1724(ice, GPIO_WRITE_MASK)); /* dummy read for pci-posting */ } @@ -206,7 +206,7 @@ static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) static void snd_vt1724_set_gpio_data(struct snd_ice1712 *ice, unsigned int data) { outw(data, ICEREG1724(ice, GPIO_DATA)); - if (! ice->vt1720) + if (!ice->vt1720) outb(data >> 16, ICEREG1724(ice, GPIO_DATA_22)); inw(ICEREG1724(ice, GPIO_DATA)); /* dummy read for pci-posting */ } @@ -214,7 +214,7 @@ static void snd_vt1724_set_gpio_data(struct snd_ice1712 *ice, unsigned int data) static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice) { unsigned int data; - if (! ice->vt1720) + if (!ice->vt1720) data = (unsigned int)inb(ICEREG1724(ice, GPIO_DATA_22)); else data = 0; @@ -399,7 +399,7 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) break; } #endif - handled = 1; + handled = 1; if (status & VT1724_IRQ_MPU_TX) { spin_lock(&ice->reg_lock); if (ice->midi_output) @@ -468,8 +468,8 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) /* ought to really handle this properly */ if (mtstat & VT1724_MULTI_FIFO_ERR) { unsigned char fstat = inb(ICEMT1724(ice, DMA_FIFO_ERR)); - outb(fstat, ICEMT1724(ice, DMA_FIFO_ERR)); - outb(VT1724_MULTI_FIFO_ERR | inb(ICEMT1724(ice, DMA_INT_MASK)), ICEMT1724(ice, DMA_INT_MASK)); + outb(fstat, ICEMT1724(ice, DMA_FIFO_ERR)); + outb(VT1724_MULTI_FIFO_ERR | inb(ICEMT1724(ice, DMA_INT_MASK)), ICEMT1724(ice, DMA_INT_MASK)); /* If I don't do this, I get machine lockup due to continual interrupts */ } @@ -733,17 +733,17 @@ static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream) outl(substream->runtime->dma_addr, ICEMT1724(ice, PLAYBACK_ADDR)); size = (snd_pcm_lib_buffer_bytes(substream) >> 2) - 1; - // outl(size, ICEMT1724(ice, PLAYBACK_SIZE)); + /* outl(size, ICEMT1724(ice, PLAYBACK_SIZE)); */ outw(size, ICEMT1724(ice, PLAYBACK_SIZE)); outb(size >> 16, ICEMT1724(ice, PLAYBACK_SIZE) + 2); size = (snd_pcm_lib_period_bytes(substream) >> 2) - 1; - // outl(size, ICEMT1724(ice, PLAYBACK_COUNT)); + /* outl(size, ICEMT1724(ice, PLAYBACK_COUNT)); */ outw(size, ICEMT1724(ice, PLAYBACK_COUNT)); outb(size >> 16, ICEMT1724(ice, PLAYBACK_COUNT) + 2); spin_unlock_irq(&ice->reg_lock); - // printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); + /* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */ return 0; } @@ -771,7 +771,7 @@ static snd_pcm_uframes_t snd_vt1724_playback_pro_pointer(struct snd_pcm_substrea ptr = inl(ICEMT1724(ice, PLAYBACK_SIZE)) & 0xffffff; ptr = (ptr + 1) << 2; ptr = bytes_to_frames(substream->runtime, ptr); - if (! ptr) + if (!ptr) ; else if (ptr <= substream->runtime->buffer_size) ptr = substream->runtime->buffer_size - ptr; @@ -815,7 +815,7 @@ static snd_pcm_uframes_t snd_vt1724_pcm_pointer(struct snd_pcm_substream *substr ptr = inw(ice->profi_port + reg->size); ptr = (ptr + 1) << 2; ptr = bytes_to_frames(substream->runtime, ptr); - if (! ptr) + if (!ptr) ; else if (ptr <= substream->runtime->buffer_size) ptr = substream->runtime->buffer_size - ptr; @@ -842,8 +842,7 @@ static const struct vt1724_pcm_reg vt1724_capture_pro_reg = { .start = VT1724_RDMA0_START, }; -static const struct snd_pcm_hardware snd_vt1724_playback_pro = -{ +static const struct snd_pcm_hardware snd_vt1724_playback_pro = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -861,8 +860,7 @@ static const struct snd_pcm_hardware snd_vt1724_playback_pro = .periods_max = 1024, }; -static const struct snd_pcm_hardware snd_vt1724_spdif = -{ +static const struct snd_pcm_hardware snd_vt1724_spdif = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -883,8 +881,7 @@ static const struct snd_pcm_hardware snd_vt1724_spdif = .periods_max = 1024, }; -static const struct snd_pcm_hardware snd_vt1724_2ch_stereo = -{ +static const struct snd_pcm_hardware snd_vt1724_2ch_stereo = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -1030,7 +1027,7 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { .pointer = snd_vt1724_pcm_pointer, }; -static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 * ice, int device) +static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; int err; @@ -1115,7 +1112,7 @@ static void update_spdif_rate(struct snd_ice1712 *ice, unsigned int rate) static int snd_vt1724_playback_spdif_prepare(struct snd_pcm_substream *substream) { struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); - if (! ice->force_pdma4) + if (!ice->force_pdma4) update_spdif_rate(ice, substream->runtime->rate); return snd_vt1724_pcm_prepare(substream); } @@ -1215,7 +1212,7 @@ static struct snd_pcm_ops snd_vt1724_capture_spdif_ops = { }; -static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 * ice, int device) +static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) { char *name; struct snd_pcm *pcm; @@ -1234,7 +1231,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 * ice, int device) ice->has_spdif = 1; } else capt = 0; - if (! play && ! capt) + if (!play && !capt) return 0; /* no spdif device */ if (ice->force_pdma4 || ice->force_rdma1) @@ -1349,7 +1346,7 @@ static struct snd_pcm_ops snd_vt1724_playback_indep_ops = { }; -static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 * ice, int device) +static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; int play; @@ -1384,11 +1381,11 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 * ice, int device) * Mixer section */ -static int __devinit snd_vt1724_ac97_mixer(struct snd_ice1712 * ice) +static int __devinit snd_vt1724_ac97_mixer(struct snd_ice1712 *ice) { int err; - if (! (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S)) { + if (!(ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S)) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; static struct snd_ac97_bus_ops ops = { @@ -1401,11 +1398,13 @@ static int __devinit snd_vt1724_ac97_mixer(struct snd_ice1712 * ice) mdelay(5); /* FIXME */ outb(inb(ICEMT1724(ice, AC97_CMD)) & ~0x80, ICEMT1724(ice, AC97_CMD)); - if ((err = snd_ac97_bus(ice->card, 0, &ops, NULL, &pbus)) < 0) + err = snd_ac97_bus(ice->card, 0, &ops, NULL, &pbus); + if (err < 0) return err; memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ice; - if ((err = snd_ac97_mixer(pbus, &ac97, &ice->ac97)) < 0) + err = snd_ac97_mixer(pbus, &ac97, &ice->ac97); + if (err < 0) printk(KERN_WARNING "ice1712: cannot initialize pro ac97, skipped\n"); else return 0; @@ -1426,7 +1425,7 @@ static inline unsigned int eeprom_triple(struct snd_ice1712 *ice, int idx) ((unsigned int)ice->eeprom.data[idx + 2] << 16); } -static void snd_vt1724_proc_read(struct snd_info_entry *entry, +static void snd_vt1724_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ice1712 *ice = entry->private_data; @@ -1468,11 +1467,11 @@ static void snd_vt1724_proc_read(struct snd_info_entry *entry, idx, inb(ice->profi_port+idx)); } -static void __devinit snd_vt1724_proc_init(struct snd_ice1712 * ice) +static void __devinit snd_vt1724_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "ice1724", &entry)) + if (!snd_card_proc_new(ice->card, "ice1724", &entry)) snd_info_set_text_ops(entry, ice, snd_vt1724_proc_read); } @@ -1492,7 +1491,7 @@ static int snd_vt1724_eeprom_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + memcpy(ucontrol->value.bytes.data, &ice->eeprom, sizeof(ice->eeprom)); return 0; } @@ -1607,13 +1606,13 @@ static int snd_vt1724_spdif_default_put(struct snd_kcontrol *kcontrol, if (val != old) update_spdif_bits(ice, val); spin_unlock_irq(&ice->reg_lock); - return (val != old); + return val != old; } static struct snd_kcontrol_new snd_vt1724_spdif_default __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_vt1724_spdif_info, .get = snd_vt1724_spdif_default_get, .put = snd_vt1724_spdif_default_put @@ -1646,7 +1645,7 @@ static struct snd_kcontrol_new snd_vt1724_spdif_maskc __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_vt1724_spdif_info, .get = snd_vt1724_spdif_maskc_get, }; @@ -1655,7 +1654,7 @@ static struct snd_kcontrol_new snd_vt1724_spdif_maskp __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_vt1724_spdif_info, .get = snd_vt1724_spdif_maskp_get, }; @@ -1692,8 +1691,8 @@ static struct snd_kcontrol_new snd_vt1724_spdif_switch __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* FIXME: the following conflict with IEC958 Playback Route */ - // .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), - .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), + /* .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), */ + .name = SNDRV_CTL_NAME_IEC958("Output ", NONE, SWITCH), .info = snd_vt1724_spdif_sw_info, .get = snd_vt1724_spdif_sw_get, .put = snd_vt1724_spdif_sw_put @@ -1713,7 +1712,7 @@ int snd_vt1724_gpio_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int shift = kcontrol->private_value & 0xff; int invert = (kcontrol->private_value & (1<<24)) ? 1 : 0; - + snd_ice1712_save_gpio_status(ice); ucontrol->value.integer.value[0] = (snd_ice1712_gpio_read(ice) & (1 << shift) ? 1 : 0) ^ invert; @@ -1768,7 +1767,7 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int i, rate; - + spin_lock_irq(&ice->reg_lock); if (ice->is_spdif_master(ice)) { ucontrol->value.enumerated.item[0] = ice->hw_rates->count; @@ -1924,7 +1923,7 @@ static int snd_vt1724_pro_route_info(struct snd_kcontrol *kcontrol, "H/W In 0", "H/W In 1", /* 1-2 */ "IEC958 In L", "IEC958 In R", /* 3-4 */ }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 5; @@ -1954,7 +1953,7 @@ static int get_route_val(struct snd_ice1712 *ice, int shift) val = inl(ICEMT1724(ice, ROUTE_PLAYBACK)); val >>= shift; - val &= 7; //we now have 3 bits per output + val &= 7; /* we now have 3 bits per output */ eitem = xlate[val]; if (eitem == 255) { snd_BUG(); @@ -2033,7 +2032,7 @@ static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route __devinitdata = static struct snd_kcontrol_new snd_vt1724_mixer_pro_spdif_route __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route", .info = snd_vt1724_pro_route_info, .get = snd_vt1724_pro_route_spdif_get, .put = snd_vt1724_pro_route_spdif_put, @@ -2056,7 +2055,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx; - + spin_lock_irq(&ice->reg_lock); for (idx = 0; idx < 22; idx++) { outb(idx, ICEMT1724(ice, MONITOR_PEAKINDEX)); @@ -2083,7 +2082,7 @@ static struct snd_ice1712_card_info no_matched __devinitdata; static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_revo_cards, - snd_vt1724_amp_cards, + snd_vt1724_amp_cards, snd_vt1724_aureon_cards, snd_vt1720_mobo_cards, snd_vt1720_pontis_cards, @@ -2121,7 +2120,7 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, wait_i2c_busy(ice); val = inb(ICEREG1724(ice, I2C_DATA)); mutex_unlock(&ice->i2c_mutex); - //printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); + /* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */ return val; } @@ -2130,7 +2129,7 @@ void snd_vt1724_write_i2c(struct snd_ice1712 *ice, { mutex_lock(&ice->i2c_mutex); wait_i2c_busy(ice); - //printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); + /* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */ outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(data, ICEREG1724(ice, I2C_DATA)); outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); @@ -2145,13 +2144,13 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, unsigned int i, size; struct snd_ice1712_card_info * const *tbl, *c; - if (! modelname || ! *modelname) { + if (!modelname || !*modelname) { ice->eeprom.subvendor = 0; if ((inb(ICEREG1724(ice, I2C_CTRL)) & VT1724_I2C_EEPROM) != 0) ice->eeprom.subvendor = (snd_vt1724_read_i2c(ice, dev, 0x00) << 0) | - (snd_vt1724_read_i2c(ice, dev, 0x01) << 8) | - (snd_vt1724_read_i2c(ice, dev, 0x02) << 16) | + (snd_vt1724_read_i2c(ice, dev, 0x01) << 8) | + (snd_vt1724_read_i2c(ice, dev, 0x02) << 16) | (snd_vt1724_read_i2c(ice, dev, 0x03) << 24); if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { @@ -2174,13 +2173,13 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, for (tbl = card_tables; *tbl; tbl++) { for (c = *tbl; c->subvendor; c++) { if (modelname && c->model && - ! strcmp(modelname, c->model)) { + !strcmp(modelname, c->model)) { printk(KERN_INFO "ice1724: Using board model %s\n", c->name); ice->eeprom.subvendor = c->subvendor; } else if (c->subvendor != ice->eeprom.subvendor) continue; - if (! c->eeprom_size || ! c->eeprom_data) + if (!c->eeprom_size || !c->eeprom_data) goto found; /* if the EEPROM is given by the driver, use it */ snd_printdd("using the defined eeprom..\n"); @@ -2322,13 +2321,13 @@ static int __devinit snd_vt1724_build_controls(struct snd_ice1712 *ice) static int snd_vt1724_free(struct snd_ice1712 *ice) { - if (! ice->port) + if (!ice->port) goto __hw_end; /* mask all interrupts */ outb(0xff, ICEMT1724(ice, DMA_INT_MASK)); outb(0xff, ICEREG1724(ice, IRQMASK)); /* --- */ - __hw_end: +__hw_end: if (ice->irq >= 0) free_irq(ice->irq, ice); pci_release_regions(ice->pci); @@ -2348,7 +2347,7 @@ static int snd_vt1724_dev_free(struct snd_device *device) static int __devinit snd_vt1724_create(struct snd_card *card, struct pci_dev *pci, const char *modelname, - struct snd_ice1712 ** r_ice1712) + struct snd_ice1712 **r_ice1712) { struct snd_ice1712 *ice; int err; @@ -2359,8 +2358,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card, *r_ice1712 = NULL; - /* enable PCI device */ - if ((err = pci_enable_device(pci)) < 0) + /* enable PCI device */ + err = pci_enable_device(pci); + if (err < 0) return err; ice = kzalloc(sizeof(*ice), GFP_KERNEL); @@ -2384,7 +2384,8 @@ static int __devinit snd_vt1724_create(struct snd_card *card, snd_vt1724_proc_init(ice); synchronize_irq(pci->irq); - if ((err = pci_request_regions(pci, "ICE1724")) < 0) { + err = pci_request_regions(pci, "ICE1724"); + if (err < 0) { kfree(ice); pci_disable_device(pci); return err; @@ -2419,9 +2420,10 @@ static int __devinit snd_vt1724_create(struct snd_card *card, */ outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); + if (err < 0) { snd_vt1724_free(ice); - return err; + return err; } snd_card_set_dev(card, &pci->dev); @@ -2459,8 +2461,9 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, strcpy(card->driver, "ICE1724"); strcpy(card->shortname, "ICEnsemble ICE1724"); - - if ((err = snd_vt1724_create(card, pci, model[dev], &ice)) < 0) { + + err = snd_vt1724_create(card, pci, model[dev], &ice); + if (err < 0) { snd_card_free(card); return err; } @@ -2472,7 +2475,8 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (c->driver) /* specific driver? */ strcpy(card->driver, c->driver); if (c->chip_init) { - if ((err = c->chip_init(ice)) < 0) { + err = c->chip_init(ice); + if (err < 0) { snd_card_free(card); return err; } @@ -2482,15 +2486,15 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, } } c = &no_matched; - __found: - /* - * VT1724 has separate DMAs for the analog and the SPDIF streams while - * ICE1712 has only one for both (mixed up). - * - * Confusingly the analog PCM is named "professional" here because it - * was called so in ice1712 driver, and vt1724 driver is derived from - * ice1712 driver. - */ +__found: + /* + * VT1724 has separate DMAs for the analog and the SPDIF streams while + * ICE1712 has only one for both (mixed up). + * + * Confusingly the analog PCM is named "professional" here because it + * was called so in ice1712 driver, and vt1724 driver is derived from + * ice1712 driver. + */ ice->pro_rate_default = PRO_RATE_DEFAULT; if (!ice->is_spdif_master) ice->is_spdif_master = stdclock_is_spdif_master; @@ -2505,46 +2509,53 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (!ice->hw_rates) set_std_hw_rates(ice); - if ((err = snd_vt1724_pcm_profi(ice, pcm_dev++)) < 0) { + err = snd_vt1724_pcm_profi(ice, pcm_dev++); + if (err < 0) { snd_card_free(card); return err; } - - if ((err = snd_vt1724_pcm_spdif(ice, pcm_dev++)) < 0) { + + err = snd_vt1724_pcm_spdif(ice, pcm_dev++); + if (err < 0) { snd_card_free(card); return err; } - - if ((err = snd_vt1724_pcm_indep(ice, pcm_dev++)) < 0) { + + err = snd_vt1724_pcm_indep(ice, pcm_dev++); + if (err < 0) { snd_card_free(card); return err; } - if ((err = snd_vt1724_ac97_mixer(ice)) < 0) { + err = snd_vt1724_ac97_mixer(ice); + if (err < 0) { snd_card_free(card); return err; } - if ((err = snd_vt1724_build_controls(ice)) < 0) { + err = snd_vt1724_build_controls(ice); + if (err < 0) { snd_card_free(card); return err; } if (ice->pcm && ice->has_spdif) { /* has SPDIF I/O */ - if ((err = snd_vt1724_spdif_build_controls(ice)) < 0) { + err = snd_vt1724_spdif_build_controls(ice); + if (err < 0) { snd_card_free(card); return err; } } if (c->build_controls) { - if ((err = c->build_controls(ice)) < 0) { + err = c->build_controls(ice); + if (err < 0) { snd_card_free(card); return err; } } - if (! c->no_mpu401) { + if (!c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { struct snd_rawmidi *rmidi; @@ -2576,7 +2587,8 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, ice->port, ice->irq); - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) { snd_card_free(card); return err; }

2 1

[alsa-devel] [PATCH] ice1712/wtm: fix coding style
by Alexander Beregalov 07 Sep '08

07 Sep '08

before: total: 2 errors, 0 warnings, 20 lines checked total: 49 errors, 2 warnings, 518 lines checked after: total: 0 errors, 0 warnings, 20 lines checked total: 0 errors, 0 warnings, 518 lines checked Compile tested, size and code are equal. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/wtm.c | 104 +++++++++++++++++++++++----------------------- sound/pci/ice1712/wtm.h | 4 +- 2 files changed, 54 insertions(+), 54 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index a08d17c..5af9e84 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -1,12 +1,12 @@ /* * ALSA driver for ICEnsemble VT1724 (Envy24HT) - * + * * Lowlevel functions for Ego Sys Waveterminal 192M * * Copyright (c) 2006 Guedez Clement <klem.dev(a)gmail.com> * Some functions are taken from the Prodigy192 driver * source - * + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -20,12 +20,12 @@ * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ + * + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -39,9 +39,9 @@ /* - * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus + * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus */ -static inline void stac9460_put(struct snd_ice1712 *ice, int reg, +static inline void stac9460_put(struct snd_ice1712 *ice, int reg, unsigned char val) { snd_vt1724_write_i2c(ice, STAC9460_I2C_ADDR, reg, val); @@ -73,7 +73,7 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char val; @@ -88,14 +88,14 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, } if (id < 6) val = stac9460_get(ice, idx); - else - val = stac9460_2_get(ice,idx - 6); + else + val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; return 0; } static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char new, old; @@ -105,8 +105,8 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); - new = (~ucontrol->value.integer.value[0]<< 7 & 0x80) | - (old & ~0x80); + new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | + (old & ~0x80); change = (new != old); if (change) { stac9460_put(ice, idx, new); @@ -117,16 +117,16 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, idx = id + STAC946X_LF_VOLUME; if (id < 6) old = stac9460_get(ice, idx); - else + else old = stac9460_2_get(ice, idx - 6); - new = (~ucontrol->value.integer.value[0]<< 7 & 0x80) | + new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id < 6) - stac9460_put(ice, idx, new); + stac9460_put(ice, idx, new); else - stac9460_2_put(ice, idx - 6, new); + stac9460_2_put(ice, idx - 6, new); } } return change; @@ -136,7 +136,7 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, * DAC volume attenuation mixer control */ static int stac9460_dac_vol_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -146,7 +146,7 @@ static int stac9460_dac_vol_info(struct snd_kcontrol *kcontrol, } static int stac9460_dac_vol_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx, id; @@ -161,14 +161,14 @@ static int stac9460_dac_vol_get(struct snd_kcontrol *kcontrol, } if (id < 6) vol = stac9460_get(ice, idx) & 0x7f; - else + else vol = stac9460_2_get(ice, idx - 6) & 0x7f; ucontrol->value.integer.value[0] = 0x7f - vol; return 0; } static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx, id; @@ -182,8 +182,8 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); if (change) { - stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); - stac9460_2_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); + stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); + stac9460_2_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); } } else { id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); @@ -191,17 +191,17 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, nvol = ucontrol->value.integer.value[0] & 0x7f; if (id < 6) tmp = stac9460_get(ice, idx); - else + else tmp = stac9460_2_get(ice, idx - 6); ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); if (change) { if (id < 6) stac9460_put(ice, idx, (0x7f - nvol) | - (tmp & 0x80)); - else + (tmp & 0x80)); + else stac9460_2_put(ice, idx-6, (0x7f - nvol) | - (tmp & 0x80)); + (tmp & 0x80)); } } return change; @@ -213,12 +213,12 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, #define stac9460_adc_mute_info snd_ctl_boolean_stereo_info static int stac9460_adc_mute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char val; int i, id; - + id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); if (id == 0) { for (i = 0; i < 2; ++i) { @@ -235,20 +235,20 @@ static int stac9460_adc_mute_get(struct snd_kcontrol *kcontrol, } static int stac9460_adc_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char new, old; int i, reg, id; int change; - + id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); if (id == 0) { for (i = 0; i < 2; ++i) { reg = STAC946X_MIC_L_VOLUME + i; old = stac9460_get(ice, reg); new = (~ucontrol->value.integer.value[i]<<7&0x80) | - (old&~0x80); + (old&~0x80); change = (new != old); if (change) stac9460_put(ice, reg, new); @@ -258,7 +258,7 @@ static int stac9460_adc_mute_put(struct snd_kcontrol *kcontrol, reg = STAC946X_MIC_L_VOLUME + i; old = stac9460_2_get(ice, reg); new = (~ucontrol->value.integer.value[i]<<7&0x80) | - (old&~0x80); + (old&~0x80); change = (new != old); if (change) stac9460_2_put(ice, reg, new); @@ -271,7 +271,7 @@ static int stac9460_adc_mute_put(struct snd_kcontrol *kcontrol, *ADC gain mixer control */ static int stac9460_adc_vol_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -281,12 +281,12 @@ static int stac9460_adc_vol_info(struct snd_kcontrol *kcontrol, } static int stac9460_adc_vol_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int i, reg, id; unsigned char vol; - + id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); if (id == 0) { for (i = 0; i < 2; ++i) { @@ -305,13 +305,13 @@ static int stac9460_adc_vol_get(struct snd_kcontrol *kcontrol, } static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int i, reg, id; unsigned char ovol, nvol; int change; - + id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); if (id == 0) { for (i = 0; i < 2; ++i) { @@ -321,7 +321,7 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, change = ((ovol & 0x0f) != nvol); if (change) stac9460_put(ice, reg, (0x0f - nvol) | - (ovol & ~0x0f)); + (ovol & ~0x0f)); } } else { for (i = 0; i < 2; ++i) { @@ -331,7 +331,7 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, change = ((ovol & 0x0f) != nvol); if (change) stac9460_2_put(ice, reg, (0x0f - nvol) | - (ovol & ~0x0f)); + (ovol & ~0x0f)); } } return change; @@ -344,23 +344,23 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, #define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char val; int id; - + id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); if (id == 0) - val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); + val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else - val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); + val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); ucontrol->value.integer.value[0] = ~val>>7 & 0x1; return 0; } static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char new, old; @@ -368,16 +368,16 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); if (id == 0) - old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); + old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else - old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0]<< 7 & 0x80) | (old & ~0x80); + old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); + new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) - stac9460_put(ice, STAC946X_GENERAL_PURPOSE, new); + stac9460_put(ice, STAC946X_GENERAL_PURPOSE, new); else - stac9460_2_put(ice, STAC946X_GENERAL_PURPOSE, new); + stac9460_2_put(ice, STAC946X_GENERAL_PURPOSE, new); } return change; } @@ -443,7 +443,7 @@ static struct snd_kcontrol_new stac9640_controls[] __devinitdata = { .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - } + } }; @@ -470,7 +470,7 @@ static int __devinit wtm_init(struct snd_ice1712 *ice) (unsigned short)-1 }; unsigned short *p; - + /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; diff --git a/sound/pci/ice1712/wtm.h b/sound/pci/ice1712/wtm.h index 03a394e..423c1a2 100644 --- a/sound/pci/ice1712/wtm.h +++ b/sound/pci/ice1712/wtm.h @@ -10,8 +10,8 @@ */ #define AK4114_ADDR 0x20 /*S/PDIF receiver*/ -#define STAC9460_I2C_ADDR 0x54 /* ADC*2 | DAC*6 */ -#define STAC9460_2_I2C_ADDR 0x56 /* ADC|DAC *2 */ +#define STAC9460_I2C_ADDR 0x54 /* ADC*2 | DAC*6 */ +#define STAC9460_2_I2C_ADDR 0x56 /* ADC|DAC *2 */ extern struct snd_ice1712_card_info snd_vt1724_wtm_cards[];

2 1

[alsa-devel] [PATCH] ice1712/aureon.c: fix coding style
by Alexander Beregalov 07 Sep '08

07 Sep '08

before: total: 304 errors, 137 warnings, 2259 lines checked after: total: 0 errors, 121 warnings, 2284 lines checked Compile tested, size is different because of include was changed, but without that change md5sum is different because of cmp cx,dx/cmp dx,cx swap and __LINE__ was changed in printk function. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/aureon.c | 677 +++++++++++++++++++++++--------------------- 1 files changed, 351 insertions(+), 326 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index a137fc1..110d16e 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -44,10 +44,9 @@ * not working: prety much everything else, at least i could verify that * we have no digital output, no capture, pretty bad clicks and poops * on mixer switch and other coll stuff. - * - */ + */ -#include <asm/io.h> +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> @@ -131,7 +130,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, snd_ice1712_gpio_write(ice, tmp); udelay(50); - /* + /* * send i2c stop condition and start condition * to obtain sane state */ @@ -152,10 +151,16 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, * skipping ack cycles inbetween */ for (j = 0; j < 3; j++) { - switch(j) { - case 0: val = dev; break; - case 1: val = reg; break; - case 2: val = data; break; + switch (j) { + case 0: + val = dev; + break; + case 1: + val = reg; + break; + case 2: + val = data; + break; } for (i = 7; i >= 0; i--) { tmp &= ~AUREON_SPI_CLK; @@ -171,7 +176,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, snd_ice1712_gpio_write(ice, tmp); udelay(40); } - tmp &= ~AUREON_SPI_CLK; + tmp &= ~AUREON_SPI_CLK; snd_ice1712_gpio_write(ice, tmp); udelay(40); tmp |= AUREON_SPI_CLK; @@ -203,7 +208,7 @@ static int aureon_universe_inmux_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; - if(uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; @@ -231,12 +236,12 @@ static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol, return -EINVAL; snd_ice1712_save_gpio_status(ice); oval = spec->pca9554_out; - if ((change = (oval != nval))) { + change = (oval != nval); + if (change) { aureon_pca9554_write(ice, PCA9554_OUT, nval); spec->pca9554_out = nval; } snd_ice1712_restore_gpio_status(ice); - return change; } @@ -256,7 +261,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, udelay(10); tmp &= ~AUREON_AC97_ADDR; snd_ice1712_gpio_write(ice, tmp); - udelay(10); + udelay(10); /* Send low-order byte to XILINX chip */ tmp &= ~AUREON_AC97_DATA_MASK; @@ -269,7 +274,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, tmp &= ~AUREON_AC97_DATA_LOW; snd_ice1712_gpio_write(ice, tmp); udelay(10); - + /* Send high-order byte to XILINX chip */ tmp &= ~AUREON_AC97_DATA_MASK; tmp |= (val >> 8) & AUREON_AC97_DATA_MASK; @@ -282,7 +287,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, tmp &= ~AUREON_AC97_DATA_HIGH; snd_ice1712_gpio_write(ice, tmp); udelay(10); - + /* Instruct XILINX chip to parse the data to the STAC9744 chip */ tmp |= AUREON_AC97_COMMIT; snd_ice1712_gpio_write(ice, tmp); @@ -290,7 +295,7 @@ static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg, tmp &= ~AUREON_AC97_COMMIT; snd_ice1712_gpio_write(ice, tmp); udelay(10); - + /* Store the data in out private buffer */ spec->stac9744[(reg & 0x7F) >> 1] = val; } @@ -304,7 +309,7 @@ static unsigned short aureon_ac97_read(struct snd_ice1712 *ice, unsigned short r /* * Initialize STAC9744 chip */ -static int aureon_ac97_init (struct snd_ice1712 *ice) +static int aureon_ac97_init(struct snd_ice1712 *ice) { struct aureon_spec *spec = ice->spec; int i; @@ -335,20 +340,21 @@ static int aureon_ac97_init (struct snd_ice1712 *ice) tmp = (snd_ice1712_gpio_read(ice) | AUREON_AC97_RESET) & ~AUREON_AC97_DATA_MASK; snd_ice1712_gpio_write(ice, tmp); udelay(3); - + tmp &= ~AUREON_AC97_RESET; snd_ice1712_gpio_write(ice, tmp); udelay(3); - + tmp |= AUREON_AC97_RESET; snd_ice1712_gpio_write(ice, tmp); udelay(3); - + memset(&spec->stac9744, 0, sizeof(spec->stac9744)); - for (i=0; ac97_defaults[i] != (unsigned short)-1; i+=2) + for (i = 0; ac97_defaults[i] != (unsigned short)-1; i += 2) spec->stac9744[(ac97_defaults[i]) >> 1] = ac97_defaults[i+1]; - - aureon_ac97_write(ice, AC97_MASTER, 0x0000); // Unmute AC'97 master volume permanently - muting is done by WM8770 + + /* Unmute AC'97 master volume permanently - muting is done by WM8770 */ + aureon_ac97_write(ice, AC97_MASTER, 0x0000); return 0; } @@ -388,7 +394,7 @@ static int aureon_ac97_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short ovol, nvol; int change; - + snd_ice1712_save_gpio_status(ice); ovol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F); @@ -396,13 +402,14 @@ static int aureon_ac97_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele if (kcontrol->private_value & AUREON_AC97_STEREO) nvol |= ((0x1F - ucontrol->value.integer.value[1]) << 8) & 0x1F00; nvol |= ovol & ~0x1F1F; - - if ((change = (ovol != nvol))) + + change = (ovol != nvol); + if (change) aureon_ac97_write(ice, kcontrol->private_value & 0x7F, nvol); snd_ice1712_restore_gpio_status(ice); - return change; + return change; } /* @@ -416,7 +423,8 @@ static int aureon_ac97_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = aureon_ac97_read(ice, kcontrol->private_value & 0x7F) & 0x8000 ? 0 : 1; + ucontrol->value.integer.value[0] = aureon_ac97_read(ice, + kcontrol->private_value & 0x7F) & 0x8000 ? 0 : 1; mutex_unlock(&ice->gpio_mutex); return 0; @@ -429,13 +437,14 @@ static int aureon_ac97_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el int change; snd_ice1712_save_gpio_status(ice); - + ovol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F); - nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x8000) | (ovol & ~ 0x8000); - - if ((change = (ovol != nvol))) + nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x8000) | (ovol & ~0x8000); + + change = (ovol != nvol); + if (change) aureon_ac97_write(ice, kcontrol->private_value & 0x7F, nvol); - + snd_ice1712_restore_gpio_status(ice); return change; @@ -465,13 +474,14 @@ static int aureon_ac97_micboost_put(struct snd_kcontrol *kcontrol, struct snd_ct int change; snd_ice1712_save_gpio_status(ice); - + ovol = aureon_ac97_read(ice, AC97_MIC); nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x0020) | (ovol & ~0x0020); - - if ((change = (ovol != nvol))) + + change = (ovol != nvol); + if (change) aureon_ac97_write(ice, AC97_MIC, nvol); - + snd_ice1712_restore_gpio_status(ice); return change; @@ -493,16 +503,15 @@ static void aureon_spi_write(struct snd_ice1712 *ice, unsigned int cs, unsigned snd_ice1712_gpio_set_mask(ice, ~(PRODIGY_SPI_MOSI|PRODIGY_SPI_CLK|PRODIGY_WM_CS)); mosi = PRODIGY_SPI_MOSI; clk = PRODIGY_SPI_CLK; - } - else { + } else { snd_ice1712_gpio_set_mask(ice, ~(AUREON_WM_RW|AUREON_SPI_MOSI|AUREON_SPI_CLK| AUREON_WM_CS|AUREON_CS8415_CS)); mosi = AUREON_SPI_MOSI; clk = AUREON_SPI_CLK; - + tmp |= AUREON_WM_RW; } - + tmp &= ~cs; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -534,7 +543,9 @@ static void aureon_spi_write(struct snd_ice1712 *ice, unsigned int cs, unsigned /* * Read data in SPI mode */ -static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned int data, int bits, unsigned char *buffer, int size) { +static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, + unsigned int data, int bits, unsigned char *buffer, int size) +{ int i, j; unsigned int tmp; @@ -544,7 +555,7 @@ static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned i snd_ice1712_gpio_write(ice, tmp); udelay(1); - for (i=bits-1; i>=0; i--) { + for (i = bits-1; i >= 0; i--) { if (data & (1 << i)) tmp |= AUREON_SPI_MOSI; else @@ -561,9 +572,9 @@ static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned i udelay(1); } - for (j=0; j<size; j++) { + for (j = 0; j < size; j++) { unsigned char outdata = 0; - for (i=7; i>=0; i--) { + for (i = 7; i >= 0; i--) { tmp = snd_ice1712_gpio_read(ice); outdata <<= 1; outdata |= (tmp & AUREON_SPI_MISO) ? 1 : 0; @@ -584,19 +595,24 @@ static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs, unsigned i snd_ice1712_gpio_write(ice, tmp); } -static unsigned char aureon_cs8415_get(struct snd_ice1712 *ice, int reg) { +static unsigned char aureon_cs8415_get(struct snd_ice1712 *ice, int reg) +{ unsigned char val; aureon_spi_write(ice, AUREON_CS8415_CS, 0x2000 | reg, 16); aureon_spi_read(ice, AUREON_CS8415_CS, 0x21, 8, &val, 1); return val; } -static void aureon_cs8415_read(struct snd_ice1712 *ice, int reg, unsigned char *buffer, int size) { +static void aureon_cs8415_read(struct snd_ice1712 *ice, int reg, + unsigned char *buffer, int size) +{ aureon_spi_write(ice, AUREON_CS8415_CS, 0x2000 | reg, 16); aureon_spi_read(ice, AUREON_CS8415_CS, 0x21, 8, buffer, size); } -static void aureon_cs8415_put(struct snd_ice1712 *ice, int reg, unsigned char val) { +static void aureon_cs8415_put(struct snd_ice1712 *ice, int reg, + unsigned char val) +{ aureon_spi_write(ice, AUREON_CS8415_CS, 0x200000 | (reg << 8) | val, 24); } @@ -654,18 +670,20 @@ static int aureon_ac97_mmute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e return 0; } -static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short ovol, nvol; int change; - + snd_ice1712_save_gpio_status(ice); - + ovol = wm_get(ice, WM_OUT_MUX1); nvol = (ovol & ~0x02) | (ucontrol->value.integer.value[0] ? 0x02 : 0x00); - if ((change = (ovol != nvol))) + change = (ovol != nvol); + if (change) wm_put(ice, WM_OUT_MUX1, nvol); - + snd_ice1712_restore_gpio_status(ice); return change; @@ -702,12 +720,12 @@ static const unsigned char wm_vol[256] = { static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) { unsigned char nvol; - + if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX]; - + wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); } @@ -736,7 +754,8 @@ static int wm_pcm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va snd_ice1712_save_gpio_status(ice); oval = wm_get(ice, WM_MUTE); nval = (oval & ~0x10) | (ucontrol->value.integer.value[0] ? 0 : 0x10); - if ((change = (nval != oval))) + change = (oval != nval); + if (change) wm_put(ice, WM_MUTE, nval); snd_ice1712_restore_gpio_status(ice); @@ -760,7 +779,7 @@ static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; int i; - for (i=0; i<2; i++) + for (i = 0; i < 2; i++) ucontrol->value.integer.value[i] = spec->master[i] & ~WM_VOL_MUTE; return 0; @@ -849,7 +868,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * /* * WM8770 mute control */ -static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { +static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = kcontrol->private_value >> 8; uinfo->value.integer.min = 0; @@ -862,7 +882,7 @@ static int wm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; int voices, ofs, i; - + voices = kcontrol->private_value >> 8; ofs = kcontrol->private_value & 0xFF; @@ -907,7 +927,7 @@ static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; - + ucontrol->value.integer.value[0] = (spec->master[0] & WM_VOL_MUTE) ? 0 : 1; ucontrol->value.integer.value[1] = @@ -1083,21 +1103,21 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts[] = { - "CD", //AIN1 - "Aux", //AIN2 - "Line", //AIN3 - "Mic", //AIN4 - "AC97" //AIN5 + "CD", /* AIN1 */ + "Aux", /* AIN2 */ + "Line", /* AIN3 */ + "Mic", /* AIN4 */ + "AC97" /* AIN5 */ }; static const char * const universe_texts[] = { - "Aux1", //AIN1 - "CD", //AIN2 - "Phono", //AIN3 - "Line", //AIN4 - "Aux2", //AIN5 - "Mic", //AIN6 - "Aux3", //AIN7 - "AC97" //AIN8 + "Aux1", /* AIN1 */ + "CD", /* AIN2 */ + "Phono", /* AIN3 */ + "Line", /* AIN4 */ + "Aux2", /* AIN5 */ + "Mic", /* AIN6 */ + "Aux3", /* AIN7 */ + "AC97" /* AIN8 */ }; struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); @@ -1108,8 +1128,7 @@ static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, universe_texts[uinfo->value.enumerated.item]); - } - else { + } else { uinfo->value.enumerated.items = 5; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; @@ -1156,8 +1175,8 @@ static int aureon_cs8415_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); static const char * const aureon_texts[] = { - "CD", //RXP0 - "Optical" //RXP1 + "CD", /* RXP0 */ + "Optical" /* RXP1 */ }; static const char * const prodigy_texts[] = { "CD", @@ -1180,10 +1199,10 @@ static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); struct aureon_spec *spec = ice->spec; - //snd_ice1712_save_gpio_status(ice); - //val = aureon_cs8415_get(ice, CS8415_CTRL2); + /* snd_ice1712_save_gpio_status(ice); */ + /* val = aureon_cs8415_get(ice, CS8415_CTRL2); */ ucontrol->value.enumerated.item[0] = spec->cs8415_mux; - //snd_ice1712_restore_gpio_status(ice); + /* snd_ice1712_restore_gpio_status(ice); */ return 0; } @@ -1206,7 +1225,7 @@ static int aureon_cs8415_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -static int aureon_cs8415_rate_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int aureon_cs8415_rate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -1215,7 +1234,7 @@ static int aureon_cs8415_rate_info (struct snd_kcontrol *kcontrol, struct snd_ct return 0; } -static int aureon_cs8415_rate_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int aureon_cs8415_rate_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char ratio; @@ -1229,7 +1248,7 @@ static int aureon_cs8415_rate_get (struct snd_kcontrol *kcontrol, struct snd_ctl */ #define aureon_cs8415_mute_info snd_ctl_boolean_mono_info -static int aureon_cs8415_mute_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int aureon_cs8415_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); snd_ice1712_save_gpio_status(ice); @@ -1238,7 +1257,7 @@ static int aureon_cs8415_mute_get (struct snd_kcontrol *kcontrol, struct snd_ctl return 0; } -static int aureon_cs8415_mute_put (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int aureon_cs8415_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned char oval, nval; @@ -1249,7 +1268,8 @@ static int aureon_cs8415_mute_put (struct snd_kcontrol *kcontrol, struct snd_ctl nval = oval & ~0x20; else nval = oval | 0x20; - if ((change = (oval != nval))) + change = (oval != nval); + if (change) aureon_cs8415_put(ice, CS8415_CTRL1, nval); snd_ice1712_restore_gpio_status(ice); return change; @@ -1258,15 +1278,17 @@ static int aureon_cs8415_mute_put (struct snd_kcontrol *kcontrol, struct snd_ctl /* * CS8415A Q-Sub info */ -static int aureon_cs8415_qsub_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { +static int aureon_cs8415_qsub_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = 10; return 0; } -static int aureon_cs8415_qsub_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_cs8415_qsub_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + snd_ice1712_save_gpio_status(ice); aureon_cs8415_read(ice, CS8415_QSUB, ucontrol->value.bytes.data, 10); snd_ice1712_restore_gpio_status(ice); @@ -1274,18 +1296,21 @@ static int aureon_cs8415_qsub_get (struct snd_kcontrol *kcontrol, struct snd_ctl return 0; } -static int aureon_cs8415_spdif_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { +static int aureon_cs8415_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; uinfo->count = 1; return 0; } -static int aureon_cs8415_mask_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_cs8415_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ memset(ucontrol->value.iec958.status, 0xFF, 24); return 0; } -static int aureon_cs8415_spdif_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { +static int aureon_cs8415_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); snd_ice1712_save_gpio_status(ice); @@ -1311,9 +1336,9 @@ static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT && ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) - tmp &= ~ AUREON_HP_SEL; + tmp &= ~AUREON_HP_SEL; else - tmp &= ~ PRODIGY_HP_SEL; + tmp &= ~PRODIGY_HP_SEL; if (tmp != tmp2) { snd_ice1712_gpio_write(ice, tmp); return 1; @@ -1325,7 +1350,7 @@ static int aureon_get_headphone_amp(struct snd_ice1712 *ice) { unsigned int tmp = snd_ice1712_gpio_read(ice); - return ( tmp & AUREON_HP_SEL )!= 0; + return (tmp & AUREON_HP_SEL) != 0; } #define aureon_hpamp_info snd_ctl_boolean_mono_info @@ -1343,7 +1368,7 @@ static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - return aureon_set_headphone_amp(ice,ucontrol->value.integer.value[0]); + return aureon_set_headphone_amp(ice, ucontrol->value.integer.value[0]); } /* @@ -1390,7 +1415,7 @@ static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return 0; } static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1434,7 +1459,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = wm_master_vol_info, .get = wm_master_vol_get, @@ -1452,7 +1477,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Front Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1471,7 +1496,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Rear Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1490,7 +1515,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Center Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1509,7 +1534,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "LFE Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1528,7 +1553,7 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Side Playback Volume", .info = wm_vol_info, .get = wm_vol_get, @@ -1539,23 +1564,23 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }; static struct snd_kcontrol_new wm_controls[] __devinitdata = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", .info = wm_pcm_mute_info, .get = wm_pcm_mute_get, .put = wm_pcm_mute_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "PCM Playback Volume", .info = wm_pcm_vol_info, .get = wm_pcm_vol_get, .put = wm_pcm_vol_put, .tlv = { .p = db_scale_wm_pcm } - }, + }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Switch", @@ -1566,7 +1591,7 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Capture Volume", .info = wm_adc_vol_info, .get = wm_adc_vol_get, @@ -1605,232 +1630,232 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { }; static struct snd_kcontrol_new ac97_controls[] __devinitdata = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "AC97 Playback Switch", .info = aureon_ac97_mmute_info, .get = aureon_ac97_mmute_get, .put = aureon_ac97_mmute_put, .private_value = AC97_MASTER - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "AC97 Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MASTER|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "AC97 Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MASTER|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_master } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "CD Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_CD - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CD Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_CD + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "CD Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_CD|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "CD Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_CD|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Aux Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_AUX, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_AUX, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Aux Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_AUX|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Aux Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_AUX|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Line Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_LINE - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_LINE + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Line Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_LINE|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Line Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_LINE|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_MIC - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_MIC + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Mic Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MIC, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Mic Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MIC, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost (+20dB)", - .info = aureon_ac97_micboost_info, - .get = aureon_ac97_micboost_get, - .put = aureon_ac97_micboost_put - } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Boost (+20dB)", + .info = aureon_ac97_micboost_info, + .get = aureon_ac97_micboost_get, + .put = aureon_ac97_micboost_put + } }; static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "AC97 Playback Switch", .info = aureon_ac97_mmute_info, .get = aureon_ac97_mmute_get, .put = aureon_ac97_mmute_put, .private_value = AC97_MASTER - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "AC97 Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MASTER|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "AC97 Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MASTER|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_master } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "CD Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_AUX - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CD Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_AUX + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "CD Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_AUX|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "CD Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_AUX|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Phono Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_CD - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Phono Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_CD + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Phono Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_CD|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Phono Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_CD|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Line Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_LINE - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_LINE + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Line Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_LINE|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Line Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_LINE|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_MIC - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_MIC + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Mic Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_MIC, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Mic Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_MIC, .tlv = { .p = db_scale_ac97_gain } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Boost (+20dB)", - .info = aureon_ac97_micboost_info, - .get = aureon_ac97_micboost_get, - .put = aureon_ac97_micboost_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Aux Playback Switch", - .info = aureon_ac97_mute_info, - .get = aureon_ac97_mute_get, - .put = aureon_ac97_mute_put, - .private_value = AC97_VIDEO, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Boost (+20dB)", + .info = aureon_ac97_micboost_info, + .get = aureon_ac97_micboost_get, + .put = aureon_ac97_micboost_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Playback Switch", + .info = aureon_ac97_mute_info, + .get = aureon_ac97_mute_get, + .put = aureon_ac97_mute_put, + .private_value = AC97_VIDEO, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Aux Playback Volume", - .info = aureon_ac97_vol_info, - .get = aureon_ac97_vol_get, - .put = aureon_ac97_vol_put, - .private_value = AC97_VIDEO|AUREON_AC97_STEREO, + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Aux Playback Volume", + .info = aureon_ac97_vol_info, + .get = aureon_ac97_vol_get, + .put = aureon_ac97_vol_put, + .private_value = AC97_VIDEO|AUREON_AC97_STEREO, .tlv = { .p = db_scale_ac97_gain } - }, + }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Aux Source", @@ -1844,43 +1869,43 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { static struct snd_kcontrol_new cs8415_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH), .info = aureon_cs8415_mute_info, .get = aureon_cs8415_mute_get, .put = aureon_cs8415_mute_put }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Source", + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Source", .info = aureon_cs8415_mux_info, .get = aureon_cs8415_mux_get, .put = aureon_cs8415_mux_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("Q-subcode ",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("Q-subcode ", CAPTURE, DEFAULT), .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = aureon_cs8415_qsub_info, .get = aureon_cs8415_qsub_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK), .access = SNDRV_CTL_ELEM_ACCESS_READ, .info = aureon_cs8415_spdif_info, .get = aureon_cs8415_mask_get }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = aureon_cs8415_spdif_info, .get = aureon_cs8415_spdif_get }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Rate", - .access =SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = aureon_cs8415_rate_info, .get = aureon_cs8415_rate_get } @@ -1905,15 +1930,14 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } - + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) { for (i = 0; i < ARRAY_SIZE(universe_ac97_controls); i++) { err = snd_ctl_add(ice->card, snd_ctl_new1(&universe_ac97_controls[i], ice)); if (err < 0) return err; } - } - else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT && + } else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT && ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) { for (i = 0; i < ARRAY_SIZE(ac97_controls); i++) { err = snd_ctl_add(ice->card, snd_ctl_new1(&ac97_controls[i], ice)); @@ -1932,7 +1956,7 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) else if ((id & 0x0F) != 0x01) snd_printk(KERN_INFO "Detected unsupported CS8415 rev. (%c)\n", (char)((id & 0x0F) + 'A' - 1)); else { - for (i = 0; i< ARRAY_SIZE(cs8415_controls); i++) { + for (i = 0; i < ARRAY_SIZE(cs8415_controls); i++) { struct snd_kcontrol *kctl; err = snd_ctl_add(ice->card, (kctl = snd_ctl_new1(&cs8415_controls[i], ice))); if (err < 0) @@ -1943,7 +1967,7 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) } snd_ice1712_restore_gpio_status(ice); } - + return 0; } @@ -2059,11 +2083,12 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* to remeber the register values of CS8415 */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (! ice->akm) + if (!ice->akm) return -ENOMEM; ice->akm_codecs = 1; - - if ((err = aureon_ac97_init(ice)) != 0) + + err = aureon_ac97_init(ice); + if (err != 0) return err; snd_ice1712_gpio_set_dir(ice, 0x5fffff); /* fix this for the time being */ @@ -2086,7 +2111,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* initialize WM8770 codec */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71 || ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT || - ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT) + ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT) p = wm_inits_prodigy; else p = wm_inits_aureon; @@ -2105,10 +2130,10 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) snd_ice1712_restore_gpio_status(ice); - /* initialize PCA9554 pin directions & set default input*/ + /* initialize PCA9554 pin directions & set default input */ aureon_pca9554_write(ice, PCA9554_DIR, 0x00); aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */ - + spec->master[0] = WM_VOL_MUTE; spec->master[1] = WM_VOL_MUTE; for (i = 0; i < ice->num_total_dacs; i++) { @@ -2215,12 +2240,12 @@ struct snd_ice1712_card_info snd_vt1724_aureon_cards[] __devinitdata = { .eeprom_data = aureon71_eeprom, .driver = "Aureon71", }, - { - .subvendor = VT1724_SUBDEVICE_AUREON71_UNIVERSE, - .name = "Terratec Aureon 7.1-Universe", + { + .subvendor = VT1724_SUBDEVICE_AUREON71_UNIVERSE, + .name = "Terratec Aureon 7.1-Universe", .model = "universe", - .chip_init = aureon_init, - .build_controls = aureon_add_controls, + .chip_init = aureon_init, + .build_controls = aureon_add_controls, .eeprom_size = sizeof(aureon71_universe_eeprom), .eeprom_data = aureon71_universe_eeprom, .driver = "Aureon71Univ", /* keep in 15 letters */

3 2

[alsa-devel] [PATCH] hda: multiple SPDIF outputs support
by Matthew Ranostay 07 Sep '08

07 Sep '08

Added support for outputting a stream to multiple SPDIF outs on supporting codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index 4f32911..696d77e 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -1454,12 +1454,22 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, codec->spdif_ctls = val; if (change) { + hda_nid_t *d; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); + + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); + } } mutex_unlock(&codec->spdif_mutex); @@ -1491,10 +1501,16 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, val |= AC_DIG1_ENABLE; change = codec->spdif_ctls != val; if (change) { + hda_nid_t *d; codec->spdif_ctls = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) @@ -1643,9 +1659,14 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, mutex_lock(&codec->spdif_mutex); change = codec->spdif_in_enable != val; if (change) { + hda_nid_t *d; codec->spdif_in_enable = val; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -2589,15 +2610,30 @@ int snd_hda_input_mux_put(struct hda_codec *codec, static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, unsigned int stream_tag, unsigned int format) { + hda_nid_t *d; + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); /* turn on again (if needed) */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); + + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + } + } /* @@ -2621,8 +2657,12 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int format, struct snd_pcm_substream *substream) { + hda_nid_t *nid; mutex_lock(&codec->spdif_mutex); setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format); + if (codec->slave_dig_outs) + for (nid = codec->slave_dig_outs; *nid; nid++) + setup_dig_out_stream(codec, *nid, stream_tag, format); mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2689,6 +2729,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream) { hda_nid_t *nids = mout->dac_nids; + hda_nid_t *d; int chs = substream->runtime->channels; int i; @@ -2702,9 +2743,16 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, mout->dig_out_used = HDA_DIG_ANALOG_DUP; setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + setup_dig_out_stream(codec, *d, + stream_tag, format); } else { mout->dig_out_used = 0; snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_cleanup_stream(codec, *d); } } mutex_unlock(&codec->spdif_mutex); diff --git a/pci/hda/hda_codec.h b/pci/hda/hda_codec.h index 780e2ff..60468f5 100644 --- a/pci/hda/hda_codec.h +++ b/pci/hda/hda_codec.h @@ -725,6 +725,7 @@ struct hda_codec { unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ + hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_hwdep *hwdep; /* assigned hwdep device */ diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 9968ee4..52b3579 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -229,6 +229,10 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0f, 0x10, 0x11 }; +static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { + 0x26, 0, +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; @@ -274,6 +278,10 @@ static hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; +static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { + 0x1e, 0, +}; + static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x10, 0x40, }; @@ -303,6 +311,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; +static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { + 0x22, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -396,9 +408,10 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[10] = { +static hda_nid_t stac92hd71bxx_pin_nids[11] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x1f, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1424,22 +1437,22 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), }; -static unsigned int ref92hd71bxx_pin_configs[10] = { +static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[10] = { +static unsigned int dell_m4_1_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[10] = { +static unsigned int dell_m4_2_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -3841,8 +3854,9 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; + codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x22, 0x25); if (!err) { if (spec->board_config < 0) { @@ -3883,6 +3897,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; spec->dmic_nids = stac92hd83xxx_dmic_nids; @@ -4033,6 +4048,7 @@ again: case 0x111d76b5: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -4065,6 +4081,7 @@ again: default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } spec->aloopback_mask = 0x20;

2 1

[alsa-devel] HP tx2000 and snd-hda-intel
by Kory Prince 07 Sep '08

07 Sep '08

Note: The other message did not go through because it was too big. I can email the schematics I mentioned if anyone is interested ************************************************************** Hello, I have an HP Pavillion TX2000z. This laptop has uses the snd-hda-intel and has the ALC861VD realtek codec. The existing configuration for the codec does not support the tx2000 series laptops fully. There are several configurations that have various levels of support. If you specify for the model to use the existing hp model (made for the tx1000 series) then: 1.Headphones Work 2.Speakers Work 3.Front Mic Works (the jack mic by the headphone jack) 4.The External Mics do not work (There are two mics at the top of the laptop... I am guessing left and right for a stereo Mic) 5.It seems like there is something mapped to an input that does not belong there. The simplest way to explain this is when I open audacity and set it to monitor input that there is significant white noise coming in. The Ext. Mics are not working so it can't come from there (as far as I know) and I have nothing plugged into the front Mic. I cannot find any way to mute this noise. When using the 3-stack model (with the option position_fix): 1. Headphones do not work (in either ports) 2. Speakers Work 3. Front Mic Works 4. The external mics work 5. Same problem as expressed above All other models do not work in any significant way. Also one of the Headphone ports (there are two on the front) has Digital Out (optical) that I have no way of confirming if it works. Now, that being said, I have tried working on the driver. I have virtually no programing experience. I was able to have the driver autodetect the soundcard and apply a certain model (before it had to be applied with a module config file.) And was able to interpret some of how the driver worked. But I am so lost. I am not sure how hard it would be for a programmer to modify the driver to support the tx2000. So I want to put out a call to anyone who could help. I am fully available for testing and some debugging if someone could try and help. I am uploading the codec file and the codec schematic ( I don't think this is actually accurate, but It may help...) Also here is the lspci -nn for the soundcard and other info I found 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) lshw *-multimedia description: Audio device product: MCP51 High Definition Audio vendor: nVidia Corporation physical id: 10.1 bus info: pci@0000:00:10.1 version: a2 width: 32 bits clock: 66MHz capabilities: pm msi ht bus_master cap_list configuration: driver=HDA Intel latency=0 maxlatency=5 mingnt=2 module=snd_hda_intel If anyone could help or use additional information I would appreciate it. Thanks, Kory p.s. if this is the wrong place to post, please redirect me.

1 0

[alsa-devel] snd_pcm_writei error 128 Key has been revoked
by Paulos Siahu 07 Sep '08

07 Sep '08

Hello I'm a newbie to ALSA, in fact to sound programming at all. I tried a simple program for a beginner. When I got to this line: err = snd_pcm_writei (playback_handle, buf, bufsize); printf("Playing Error: %d %s\n", err, snd_strerror(err)); the first time it played the sound that I wanted then I Ctrl-C the app. From then on, every time I run the program it gives me: Playing Error: 128 Key has been revoked I tried to google around but no luck. Hope someone can help. Thanks Paulos Siahu

1 0

[alsa-devel] Clearification needed about structure _snd_pcm
by William Estrada 06 Sep '08

06 Sep '08

Hi group, I have found the documentation about ALSA very confusing. Where can I find the definitions about the fields in the structure_snd_pcm? Specifically Buffer, Rate, Period, Buffer Size, Period size, etc. Basically all of these: > Device: 8c297b8 > PCM Device name = 'default' PCM state = RUNNING > access type = MMAP_NONINTERLEAV format = 'U8' (Unsigned > 8 bit) > subformat = 'STD' (Standard) channels = 1 > rate = 8000 bps period time = 10000 us > period size = 80 frame buffer time = 0 us > buffer size = 65536 frames periods/buffer = 0 frames > exact rate = 8000/1 bps significant bits = 8000 > is batch = 0 is block transfer = 1 > is double = 0 is half duplex = 0 > is joint duplex = 0 can overrange = 0 > can mmap = 1 can pause = 0 > can resume = 1 can sync start = 1 My program is not honoring the 'packet size'. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] snd_pcm_hw_param_any returns bad data
by William Estrada 06 Sep '08

06 Sep '08

Hi group, I'm using snd_pcm_hw_param_any to retrieve the parameters from an opened device and I'm getting garbage back in the structures. My code is here: http://64.124.13.3/_ALSA_/ This is a simple test to show the problem. The output is shown in show.txt. as you can see the before ( within the open function ) is much different than the after. I am assuming that it is my code that has the problem, any ideas??? Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] [PATCH] ALSA: ASoC V2: optimize init sequence of Freescale MPC8610 sound drivers
by Timur Tabi 06 Sep '08

06 Sep '08

In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Some comment blocks were not near the code they reference, so they were moved. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2 only. sound/soc/fsl/fsl_dma.c | 283 ++++++++++++++++++++++------------------------- sound/soc/fsl/fsl_ssi.c | 20 ++-- 2 files changed, 141 insertions(+), 162 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 5da0069..8821289 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -319,9 +319,66 @@ error: } /** - * fsl_dma_open: open a new substream. + * fsl_dma_open: open a new substream and allocate DMA resources for it * * Each substream has its own DMA buffer. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that cycle from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. */ static int fsl_dma_open(struct snd_pcm_substream *substream) { @@ -370,8 +427,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - ret = request_irq(dma_private->dma_info->irq, - fsl_dma_isr, 0, "DMA", dma_private); + ret = request_irq(dma_private->dma_info->irq, fsl_dma_isr, 0, + "fsl-audio-dma", dma_private); if (ret) { dev_err(substream->pcm->card->dev, "can't register ISR for IRQ %u (ret=%i)\n", @@ -393,17 +450,14 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) temp_link = dma_private->ld_buf_phys + sizeof(struct fsl_dma_link_descriptor); - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); + for (i = 0; i < NUM_DMA_LINKS - 1; i++) { + dma_private->link[i].next = cpu_to_be64(temp_link); temp_link += sizeof(struct fsl_dma_link_descriptor); } /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + dma_private->link[NUM_DMA_LINKS - 1].next = + cpu_to_be64(dma_private->ld_buf_phys); /* Tell the DMA controller where the first link descriptor is */ out_be32(&dma_channel->clndar, @@ -417,11 +471,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) /* * Program the mode register for interrupts, external master control, * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* + * * We want External Master Start and External Master Pause enabled, * because the SSI is controlling the DMA controller. We want the DMA * controller to be set up in advance, and then we signal only the SSI @@ -436,7 +486,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) * We want Error Interrupt enabled, so that we can get an error if * the DMA controller is mis-programmed somehow. */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + mr = CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | CCSR_DMA_MR_EMS_EN; /* For playback, we want the destination address to be held. For @@ -450,78 +500,47 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) } /** - * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors. - * - * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link - * descriptors that ping-pong from one period to the next. For example, if - * there are six periods and two link descriptors, this is how they look - * before playback starts: - * - * The last link descriptor - * ____________ points back to the first - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * | | - * V V - * _________________________________________ - * | | | | | | | The DMA buffer is - * | | | | | | | divided into 6 parts - * |______|______|______|______|______|______| + * fsl_dma_hw_params: program the audio paramters into the DMA controller * - * and here's how they look after the first period is finished playing: + * This function takes the audio parameters (period size, sample size, etc) + * and finishes programming the DMA controller accordingly. * - * ____________ - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * |______________ - * | | - * V V - * _________________________________________ - * | | | | | | | - * | | | | | | | - * |______|______|______|______|______|______| + * The actual address in STX0 (destination for playback, source for capture) + * is based on the sample size. * - * The first link descriptor now points to the third period. The DMA - * controller is currently playing the second period. When it finishes, it - * will jump back to the first descriptor and play the third period. + * One of the drawbacks with big-endian is that when copying integers of + * different sizes to a fixed-sized register, the address to which the + * integer must be copied is dependent on the size of the integer. * - * There are four reasons we do this: + * For example, if P is the address of a 32-bit register, and X is a 32-bit + * integer, then X should be copied to address P. However, if X is a 16-bit + * integer, then it should be copied to P+2. If X is an 8-bit register, + * then it should be copied to P+3. * - * 1. The only way to get the DMA controller to automatically restart the - * transfer when it gets to the end of the buffer is to use chaining - * mode. Basic direct mode doesn't offer that feature. - * 2. We need to receive an interrupt at the end of every period. The DMA - * controller can generate an interrupt at the end of every link transfer - * (aka segment). Making each period into a DMA segment will give us the - * interrupts we need. - * 3. By creating only two link descriptors, regardless of the number of - * periods, we do not need to reallocate the link descriptors if the - * number of periods changes. - * 4. All of the audio data is still stored in a single, contiguous DMA - * buffer, which is what ALSA expects. We're just dividing it into - * contiguous parts, and creating a link descriptor for each one. + * So for playback of 8-bit samples, the DMA controller must transfer single + * bytes from the DMA buffer to the last byte of the STX0 register, i.e. + * offset by 3 bytes. For 16-bit samples, the offset is two bytes. * - * Note that due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we don't program - * the dest_addr (for playback -- source_addr for capture) fields in the - * link descriptors here. We do that in fsl_dma_prepare() + * For 24-bit samples, the offset is 1 byte. However, the DMA controller + * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, + * and 8 bytes at a time). So we do not support packed 24-bit samples. + * 24-bit data must be padded to 32 bits. */ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = + dma_private->dma_info->channel; + + dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ dma_addr_t temp_addr; /* Pointer to next period */ + uint32_t mr; + unsigned int sample_size; /* Number of bits per sample */ + unsigned int frame_size; /* Number of bytes per frame */ unsigned int i; /* Get all the parameters we need */ @@ -537,83 +556,16 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, if (dma_private->dma_buf_next >= dma_private->dma_buf_end) dma_private->dma_buf_next = dma_private->dma_buf_phys; - /* - * The actual address in STX0 (destination for playback, source for - * capture) is based on the sample size, but we don't know the sample - * size in this function, so we'll have to adjust that later. See - * comments in fsl_dma_prepare(). - * - * The DMA controller does not have a cache, so the CPU does not - * need to tell it to flush its cache. However, the DMA - * controller does need to tell the CPU to flush its cache. - * That's what the SNOOP bit does. - * - * Also, even though the DMA controller supports 36-bit addressing, for - * simplicity we currently support only 32-bit addresses for the audio - * buffer itself. - */ - temp_addr = substream->dma_buffer.addr; - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - link->source_addr = cpu_to_be32(temp_addr); - else - link->dest_addr = cpu_to_be32(temp_addr); - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_prepare - prepare the DMA registers for playback. - * - * This function is called after the specifics of the audio data are known, - * i.e. snd_pcm_runtime is initialized. - * - * In this function, we finish programming the registers of the DMA - * controller that are dependent on the sample size. - * - * One of the drawbacks with big-endian is that when copying integers of - * different sizes to a fixed-sized register, the address to which the - * integer must be copied is dependent on the size of the integer. - * - * For example, if P is the address of a 32-bit register, and X is a 32-bit - * integer, then X should be copied to address P. However, if X is a 16-bit - * integer, then it should be copied to P+2. If X is an 8-bit register, - * then it should be copied to P+3. - * - * So for playback of 8-bit samples, the DMA controller must transfer single - * bytes from the DMA buffer to the last byte of the STX0 register, i.e. - * offset by 3 bytes. For 16-bit samples, the offset is two bytes. - * - * For 24-bit samples, the offset is 1 byte. However, the DMA controller - * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, - * and 8 bytes at a time). So we do not support packed 24-bit samples. - * 24-bit data must be padded to 32 bits. - */ -static int fsl_dma_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = - dma_private->dma_info->channel; - uint32_t mr; - unsigned int i; - dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ - unsigned int frame_size; /* Number of bytes per frame */ + /* Program the sample size and I/O address into the DMA controller */ ssi_sxx_phys = dma_private->dma_info->ssi_sxx_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - switch (runtime->sample_bits) { + sample_size = snd_pcm_format_physical_width(params_format(hw_params)); + + switch (sample_size) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -631,7 +583,8 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) return -EINVAL; } - frame_size = runtime->frame_bits / 8; + frame_size = 2 * (sample_size / 8); + /* * BWC should always be a multiple of the frame size. BWC determines * how many bytes are sent/received before the DMA controller checks the @@ -648,16 +601,45 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) out_be32(&dma_channel->mr, mr); - /* - * Program the address of the DMA transfer to/from the SSI. - */ + temp_addr = substream->dma_buffer.addr; + for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->count = cpu_to_be32(period_size); + + /* Even though the DMA controller supports 36-bit addressing, + * for simplicity we allow only 32-bit addresses for the audio + * buffer itself. This was enforced in fsl_dma_new() with the + * DMA mask. + * + * The snoop bit tells the DMA controller whether it should tell + * the ECM to snoop during a read or write to an address. For + * audio, we use DMA to transfer data between memory and an I/O + * device (the SSI's STX0 or SRX0 register). Snooping is only + * needed if there is a cache, so we need to snoop memory + * addresses only. For playback, that means we snoop the source + * but not the destination. For capture, we snoop the + * destination but not the source. + * + * Note that this is unlikely to cause cache incoherency unless + * the DMA buffer is smaller than the size of L1 cache. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link->source_addr = cpu_to_be32(temp_addr); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_addr = cpu_to_be32(ssi_sxx_phys); - else + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + } else { link->source_addr = cpu_to_be32(ssi_sxx_phys); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + + link->dest_addr = cpu_to_be32(temp_addr); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + } + + temp_addr += period_size; } return 0; @@ -800,7 +782,6 @@ static struct snd_pcm_ops fsl_dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_dma_hw_params, .hw_free = fsl_dma_hw_free, - .prepare = fsl_dma_prepare, .pointer = fsl_dma_pointer, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 319cc97..fc52d30 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -344,30 +344,28 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** - * fsl_ssi_prepare: prepare the SSI. + * fsl_ssi_hw_params: program the sample size * * Most of the SSI registers have been programmed in the startup function, - * but the word length must be programmed here. Unfortunately, programming - * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can - * cause a problem with supporting simultaneous playback and capture. If - * the SSI is already playing a stream, then that stream may be temporarily - * stopped when you start capture. + * but the word length must be programmed here. Since we support + * synchronous mode only, only the master stream can program STCRR[WL]. * * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { struct fsl_ssi_info *ssi_info = cpu_dai->private_data; if (substream == ssi_info->master_stream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct ccsr_ssi __iomem *ssi = ssi_info->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); u32 wl; /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); @@ -588,7 +586,7 @@ static struct snd_soc_dai_caps capture = { static struct snd_soc_dai_ops ops = { .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, + .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, -- 1.5.5

2 1

[alsa-devel] HP tx2000 and snd-hda-intel
by Kory Prince 06 Sep '08

06 Sep '08

Hello, I have an HP Pavillion TX2000z. This laptop has uses the snd-hda-intel and has the ALC861VD realtek codec. The existing configuration for the codec does not support the tx2000 series laptops fully. There are several configurations that have various levels of support. If you specify for the model to use the existing hp model (made for the tx1000 series) then: 1.Headphones Work 2.Speakers Work 3.Front Mic Works (the jack mic by the headphone jack) 4.The External Mics do not work (There are two mics at the top of the laptop... I am guessing left and right for a stereo Mic) 5.It seems like there is something mapped to an input that does not belong there. The simplest way to explain this is when I open audacity and set it to monitor input that there is significant white noise coming in. The Ext. Mics are not working so it can't come from there (as far as I know) and I have nothing plugged into the front Mic. I cannot find any way to mute this noise. When using the 3-stack model (with the option position_fix): 1. Headphones do not work (in either ports) 2. Speakers Work 3. Front Mic Works 4. The external mics work 5. Same problem as expressed above All other models do not work in any significant way. Also one of the Headphone ports (there are two on the front) has Digital Out (optical) that I have no way of confirming if it works. Now, that being said, I have tried working on the driver. I have virtually no programing experience. I was able to have the driver autodetect the soundcard and apply a certain model (before it had to be applied with a module config file.) And was able to interpret some of how the driver worked. But I am so lost. I am not sure how hard it would be for a programmer to modify the driver to support the tx2000. So I want to put out a call to anyone who could help. I am fully available for testing and some debugging if someone could try and help. I am uploading the codec file and the codec schematic ( I don't think this is actually accurate, but It may help...) Also here is the lspci -nn for the soundcard and other info I found 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) lshw *-multimedia description: Audio device product: MCP51 High Definition Audio vendor: nVidia Corporation physical id: 10.1 bus info: pci@0000:00:10.1 version: a2 width: 32 bits clock: 66MHz capabilities: pm msi ht bus_master cap_list configuration: driver=HDA Intel latency=0 maxlatency=5 mingnt=2 module=snd_hda_intel If anyone could help or use additional information I would appreciate it. Thanks, Kory p.s. if this is the wrong place to post, please redirect me.

1 0

[alsa-devel] ALSA SOC VI API's for 2 PCM DEVICE ON SAME CARD
by dinesh 05 Sep '08

05 Sep '08

Hi, Actually i am stuck with a problem i have two register 2 channels of my SOC device as 2 different pcm devices on same card. But my driver uses ALSA V1 api's and i am not able to find out any example code for this using v1 api's. (SIMILAR TO MPC5200_PSC.C) If anyone can help me please. Thanks, Dinesh

2 1

[alsa-devel] [PATCH 0/14] ASoC: Blackfin and I2C updats
by Mark Brown 05 Sep '08

05 Sep '08

The following changes since commit 44e2c3045f77c69d18ba4afda888a4cdec4a33fd: Cliff Cai (1): ALSA: ASoC codec: fix compiling error in ad1980 driver after ASoC API changed are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Cliff Cai (10): ASoC codec: SSM2602 audio codec driver ASoC: Blackfin: SPORT peripheral interface driver ASoC: Blackfin: DMA Driver for AC97 sound chip ASoC: Blackfin: AC97 Blackfin CPU DAI driver ASoC: Blackfin: DMA Driver for I2S sound chip ASoC: Blackfin: I2S CPU DAI driver ASoC: Blackfin: board driver for AD1980/1 audio codec ASoC: Blackfin: board driver for SSM2602 sound chip ASoC: Blackfin: add Blackfin arch ASoC Kconfig and Makefile ASoC: Blackfin: Include Blackfin architecture support in build Jean Delvare (4): ASoC: Fix an error path in neo1973_wm8753 ASoC: Convert wm8753 to a new-style i2c driver ASoC: Convert neo1973/lm4857 to a new-style i2c driver ASoC: Convert wm8510 to a new-style i2c driver sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/at32/playpaq_wm8510.c | 1 + sound/soc/blackfin/Kconfig | 85 +++ sound/soc/blackfin/Makefile | 20 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 429 +++++++++++++++ sound/soc/blackfin/bf5xx-ac97-pcm.h | 29 + sound/soc/blackfin/bf5xx-ac97.c | 407 ++++++++++++++ sound/soc/blackfin/bf5xx-ac97.h | 36 ++ sound/soc/blackfin/bf5xx-ad1980.c | 113 ++++ sound/soc/blackfin/bf5xx-i2s-pcm.c | 288 ++++++++++ sound/soc/blackfin/bf5xx-i2s-pcm.h | 29 + sound/soc/blackfin/bf5xx-i2s.c | 292 ++++++++++ sound/soc/blackfin/bf5xx-i2s.h | 14 + sound/soc/blackfin/bf5xx-sport.c | 1032 +++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-sport.h | 192 +++++++ sound/soc/blackfin/bf5xx-ssm2602.c | 186 +++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 776 ++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 130 +++++ sound/soc/codecs/wm8510.c | 110 ++-- sound/soc/codecs/wm8510.h | 1 + sound/soc/codecs/wm8753.c | 106 ++-- sound/soc/codecs/wm8753.h | 1 + sound/soc/s3c24xx/neo1973_wm8753.c | 113 +++-- 26 files changed, 4237 insertions(+), 162 deletions(-) create mode 100644 sound/soc/blackfin/Kconfig create mode 100644 sound/soc/blackfin/Makefile create mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-ac97.c create mode 100644 sound/soc/blackfin/bf5xx-ac97.h create mode 100644 sound/soc/blackfin/bf5xx-ad1980.c create mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-i2s.c create mode 100644 sound/soc/blackfin/bf5xx-i2s.h create mode 100644 sound/soc/blackfin/bf5xx-sport.c create mode 100644 sound/soc/blackfin/bf5xx-sport.h create mode 100644 sound/soc/blackfin/bf5xx-ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h

2 15

[alsa-devel] proper use of snd_mixer_selem_set_capture_switch ?
by Aaron J. Grier 05 Sep '08

05 Sep '08

does anyone have a good example or walkthrough how to change capture sources? I'm having a heck of a time. my calls go like this: /* setup */ snd_mixer_selem_id_alloca snd_mixer_open snd_mixer_attach snd_mixer_selem_register snd_mixer_load /* set up PCM element, unused in this example */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* set up Line element (linecapture) */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* set up Mic element (miccapture) */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* here's that part that throws me... I'm trying to sit in a loop and * trade capture back and forth between Line and Mic. code reproduced * here verbatim... */ do { snd_mixer_selem_channel_id_t chn; for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { int junk; if (snd_mixer_selem_has_capture_channel(linecapture, chn)) { status = snd_mixer_selem_get_capture_switch(linecapture, chn, &junk); assert(0 == status); printf("line, chn %d, junk %d, ", chn, junk); status = snd_mixer_selem_set_capture_switch(linecapture, chn, lineflag); printf("set %d\n", lineflag); assert(0 == status); } if (snd_mixer_selem_has_capture_channel(miccapture, chn)) { status = snd_mixer_selem_get_capture_switch(miccapture, chn, &junk); printf("mic, chn %d, junk %d, ", chn, junk); assert(0 == status); status = snd_mixer_selem_set_capture_switch(miccapture, chn, micflag); printf("set %d\n", micflag); assert(0 == status); } } for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { if (snd_mixer_selem_has_capture_channel(linecapture, chn)) { status = snd_mixer_selem_get_capture_switch(linecapture, chn, &lineflag); printf("read line, chn %d: %d\n", chn, lineflag); assert(0 == status); } if (snd_mixer_selem_has_capture_channel(miccapture, chn)) { status = snd_mixer_selem_get_capture_switch(miccapture, chn, &micflag); printf("read mic, chn %d: %d\n", chn, micflag); assert(0 == status); } } if (lineflag) { lineflag = 0; micflag = 1; } else { lineflag = 1; micflag = 0; } getchar(); } while (1); I get output like this: line, chn 0, junk 0, set 1 mic, chn 0, junk 1, set 0 line, chn 1, junk 1, set 1 mic, chn 1, junk 0, set 0 read line, chn 0: 1 read mic, chn 0: 0 read line, chn 1: 1 read mic, chn 1: 0 [all is well, alsamixer shows LR capture on Line input, smash return and go for another loop] line, chn 0, junk 1, set 0 mic, chn 0, junk 0, set 1 line, chn 1, junk 1, set 0 mic, chn 1, junk 0, set 1 read line, chn 0: 1 read mic, chn 0: 1 read line, chn 1: 1 read mic, chn 1: 1 [here's where it goes funny. the "junk" is what I set last time, as expected, but the readbacks after the set_capture_switch indicate both line and mic are being captured? that's not right. alsamixer shows LR capture on Mic input. smack return again to see what happens.] line, chn 0, junk 1, set 0 mic, chn 0, junk 1, set 1 line, chn 1, junk 1, set 0 mic, chn 1, junk 1, set 1 read line, chn 0: 0 read mic, chn 0: 1 read line, chn 1: 1 read mic, chn 1: 1 [this gets really weird. we make the same write calls we did last time, and now we get back something different? alsamixer shows R Line and L Capture. screwey.] I feel I must be missing something obvious, like a flush operation, or something very simple. who knew changing a mux could be such an involved procedure? (= -- Aaron J. Grier | "Not your ordinary poofy goof." | agrier(a)poofygoof.com

2 2

[alsa-devel] [PATCH 0/4] ARM: OMAP2: Add support for Gumstix Overo (resubmission)
by sakoman＠gmail.com 05 Sep '08

05 Sep '08

This patch series adds support for the Gumstix Overo board. Since it includes an ASoC driver I am copying both linux-omap and alsa-devel This version incorporates feedback from both linux-omap and alsa-devel

5 21

[alsa-devel] [PATCH 1/1] ASoC codec: SSM2602 audio codec driver (v3)
by Bryan Wu 05 Sep '08

05 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v2-v3: - add the codec to SND_SOC_ALL_CODECS - coding style fixing - rename registers' name - fix an issue with DAPM and the bias configuration. v1-v2: - coding style fixing - use pr_xxx macros to replace printk(KERN_XXX...) - use new-style i2c interface - update to use latest ASoC API Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 775 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 130 ++++++++ 4 files changed, 911 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5d77dc3..2223993 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X + select SND_SOC_SSM2602 help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -92,3 +93,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_SSM2602 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 35daaa9..0cd55ee 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-ssm2602-objs := ssm2602.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o @@ -33,3 +34,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c new file mode 100644 index 0000000..d43459f --- /dev/null +++ b/sound/soc/codecs/ssm2602.c @@ -0,0 +1,775 @@ +/* + * File: sound/soc/codecs/ssm2602.c + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * Description: Driver for ssm2602 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ssm2602.h" + +#define AUDIO_NAME "ssm2602" +#define SSM2602_VERSION "0.1" + +struct snd_soc_codec_device soc_codec_dev_ssm2602; + +/* codec private data */ +struct ssm2602_priv { + unsigned int sysclk; + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* + * ssm2602 register cache + * We can't read the ssm2602 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { + 0x0017, 0x0017, 0x0079, 0x0079, + 0x0000, 0x0000, 0x0000, 0x000a, + 0x0000, 0x0000 +}; + +/* + * read ssm2602 register cache + */ +static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == SSM2602_RESET) + return 0; + if (reg >= SSM2602_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ssm2602 register cache + */ +static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= SSM2602_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the ssm2602 register space + */ +static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 ssm2602 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + ssm2602_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0) + +/*Appending several "None"s just for OSS mixer use*/ +static const char *ssm2602_input_select[] = { + "Line", "Mic", "None", "None", "None", + "None", "None", "None", +}; + +static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum ssm2602_enum[] = { + SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select), + SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), +}; + +static const struct snd_kcontrol_new ssm2602_snd_controls[] = { + +SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, + 0, 127, 0), +SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V, + 7, 1, 0), + +SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), + +SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), +SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), + +SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), + +SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), + +SOC_ENUM("Capture Source", ssm2602_enum[0]), + +SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), +}; + +/* add non dapm controls */ +static int ssm2602_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Output Mixer */ +static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ssm2602_input_mux_controls = +SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); + +static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, + &ssm2602_output_mixer_controls[0], + ARRAY_SIZE(ssm2602_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), +SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), +SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("RLINEIN"), +SND_SOC_DAPM_INPUT("LLINEIN"), +}; + +static const struct snd_soc_dapm_route audio_conn[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Line", "Line Input"}, + {"Input Mux", "Mic", "Mic Bias"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, + {"Mic Bias", NULL, "MICIN"}, +}; + +static int ssm2602_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets, + ARRAY_SIZE(ssm2602_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return i; +} + +static int ssm2602_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + u16 srate; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; + int i = get_coeff(ssm2602->sysclk, params_rate(params)); + + /*no match is found*/ + if (i == ARRAY_SIZE(coeff_div)) + return -EINVAL; + + srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + ssm2602_write(codec, SSM2602_IFACE, iface); + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); + return 0; +} + +static int ssm2602_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + /* The DAI has shared clocks so if we already have a playback or + * capture going then constrain this substream to match it. + */ + if (ssm2602->master_substream) { + master_runtime = ssm2602->master_substream->runtime; + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + ssm2602->slave_substream = substream; + } else + ssm2602->master_substream = substream; + + return 0; +} + +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* set active */ + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); + + return 0; +} + +static void ssm2602_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* deactivate */ + if (!codec->active) + ssm2602_write(codec, SSM2602_ACTIVE, 0); +} + +static int ssm2602_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) + ssm2602_write(codec, SSM2602_APDIGI, + mute_reg | APDIGI_ENABLE_DAC_MUTE); + else + ssm2602_write(codec, SSM2602_APDIGI, mute_reg); + return 0; +} + +static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2602->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + ssm2602_write(codec, SSM2602_IFACE, iface); + return 0; +} + +static int ssm2602_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + ssm2602_write(codec, SSM2602_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + ssm2602_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_PWR, 0xffff); + break; + + } + codec->bias_level = level; + return 0; +} + +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +struct snd_soc_dai ssm2602_dai = { + .name = "SSM2602", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + }, + .dai_ops = { + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(ssm2602_dai); + +static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2602_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(ssm2602_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ssm2602_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the ssm2602 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ssm2602_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int reg, ret = 0; + + codec->name = "SSM2602"; + codec->owner = THIS_MODULE; + codec->read = ssm2602_read_reg_cache; + codec->write = ssm2602_write; + codec->set_bias_level = ssm2602_set_bias_level; + codec->dai = &ssm2602_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ssm2602_reg); + codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + ssm2602_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("ssm2602: failed to create pcms\n"); + goto pcm_err; + } + /*power on device*/ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + /* set the update bits */ + reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL); + ssm2602_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL); + ssm2602_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V); + ssm2602_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); + ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + /*select Line in as default input*/ + ssm2602_write(codec, SSM2602_APANA, + APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); + ssm2602_write(codec, SSM2602_PWR, 0); + + ssm2602_add_controls(codec); + ssm2602_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("ssm2602: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *ssm2602_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = ssm2602_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = ssm2602_init(socdev); + if (ret < 0) + pr_err("failed to initialise SSM2602\n"); + + return ret; +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +/* corgi i2c codec control layer */ +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "SSM2602 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; + +static int ssm2602_add_i2c_device(struct platform_device *pdev, + const struct ssm2602_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ssm2602_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ssm2602", I2C_NAME_SIZE); + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + return 0; +err_driver: + i2c_del_driver(&ssm2602_i2c_driver); + return -ENODEV; +} + +#endif + +static int ssm2602_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ssm2602_setup_data *setup; + struct snd_soc_codec *codec; + struct ssm2602_priv *ssm2602; + int ret = 0; + + pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + if (ssm2602 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ssm2602; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ssm2602_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = ssm2602_add_i2c_device(pdev, setup); + } +#else + /* other interfaces */ +#endif + return ret; +} + +/* remove everything here */ +static int ssm2602_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&ssm2602_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ssm2602 = { + .probe = ssm2602_probe, + .remove = ssm2602_remove, + .suspend = ssm2602_suspend, + .resume = ssm2602_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); + +MODULE_DESCRIPTION("ASoC ssm2602 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h new file mode 100644 index 0000000..f344e6d --- /dev/null +++ b/sound/soc/codecs/ssm2602.h @@ -0,0 +1,130 @@ +/* + * File: sound/soc/codecs/ssm2602.h + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef _SSM2602_H +#define _SSM2602_H + +/* SSM2602 Codec Register definitions */ + +#define SSM2602_LINVOL 0x00 +#define SSM2602_RINVOL 0x01 +#define SSM2602_LOUT1V 0x02 +#define SSM2602_ROUT1V 0x03 +#define SSM2602_APANA 0x04 +#define SSM2602_APDIGI 0x05 +#define SSM2602_PWR 0x06 +#define SSM2602_IFACE 0x07 +#define SSM2602_SRATE 0x08 +#define SSM2602_ACTIVE 0x09 +#define SSM2602_RESET 0x0f + +/*SSM2602 Codec Register Field definitions + *(Mask value to extract the corresponding Register field) + */ + +/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/ +#define LINVOL_LIN_VOL 0x01F /* Left Channel PGA Volume control */ +#define LINVOL_LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */ +#define LINVOL_LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */ + +/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/ +#define RINVOL_RIN_VOL 0x01F /* Right Channel PGA Volume control */ +#define RINVOL_RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */ +#define RINVOL_RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */ + +/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/ +#define LOUT1V_LHP_VOL 0x07F /* Left Channel Headphone volume control */ +#define LOUT1V_ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */ +#define LOUT1V_LRHP_BOTH 0x100 /* Left Channel Headphone volume update */ + +/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/ +#define ROUT1V_RHP_VOL 0x07F /* Right Channel Headphone volume control */ +#define ROUT1V_ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */ +#define ROUT1V_RLHP_BOTH 0x100 /* Right Channel Headphone volume update */ + +/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/ +#define APANA_ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */ +#define APANA_ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */ +#define APANA_ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */ +#define APANA_ENABLE_BYPASS 0x008 /* Line input bypass to line output */ +#define APANA_SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */ +#define APANA_ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */ +#define APANA_SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */ +#define APANA_ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */ + +/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/ +#define APDIGI_ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */ +#define APDIGI_DE_EMPHASIS 0x006 /* De-Emphasis Control */ +#define APDIGI_ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */ +#define APDIGI_STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */ + +/*Power Down Control (SSM2602_REG_POWER) + *(1=Enable PowerDown, 0=Disable PowerDown) + */ +#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */ +#define PWR_MIC_PDN 0x002 /* Microphone Input & Bias Power Down */ +#define PWR_ADC_PDN 0x004 /* ADC Power Down */ +#define PWR_DAC_PDN 0x008 /* DAC Power Down */ +#define PWR_OUT_PDN 0x010 /* Outputs Power Down */ +#define PWR_OSC_PDN 0x020 /* Oscillator Power Down */ +#define PWR_CLK_OUT_PDN 0x040 /* CLKOUT Power Down */ +#define PWR_POWER_OFF 0x080 /* POWEROFF Mode */ + +/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/ +#define IFACE_IFACE_FORMAT 0x003 /* Digital Audio input format control */ +#define IFACE_AUDIO_DATA_LEN 0x00C /* Audio Data word length control */ +#define IFACE_DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */ +#define IFACE_DAC_LR_SWAP 0x020 /* Swap DAC data control */ +#define IFACE_ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */ +#define IFACE_BCLK_INVERT 0x080 /* Bit Clock Inversion control */ + +/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/ +#define SRATE_ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */ +#define SRATE_BOS_RATE 0x002 /* Base Over-Sampling rate */ +#define SRATE_SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */ +#define SRATE_CORECLK_DIV2 0x040 /* Core Clock divider select */ +#define SRATE_CLKOUT_DIV2 0x080 /* Clock Out divider select */ + +/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/ +#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */ + +/*********************************************************************/ + +#define SSM2602_CACHEREGNUM 10 + +#define SSM2602_SYSCLK 0 +#define SSM2602_DAI 0 + +struct ssm2602_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ssm2602_dai; +extern struct snd_soc_codec_device soc_codec_dev_ssm2602; + +#endif -- 1.5.6

2 1

[alsa-devel] [PATCH] ALSA: ASoC: Fix memory leaks for the CS4270 driver
by Timur Tabi 05 Sep '08

05 Sep '08

The CS4270 ASoC driver was not cleaning up resources completely during an I2C unbind. The same problem could occur if the bind failed. Rearranged a couple functions to eliminate an unnecessary function declaration. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is a fix for 2.6.27. It applies only to the ASoC V1 version of the driver. A similar fix for the V2 version will be posted later. sound/soc/codecs/cs4270.c | 94 ++++++++++++++++++++++++++++++-------------- 1 files changed, 64 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 82d94f0..eb38e92 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -490,29 +490,6 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) #endif -static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); - -/* A list of non-DAPM controls that the CS4270 supports */ -static const struct snd_kcontrol_new cs4270_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) -}; - -static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4270_id); - -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "CS4270 I2C", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, -}; - /* * Global variable to store socdev for i2c probe function. * @@ -530,6 +507,12 @@ static struct i2c_driver cs4270_i2c_driver = { */ static struct snd_soc_device *cs4270_socdev; +/* A list of non-DAPM controls that the CS4270 supports */ +static const struct snd_kcontrol_new cs4270_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) +}; + /* * Initialize the I2C interface of the CS4270 * @@ -559,8 +542,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); if (!codec->reg_cache) { printk(KERN_ERR "cs4270: could not allocate register cache\n"); - ret = -ENOMEM; - goto error; + return -ENOMEM; } /* Verify that we have a CS4270 */ @@ -610,20 +592,72 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, return 0; error: - if (codec->control_data) { - i2c_detach_client(i2c_client); - codec->control_data = NULL; + for (i = 1; i <= ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_ctl_find_numid(codec->card, i); + + if (kctrl) + snd_ctl_remove(codec->card, kctrl); } + codec->control_data = NULL; + kfree(codec->reg_cache); codec->reg_cache = NULL; codec->reg_cache_size = 0; - kfree(i2c_client); - return ret; } +/* + * Remove I2C interface of the CS4270 + * + * Since this driver cannot be compiled as a module, the only way this + * function can be called is if the I2C subsystem unbinds it. This can be + * triggered, for instance, by writing the device ID to this drivers + * 'unbind' file in /sys. + */ +static int cs4270_i2c_remove(struct i2c_client *i2c_client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(i2c_client); + + if (codec) { + unsigned int i; + + for (i = 1; i <= ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_ctl_find_numid(codec->card, i); + + if (kctrl) + snd_ctl_remove(codec->card, kctrl); + } + + codec->control_data = NULL; + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + codec->reg_cache_size = 0; + } + + return 0; +} + +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, +}; + #endif /* USE_I2C*/ struct snd_soc_dai cs4270_dai = { -- 1.5.5

2 9

[alsa-devel] [PATCH 0/9] ASoC Blackfin supporting (v2)
by Bryan Wu 05 Sep '08

05 Sep '08

Hi Mark, With Cliff's effort, we update this patch series quickly according to your review. v1-v2: - move ASoC Blackfin Kconfig and Makefile patch to be the last one in the series - fix coding style issues - The SND_SOC_DAFIMT_LEFT_J: ought to be default - fix other minor issues v0-v1: - fix coding style issues - use latest ASoC API - split the whole patch into this 9 patches in a patchset Thanks a lot -Bryan

2 14

[alsa-devel] Problems recording with HP DV2839TX laptop.
by Travis Place 05 Sep '08

05 Sep '08

Hello everyone, Im Travis (wishie) once of the many helpers who man the unofficial #alsa on freenode irc network.. I usually spend my time fixing other peoples sound issues, but now i have my own. I have a HP DV2839TX laptop, and i get ALOT of static when recording via the inbuilt mic. Also, the mic controls are very strange.. I have (in the Capture section of alsamixer).. Digital, Docking Mic, External Mic, Internal Mic.. only the "Digital" control seems to effect my recording volume at all. Strangely, though, the '* Mic' controls in the PLAYBACK section of alsamixer do change the recording properties somewhat (volume being the main factor). In any event, i get a rather large amount of static, no matter what mixer settings i try. This isnt a problem in the 'other' OS. My alsa-info.sh output for alsa-driver 1.0.16 is at : http://www.alsa-project.org/db/?f=0d1df6446ff1c71ad305e7b7b367f0a6b22b80c8 My alsa-info.sh output for alsa-driver 1.0.18rc1 is at : http://www.alsa-project.org/db/?f=9e5cd99f4822e0d671866fa630698bd1b0cd954b

2 2

[alsa-devel] WM8750 mixer controls
by Harald Radke 05 Sep '08

05 Sep '08

Hi there! I noticed that using "alsamixer" several controls that are connected to sound input (ALC) are listed in the "playback" tab as well as in "capture", as well as certain playback controls as "bass" "trebel" also can be found under "capture". However, changing those controls' values only seem to take effect in the "playback" tab. I was wondering if this is intentional or unintentional done by the WM8750 codec driver or an alsamixer "feature". Thx Greez, Harry

2 1

Re: [alsa-devel] [PATCH] add the nvidia HDMI codec driver for MCP77/79
by Takashi Iwai 05 Sep '08

05 Sep '08

At Wed, 3 Sep 2008 15:58:17 +0800, peerchen wrote: > > Add the nvidia HDMI codec driver for MCP77/79. > The patch based on kernel 2.6.27-rc5. > > Signed-off-by: Wei Ni <wni(a)nvidia.com> > Signed-off-by: Peer Chen <peerchen(a)gmail.com> Thanks for the patch. Some comments below. > diff -uprN -X linux-2.6.26-Orig/Documentation/dontdiff linux-2.6.26-Orig/sound/pci/hda/hda_intel.c linux-2.6.26-New/sound/pci/hda/hda_intel.c > --- linux-2.6.26-Orig/sound/pci/hda/hda_intel.c 2008-08-28 13:47:20.000000000 +0800 > +++ linux-2.6.26-New/sound/pci/hda/hda_intel.c 2008-08-28 15:33:51.000000000 +0800 > @@ -223,8 +223,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO > #define RIRB_INT_MASK 0x05 > > /* STATESTS int mask: SD2,SD1,SD0 */ > -#define AZX_MAX_CODECS 3 > -#define STATESTS_INT_MASK 0x07 > +#define AZX_MAX_CODECS 4 > +#define STATESTS_INT_MASK 0x0f This may break other platforms. Expanding STATESTS_INT_MASK is OK and would be harmless, but AZX_MAX_CODECS is not. AZX_MAX_CODECS is the default number of max codecs. The chipset specific max number of codecs is defined in azx_max_codecs[]. If Nvidia chipsets supports properly more than 3, change azx_max_codecs[AZX_DRIVER_NVIDIA] to 4. > +static unsigned short convert_from_spdif_status_nv(unsigned int sbits) > +{ > + unsigned short val = 0; > + > + if (sbits & IEC958_AES0_PROFESSIONAL) > + val |= AC_DIG1_PROFESSIONAL; > + if (sbits & IEC958_AES0_NONAUDIO) > + val |= AC_DIG1_NONAUDIO; > + if (sbits & IEC958_AES0_PROFESSIONAL) { > + if ((sbits & IEC958_AES0_PRO_EMPHASIS) == > + IEC958_AES0_PRO_EMPHASIS_5015) > + val |= AC_DIG1_EMPHASIS; > + } else { > + if ((sbits & IEC958_AES0_CON_EMPHASIS) == > + IEC958_AES0_CON_EMPHASIS_5015) > + val |= AC_DIG1_EMPHASIS; > + if (!(sbits & IEC958_AES0_CON_NOT_COPYRIGHT)) > + val |= AC_DIG1_COPYRIGHT; > + if (sbits & (IEC958_AES1_CON_ORIGINAL << 8)) > + val |= AC_DIG1_LEVEL; > + val |= sbits & (IEC958_AES1_CON_CATEGORY << 8); > + } > + return val; > +} > + > +static unsigned int convert_to_spdif_status_nv(unsigned short val) > +{ > + unsigned int sbits = 0; > + > + if (val & AC_DIG1_NONAUDIO) > + sbits |= IEC958_AES0_NONAUDIO; > + if (val & AC_DIG1_PROFESSIONAL) > + sbits |= IEC958_AES0_PROFESSIONAL; > + if (sbits & IEC958_AES0_PROFESSIONAL) { > + if (sbits & AC_DIG1_EMPHASIS) > + sbits |= IEC958_AES0_PRO_EMPHASIS_5015; > + } else { > + if (val & AC_DIG1_EMPHASIS) > + sbits |= IEC958_AES0_CON_EMPHASIS_5015; > + if (!(val & AC_DIG1_COPYRIGHT)) > + sbits |= IEC958_AES0_CON_NOT_COPYRIGHT; > + if (val & AC_DIG1_LEVEL) > + sbits |= (IEC958_AES1_CON_ORIGINAL << 8); > + sbits |= val & (0x7f << 8); > + } > + return sbits; > +} > + > +static int snd_hda_hdmi_cmask_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | > + IEC958_AES0_NONAUDIO | > + IEC958_AES0_CON_EMPHASIS_5015 | > + IEC958_AES0_CON_NOT_COPYRIGHT; > + ucontrol->value.iec958.status[1] = IEC958_AES1_CON_CATEGORY | > + IEC958_AES1_CON_ORIGINAL; > + return 0; > +} > + > +static int snd_hda_hdmi_pmask_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | > + IEC958_AES0_NONAUDIO | > + IEC958_AES0_PRO_EMPHASIS_5015; > + return 0; > +} > + > +static int snd_hda_hdmi_mask_info(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_info *uinfo) > +{ > + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; > + uinfo->count = 1; > + return 0; > +} > + > +static int snd_hda_hdmi_default_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + > + ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff; > + ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff; > + ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff; > + ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff; > + > + return 0; > +} > + > +static int snd_hda_hdmi_default_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + hda_nid_t nid = kcontrol->private_value; > + unsigned short val; > + int change; > + > + mutex_lock(&codec->spdif_mutex); > + codec->spdif_status = ucontrol->value.iec958.status[0] | > + ((unsigned int)ucontrol->value.iec958.status[1] << 8) | > + ((unsigned int)ucontrol->value.iec958.status[2] << 16) | > + ((unsigned int)ucontrol->value.iec958.status[3] << 24); > + val = convert_from_spdif_status_nv(codec->spdif_status); > + val |= codec->spdif_ctls & 1; > + change = codec->spdif_ctls != val; > + codec->spdif_ctls = val; > + > + if (change) { > + snd_hda_codec_write_cache(codec, nid, 0, > + AC_VERB_SET_DIGI_CONVERT_1, > + val & 0xff); > + snd_hda_codec_write_cache(codec, nid, 0, > + AC_VERB_SET_DIGI_CONVERT_2, > + val >> 8); > + } > + > + mutex_unlock(&codec->spdif_mutex); > + return change; > +} > + > +#define snd_hda_hdmi_out_switch_info snd_ctl_boolean_mono_info > + > +static int snd_hda_hdmi_out_switch_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + > + ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE; > + return 0; > +} > + > +static int snd_hda_hdmi_out_switch_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); > + hda_nid_t nid = kcontrol->private_value; > + unsigned short val; > + int change; > + > + mutex_lock(&codec->spdif_mutex); > + val = codec->spdif_ctls & ~AC_DIG1_ENABLE; > + if (ucontrol->value.integer.value[0]) > + val |= AC_DIG1_ENABLE; > + change = codec->spdif_ctls != val; > + if (change) { > + codec->spdif_ctls = val; > + snd_hda_codec_write_cache(codec, nid, 0, > + AC_VERB_SET_DIGI_CONVERT_1, > + val & 0xff); > + /* unmute amp switch (if any) */ > + if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && > + (val & AC_DIG1_ENABLE)) > + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, > + HDA_AMP_MUTE, 0); > + } > + mutex_unlock(&codec->spdif_mutex); > + return change; > +} These are almost identical with the functions in hda_codec.c. Is your intention to allow a different set of IEC958 status bits controls for HDMI *in addition* to SPDIF? If so, we should change the codes in hda_codec.c to allow other control names. > +struct snd_kcontrol_new hdmi_mixes[] = { > + { > + .access = SNDRV_CTL_ELEM_ACCESS_READ, > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "IEC958 Playback Con Mask HDMI", > + .info = snd_hda_hdmi_mask_info, > + .get = snd_hda_hdmi_cmask_get, > + }, These have also different names from the standard ones. Any reason? > +static struct hda_pcm_stream nvhdmi_pcm_digital_playback = { > + .substreams = 1, > + .channels_min = 2, > + .channels_max = 2, > + .nid = 0x4, /* NID to query formats and rates and setup streams */ > + .rates = SNDRV_PCM_RATE_48000, > + .maxbps = 16, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, Use tabs please. > +static int nvhdmi_build_pcms(struct hda_codec *codec) > +{ > + struct nvhdmi_spec *spec = codec->spec; > + struct hda_pcm *info = &spec->pcm_rec; > + > + codec->num_pcms = 1; > + codec->pcm_info = info; > + > + info->name = "NVIDIA HDMI"; > + info->pcm_type = HDA_PCM_TYPE_HDMI Ditto. thanks, Takashi

3 5

Re: [alsa-devel] noise and consume pcm faster
by Nie Jun 05 Sep '08

05 Sep '08

It is strange that there is no noise in the first several seconds. It seems like a buffer overflow with too much data accumulated. However, there should not be overflow in playback, right? 2008/9/5 Nie Jun <niej0001(a)gmail.com> > I do not know how these two params will be modified by ALSA lib. It does > not directly accept specified value. > With this cmd: > ./aplay 48k_stero.wav -F5120 -B106400 > I get the nearest setting as following, but still has noise. > > channels: 2 > rate: 48000 (48000/1) > period_size: 248 > buffer_size: 5208 > > > > 2008/9/5 Takashi Iwai <tiwai(a)suse.de> > > At Fri, 5 Sep 2008 14:34:51 +0800, >> Nie Jun wrote: >> > >> > I can not figure out the reason of noise and faster consumption from >> these >> > different. Do you have any idea about these different? Or any other data >> may >> > contain clue? >> >> So, do you mean that the problem persists even if you use the exactly >> same period_size and buffer_size as OSS (i.e. 256 and 5120)? >> >> >> Takashi >> > >

1 0

[alsa-devel] Making envy24control/kenvy24 work on Envy24 HT/HT-S cards
by Vedran Miletić 05 Sep '08

05 Sep '08

Hi, any ideas if it would be possible to make envy24control or kenvy24 work on HT/HT-S cards? I'm referring to VU-meters / mixer / settings / profiles part, not the routing since those chipsets don't support it, according to VIA. -- Vedran Miletić

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Re: [alsa-devel] noise and consume pcm faster
by Takashi Iwai 05 Sep '08

05 Sep '08

At Fri, 5 Sep 2008 14:34:51 +0800, Nie Jun wrote: > > I can not figure out the reason of noise and faster consumption from these > different. Do you have any idea about these different? Or any other data may > contain clue? So, do you mean that the problem persists even if you use the exactly same period_size and buffer_size as OSS (i.e. 256 and 5120)? Takashi

1 0

[alsa-devel] Playing silence instead of pausing during underrun
by Timur Tabi 05 Sep '08

05 Sep '08

If I understand it correctly, when ALSA detects an underrun during playback, it throttles the driver by sending pause and resume commands. It turns out that our hardware doesn't really like this. We're looking at various solutions, but one proposal would be to keep the hardware running during a pause, but just have it play silence instead. That is, ALSA keeps feeding silence data during an underrun recovery. Is there an easy way to do this in ALSA? Is there another driver that does this? -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] noise and consume pcm faster
by Nie Jun 05 Sep '08

05 Sep '08

Thank Takashi for your advice. I check /proc/asound/card0/pcm0p/sub0/* files, buffer_sizes of both ALSA and OSS have integer period_size. Maybe other setting causes this issue, but I can not find it. Would you guys please help have a look at these file? Thank you very much! =========== ALSA hw_params =========== access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 480 buffer_size: 9600 ----------------- OSS hw_params --------------------- access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 5120 OSS format: S16_LE OSS channels: 2 OSS rate: 48000 OSS period bytes: 1024 OSS periods: 20 OSS period frames: 256 =========== ALSA sw_params =========== tstamp_mode: NONE period_step: 1 avail_min: 480 start_threshold: 9600 stop_threshold: 9600 silence_threshold: 0 silence_size: 0 boundary: 1258291200 ----------------- OSS sw_params --------------------- tstamp_mode: NONE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 5120 silence_threshold: 272 silence_size: 272 boundary: 1342177280 =========== ALSA status ========== state: RUNNING trigger_time: 946744218.914947510 tstamp : 946744304.440704346 delay : 8924 avail : 676 avail_max : 1940 ----- hw_ptr : 4105180 appl_ptr : 4114104 ---------------------- OSS status -------------------------- state: RUNNING trigger_time: 946744382.661376953 tstamp : 946744410.868988037 delay : 4908 avail : 212 avail_max : 4876 ----- hw_ptr : 1353952 appl_ptr : 1358860 ============= ALSA OSS has same info ----------------------------- card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: I2S HiFi I2S HiFi-PCM-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 2008/9/3 Takashi Iwai <tiwai(a)suse.de>:- 显示引用文字 -

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Re: [alsa-devel] are freebob firewire cards likely to be supported by alsa in the future?
by Jonathan Woithe 05 Sep '08

05 Sep '08

Hi Andy > Just wondering if there are any plans to develop alsa support for > firewire freebob cards like the M-Audio firewire solo (which I have). I > have this working fine under jackd using the freebob driver, but as far > as I know there isn't an alsa driver for it. I assume that since there > is already open source code to drive the chip, it should be possible to > write an alsa driver? Yes, it would be possible. However, doing so would require that almost everything freebob/ffado currently does (which is a lot) would have to be ported and/or reimplemented. This doesn't seem like the best use of anyone's time. Instead, there are plans to write an alsa driver interface to libffado and some preliminary work has been done, but there's no ETA on this at present. Currently our (ffado's) focus is on getting things working well under JACK because users of firewire devices will also tend to be JACK users due to the nature of the firewire devices and the kind of work one normally does with them. > This isn't an urgent thing for me - for the moment I'm quite happy to use > my onboard card for playing music / movies etc, and keep the firewire card > for recording music etc, but it might be useful sometimes to have an alsa > driver for this card as well. Many media players are capable of connecting to JACK. xine, ecasound and mplayer are a few which come to mind off the top of my head. If your chosen applications don't do JACK, then you might have some success with oss2jack assuming your applications can use OSS devices (I haven't personally tried it so I don't know how well it works). There might even be an equivalent "alsa2jack" bridge but I haven't seen it if there is. Obviously your best bet would be to find media players which support JACK natively. Regards jonathan

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[alsa-devel] alsa-lib, dmix, and snd_pcm_open_conf function
by Simon Tremblay 05 Sep '08

05 Sep '08

Hi, I have a problem using dmix on my embedded system (manual installation of alsa-lib 1.0.15) When I play a sound with aplay, it works well. When I try -D dmix or -D plug:dmix parameter, it doesn't work. The output is: aplay: main:547: audio open error: Operation not permitted I located my problem in alsa-lib, around line 2154 of alsa-lib/src/pcm/pcm.c: err = snd_pcm_open_conf(pcmp, name, root, pcm_conf, stream, mode); (in snd_pcm_open_noupdate function) err result is -1 when it's used with dmix... I don't know how works alsa-lib. I suppose snd_pcm_open_conf has something related with configuration files... Since I have all default configuration files... Maybe I must tweak some to be able to use dmix...

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[alsa-devel] Using GDB for ALSA How to?
by William Estrada 05 Sep '08

05 Sep '08

Hi group, I am trying to use GDB with my ALSA program. I wish to display (print) a handle for an snd_pcm_t pointer. I am getting the following error: > (gdb) p Device > $9 = (snd_pcm_t *) 0x92be458 > (gdb) p *Device > $10 = <incomplete type> How do I display the snd_pcm_t structure? Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] are freebob firewire cards likely to be supported by alsa in the future?
by andy baxter 05 Sep '08

05 Sep '08

Hello, Just wondering if there are any plans to develop alsa support for firewire freebob cards like the M-Audio firewire solo (which I have). I have this working fine under jackd using the freebob driver, but as far as I know there isn't an alsa driver for it. I assume that since there is already open source code to drive the chip, it should be possible to write an alsa driver? This isn't an urgent thing for me - for the moment I'm quite happy to use my onboard card for playing music / movies etc, and keep the firewire card for recording music etc, but it might be useful sometimes to have an alsa driver for this card as well. andy

1 0

[alsa-devel] [ANNOUNCE] Git User's Survey 2008
by Jakub Narebski 04 Sep '08

04 Sep '08

Hi all, We would like to ask you a few questions about your use of the Git version control system. This survey is mainly to understand who is using Git, how and why. The results will be published on the Git Wiki http://git.or.cz/gitwiki/GitSurvey2008 and discussed on the git mailing list. The survey would be open from 1 September till 10 October. Please devote a bit of your time to fill this simple questionnaire; it will help a lot the Git community to understand your needs, what you like of Git, and of course what you don't like of it. The survey can be found here: http://www.survs.com/survey?id=M3PIVU72&channel=2WXE4BVTW8 http://tinyurl.com/GitSurvey2008 -- Git Development Community

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Re: [alsa-devel] snd-soc-cs4270: Convert to a new-style i2c driver (work in progress)
by Jean Delvare 04 Sep '08

04 Sep '08

Hi Timur, On Sun, 31 Aug 2008 11:28:08 -0500, Timur Tabi wrote: > On Sun, Aug 31, 2008 at 9:18 AM, Jean Delvare <khali(a)linux-fr.org> wrote: > > Hi Tibur, > > It's "Timur" Oops, apparently I mixed your first name and last name. Sorry about that. > > I am in the process of converting your cs4270 codec driver from the > > legacy i2c model to the new (standard) one. > > It has already been converted. > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=0d… Ah, excellent. One less thing for me to do :) Thanks! Two comments about your patch: * Your driver now lacks a remove method. Unless I miss something, if the snd-soc-cs4270 driver is unloaded, you will leave dangling resources behind (codec->reg_cache in particular.) * I2C_DRIVERID_CS4270 must be removed from i2c-id.h. > I don't know why this commit hasn't been pushed upstream, though. As I understand it, without that patch the mpc8610_hpcd doesn't work, as the I2C address of the sound codec will be made busy by the platform code and thus the snd-soc-cs4270 driver won't be able to attach to it. If I am correct then I suggest that you ask Takashi to push the patch to Linus now to fix that. > > This is work in progress. > > The patch below converts the cs4270 driver itself. However we also need > > to convert its users. As far as I can see there's only one user at this > > point: mpc8610_hpcd. > > That's correct. I've never seen any interest in the CS4270 outside of > Freescale. > > > The problem is that this driver doesn't look like the other codec > > drivers I have already converted. So, we need to add code to > > instantiate the cs4270 i2c device, but I don't know where this should > > happen. Given that the mpc8610_hpcd is apparently based on Open > > Firmware, I guess that the i2c device should be instantiated directly > > by the platform code. I see that the device is declared in > > mpc8610_hpcd.dts, so maybe it's already done and my patch should work > > already? What do you think? > > I think you need to use the right git repository for your development. :-) Very good point indeed. I'll make sure to check Takashi's sound-2.6 tree before attempting to convert any other SoC codec driver. Do you know off the top of your head if other drivers have already been converted? Thanks, -- Jean Delvare

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[alsa-devel] Pull request for alsa-plugins
by Lennart Poettering 04 Sep '08

04 Sep '08

Takashi, Jaroslav, please pull and merge a series of 26 patches for alsa-plugins' PulseAudio driver which I prepared in this repository of mine: git://git.0pointer.de/alsa-plugins.git gitweb is here: http://git.0pointer.de/?p=alsa-plugins.git There are quite a few bug fixes and feature enhancements among those patches. It also includes some clean-ups in the coding style. i.e. previously the indenting of the PA driver was pure chaos, every single patch seemed to introduce a new indentation width. The first patch in this series simply reindents the driver to Linux style. All other patches are on top of that one. Here's the shortlog: Lennart Poettering (26): Reindent to Linux kernel style Add Emacs-style /*-*- linux-c -*-*/ header comment Make pulse_new() a proper C function Don't modify the SIGPIPE handler Call pa_context_disconnect() explicitly use SNDERR instead of fprintf to print error messages Support S32 sample types Add trailing NUL character to snprintf output Get rid of pulse_poll_revents() Add more error checking Remove fix for bug 0003470 Rework hardware parameter selection A bag of minor clean ups for ctl_pulse.c Make pulse_ext_callback const Drop our own implementation of the poll() callbacks A bag of minor clean-ups for pulse.c Split out O_NONBLOCK setting into seperate function Save a byte of memory Adjust buffering metrics to match what PA internally uses Make sure we always have a sensible channel mapping Use PA_STREAM_EARLY_REQUESTS if available Use S32/FLOAT32 only where available in the PA libs Add const to our snd_pcm_ioplug_callback_t instances Don't implement our own poll handlers, we can use the default ones Remove our own poll handler implementation entirely A bag of clean-ups for pcm_ctl.c Those patches only touch the pulse/ subdir. The tree is freshly rebased against current alsa-plugins master. May I ask you to make me the "semi-official" maintainer of this driver? I.e. I'd like to be consulted (as in 'Signed-Off-by') before any patches for it are merged? If requested I can also mail this as series of 26 seperate patches. Thank you, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] DaVinci ASoC DMA stalls
by Mark Lokowich 04 Sep '08

04 Sep '08

I've been using the DaVinci ASoC for a few months and have recently upgraded to the 2.6.26 based DaVinci git kernel. Now my audio DMA stalls more readily after stopping an active stream. I can manually trigger the event by poking the ESR to reactivate the stalled stream, suggesting the problem is in the ASP-to-DMA XEVT interface. This problem is less prevalent in the 2.6.25 based kernel. Any help? Regards, Mark Lokowich Advanced Communication Design

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[alsa-devel] how to convert stereo capture to mono stream (hardware incompatibility)
by brunal 04 Sep '08

04 Sep '08

Hi all, I've already asked my question in the alsa-user list, but as nobody was able to answer, I'm trying here if I have more chance. I'm using a USB M-Audio fasttrack pro sound card under linux 2.6.24.rt, and I want to plug it into a voice recognition software named Julius. There is an implementation of alsa in Julius, but I can't make it working with my USB sound card, because when alsa tries to set channel capture to 1, I get an error from alsa. Here is the code : /* set number of channels */ if ((err = snd_pcm_hw_params_set_channels(handle, hwparams, 1)) < 0) { jlog("Error: adin_alsa: cannot set PCM channel to %d (%s)\n", 1, snd_strerror(err)); return(FALSE); } Which always return FALSE. The thing is that when I change this line to set stereo channels, Julius recognition engine complains when it runs, and strip lots of audio packets, which seems to broke recognition. So my question is : how can I by-pass this problem? I've tried to do it with the .asoundrc file, but it seems that plugin or dmix stuff only works with playback, not recording. Does someone have an idea how to make a recording plugin that accept stereo stream from hardware and output only the left channel? I didn't understand how stereo stream are written in the audio buffer gave by alsa. Can someone explain me how to separate the two stream once it is written in the buffer? I really thanks all who would take little attention to my questions, it's been 5 days I'm digging on the web but I only get code working for setreo applications, which doesn't help me so much. And I know I might have an answer if I look into the audacity code, but it seems to me it's like using a bazooka for fishing redfish.... Best Regards, Bruno.

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[alsa-devel] [PATCH 1/1] ASoC codec: SSM2602 audio codec driver (v2)
by Bryan Wu 04 Sep '08

04 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v1-v2: - coding style fixing - use pr_xxx macros to replace printk(KERN_XXX...) - use new-style i2c interface - update to use latest ASoC API Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 773 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 131 ++++++++ 4 files changed, 909 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5d77dc3..8ef5ce6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -92,3 +92,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_SSM2602 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 35daaa9..0cd55ee 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-ssm2602-objs := ssm2602.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o @@ -33,3 +34,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c new file mode 100644 index 0000000..feb462e --- /dev/null +++ b/sound/soc/codecs/ssm2602.c @@ -0,0 +1,773 @@ +/* + * File: sound/soc/codecs/ssm2602.c + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * Description: Driver for ssm2602 sound chip built in ADSP-BF52xC + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ssm2602.h" + +#define AUDIO_NAME "ssm2602" +#define SSM2602_VERSION "0.1" + +struct snd_soc_codec_device soc_codec_dev_ssm2602; + +/* codec private data */ +struct ssm2602_priv { + unsigned int sysclk; + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* + * ssm2602 register cache + * We can't read the ssm2602 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { + 0x0017, 0x0017, 0x0079, 0x0079, + 0x0000, 0x0000, 0x0000, 0x000a, + 0x0000, 0x0000 +}; + +/* + * read ssm2602 register cache + */ +static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == SSM2602_RESET) + return 0; + if (reg >= SSM2602_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ssm2602 register cache + */ +static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= SSM2602_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the ssm2602 register space + */ +static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 ssm2602 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + ssm2602_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0) +/*Appending several "None"s just for OSS mixer use*/ +static const char *ssm2602_input_select[] = { + "Line", "Mic", "None", "None", "None", + "None", "None", "None", +}; + +static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum ssm2602_enum[] = { + SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select), + SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), +}; + +static const struct snd_kcontrol_new ssm2602_snd_controls[] = { + +SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, + 0, 127, 0), +SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V, + 7, 1, 0), + +SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), + +SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), +SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), + +SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), + +SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), + +SOC_ENUM("Capture Source", ssm2602_enum[0]), + +SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), +}; + +/* add non dapm controls */ +static int ssm2602_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Output Mixer */ +static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ssm2602_input_mux_controls = +SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); + +static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, + &ssm2602_output_mixer_controls[0], + ARRAY_SIZE(ssm2602_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), +SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), +SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("RLINEIN"), +SND_SOC_DAPM_INPUT("LLINEIN"), +}; + +static const struct snd_soc_dapm_route audio_conn[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Line", "Line Input"}, + {"Input Mux", "Mic", "Mic Bias"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, + {"Mic Bias", NULL, "MICIN"}, + + /* terminator */ + {NULL, NULL, NULL}, +}; + +static int ssm2602_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets, + ARRAY_SIZE(ssm2602_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return 0; +} + +static int ssm2602_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; + int i = get_coeff(ssm2602->sysclk, params_rate(params)); + u16 srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + ssm2602_write(codec, SSM2602_IFACE, iface); + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVATE_CODEC); + return 0; +} + +static int ssm2602_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + /* The DAI has shared clocks so if we already have a playback or + * capture going then constrain this substream to match it. + */ + if (ssm2602->master_substream) { + master_runtime = ssm2602->master_substream->runtime; + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + ssm2602->slave_substream = substream; + } else + ssm2602->master_substream = substream; + + return 0; +} + +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* set active */ + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVATE_CODEC); + + return 0; +} + +static void ssm2602_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* deactivate */ + if (!codec->active) { + udelay(50); + ssm2602_write(codec, SSM2602_ACTIVE, 0); + } +} + +static int ssm2602_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & 0xfff7; + if (mute) + ssm2602_write(codec, SSM2602_APDIGI, + mute_reg | ENABLE_DAC_MUTE); + else + ssm2602_write(codec, SSM2602_APDIGI, mute_reg); + return 0; +} + +static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2602->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + ssm2602_write(codec, SSM2602_IFACE, iface); + return 0; +} + +static int ssm2602_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + ssm2602_write(codec, SSM2602_PWR, 0); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + ssm2602_write(codec, SSM2602_PWR, reg | CLK_OUT_PDN); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_PWR, 0xffff); + break; + + } + codec->bias_level = level; + return 0; +} + +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +struct snd_soc_dai ssm2602_dai = { + .name = "SSM2602", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + }, + .dai_ops = { + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(ssm2602_dai); + +static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2602_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(ssm2602_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ssm2602_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the ssm2602 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ssm2602_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int reg, ret = 0; + + codec->name = "SSM2602"; + codec->owner = THIS_MODULE; + codec->read = ssm2602_read_reg_cache; + codec->write = ssm2602_write; + codec->set_bias_level = ssm2602_set_bias_level; + codec->dai = &ssm2602_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ssm2602_reg); + codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + ssm2602_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("ssm2602: failed to create pcms\n"); + goto pcm_err; + } + /*power on device*/ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + /* set the update bits */ + reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL); + ssm2602_write(codec, SSM2602_LINVOL, reg | LRIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL); + ssm2602_write(codec, SSM2602_RINVOL, reg | RLIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V); + ssm2602_write(codec, SSM2602_LOUT1V, reg | LRHP_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); + ssm2602_write(codec, SSM2602_ROUT1V, reg | RLHP_BOTH); + /*select Line in as default input*/ + ssm2602_write(codec, SSM2602_APANA, + ENABLE_MIC_BOOST2 | SELECT_DAC | ENABLE_MIC_BOOST); + ssm2602_write(codec, SSM2602_PWR, 0); + + ssm2602_add_controls(codec); + ssm2602_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("ssm2602: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *ssm2602_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = ssm2602_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = ssm2602_init(socdev); + if (ret < 0) + pr_err("failed to initialise SSM2602\n"); + + return ret; +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +/* corgi i2c codec control layer */ +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "SSM2602 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; + +static int ssm2602_add_i2c_device(struct platform_device *pdev, + const struct ssm2602_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ssm2602_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ssm2602", I2C_NAME_SIZE); + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + return 0; +err_driver: + i2c_del_driver(&ssm2602_i2c_driver); + return -ENODEV; +} + +#endif + +static int ssm2602_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ssm2602_setup_data *setup; + struct snd_soc_codec *codec; + struct ssm2602_priv *ssm2602; + int ret = 0; + + pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + if (ssm2602 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ssm2602; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ssm2602_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = ssm2602_add_i2c_device(pdev, setup); + } +#else + /* other interfaces */ +#endif + return ret; +} + +/* remove everything here */ +static int ssm2602_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&ssm2602_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ssm2602 = { + .probe = ssm2602_probe, + .remove = ssm2602_remove, + .suspend = ssm2602_suspend, + .resume = ssm2602_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); + +MODULE_DESCRIPTION("ASoC ssm2602 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h new file mode 100644 index 0000000..ec5e604 --- /dev/null +++ b/sound/soc/codecs/ssm2602.h @@ -0,0 +1,131 @@ +/* + * File: sound/soc/codecs/ssm2602.h + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef _SSM2602_H +#define _SSM2602_H + +/* SSM2602 Codec Register definitions */ + +#define SSM2602_LINVOL 0x00 +#define SSM2602_RINVOL 0x01 +#define SSM2602_LOUT1V 0x02 +#define SSM2602_ROUT1V 0x03 +#define SSM2602_APANA 0x04 +#define SSM2602_APDIGI 0x05 +#define SSM2602_PWR 0x06 +#define SSM2602_IFACE 0x07 +#define SSM2602_SRATE 0x08 +#define SSM2602_ACTIVE 0x09 +#define SSM2602_RESET 0x0f + +/*SSM2602 Codec Register Field definitions + *(Mask value to extract the corresponding Register field) + */ + +/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/ +#define LIN_VOL 0x01F /* Left Channel PGA Volume control */ +#define LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */ +#define LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */ + +/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/ +#define RIN_VOL 0x01F /* Right Channel PGA Volume control */ +#define RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */ +#define RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */ + +/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/ +#define LHP_VOL 0x07F /* Left Channel Headphone volume control */ +#define ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */ +#define LRHP_BOTH 0x100 /* Left Channel Headphone volume update */ + +/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/ +#define RHP_VOL 0x07F /* Right Channel Headphone volume control */ +#define ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */ +#define RLHP_BOTH 0x100 /* Right Channel Headphone volume update */ + +/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/ +#define ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */ +#define ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */ +#define ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */ +#define ENABLE_BYPASS 0x008 /* Line input bypass to line output */ +#define SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */ +#define ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */ +#define SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */ +#define ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */ + +/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/ +#define ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */ +#define DE_EMPHASIS 0x006 /* De-Emphasis Control */ +#define ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */ +#define STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */ + +/*Power Down Control (SSM2602_REG_POWER) + *(1=Enable PowerDown, 0=Disable PowerDown) + */ +#define LINE_IN_PDN 0x001 /* Line Input Power Down */ +#define MIC_PDN 0x002 /* Microphone Input & Bias Power Down */ +#define ADC_PDN 0x004 /* ADC Power Down */ +#define DAC_PDN 0x008 /* DAC Power Down */ +#define OUT_PDN 0x010 /* Outputs Power Down */ +#define OSC_PDN 0x020 /* Oscillator Power Down */ +#define CLK_OUT_PDN 0x040 /* CLKOUT Power Down */ +#define POWER_OFF 0x080 /* POWEROFF Mode */ + +/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/ +#define IFACE_FORMAT 0x003 /* Digital Audio input format control */ +#define AUDIO_DATA_LEN 0x00C /* Audio Data word length control */ +#define DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */ +#define DAC_LR_SWAP 0x020 /* Swap DAC data control */ +#define ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */ +#define BCLK_INVERT 0x080 /* Bit Clock Inversion control */ + +/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/ +#define ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */ +#define BOS_RATE 0x002 /* Base Over-Sampling rate */ +#define SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */ +#define CORECLK_DIV2 0x040 /* Core Clock divider select */ +#define CLKOUT_DIV2 0x080 /* Clock Out divider select */ + +/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/ +#define ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */ + +/*********************************************************************/ + +#define SSM2602_CACHEREGNUM 10 + +#define SSM2602_SYSCLK 0 +#define SSM2602_DAI 0 + +struct ssm2602_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ssm2602_dai; +extern struct snd_soc_codec_device soc_codec_dev_ssm2602; + +#endif -- 1.5.6

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[alsa-devel] [PATCH 1/1] ASoC codec: fix compiling error in ad1980 driver after ASoC API changed
by Bryan Wu 04 Sep '08

04 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad1980.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index bfbab3d..4e09c1f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -141,7 +141,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -struct snd_soc_codec_dai ad1980_dai = { +struct snd_soc_dai ad1980_dai = { .name = "AC97", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h index 5d4710d..db6c850 100644 --- a/sound/soc/codecs/ad1980.h +++ b/sound/soc/codecs/ad1980.h @@ -17,7 +17,7 @@ #define PR5 0x2000 #define PR6 0x4000 -extern struct snd_soc_codec_dai ad1980_dai; +extern struct snd_soc_dai ad1980_dai; extern struct snd_soc_codec_device soc_codec_dev_ad1980; #endif -- 1.5.6

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[alsa-devel] Does Gravis Classic driver work?
by Krzysztof Helt 04 Sep '08

04 Sep '08

Hi, I have acquired an unknown clone of the GF1 classic card. I load the snd-gusclassic driver with modprobe snd-gusclassic port=0x220 and the /proc/asound/cards says 0 [G37 ]: GUS - Gravis UltraSound Classic (3.7) Gravis UltraSound Classic (3.7) at 0x220, irq 5, dma 5&5 The alsamixer works. I set maximum volume for the PCM and master controls. The aplay -l says: **** List of PLAYBACK Hardware Devices **** card 0: G37 [Gravis UltraSound Classic (3.7)], device 0: GF1 [GF1 (synth)] Subdevices: 1/1 Subdevice #0: subdevice #0 but the aplay -L says nothing. the aplay with the .wav file does not play. aplay /usr/share/sounds/alsa/Front_Center.wav ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:545: audio open error: No such file or directory I have tried to follow the alsa wiki but it has not helped (btw, there is no driver the "gusclassic" listed, only the "gus"). Could you point me to any piece of information how to set up the card? The card works with DOS game with Miles (?) drivers. Regards, Krzysztof ---------------------------------------------------------------------- Teraz taniej na zagraniczne komorki! Sprawdz >> http://link.interia.pl/f1ef7

2 1

[alsa-devel] noise and consume pcm faster
by Nie Jun 03 Sep '08

03 Sep '08

Hi, I found a strange issue that ALSA consume pcm data faster than normal. Does any one had come across it? When I sychronize video frame against audio, the video thread will check how much pcm data has been written to ALSA API but has not been fetched by DMA. This query causes ALSA consume pcm data faster than normal, noise appears too. A stream that is 131 seconds will playback just 112 seconds. It only appears when audio is stereo and sampling rate is directly supported by hardware. This bug can be reproduced with aplay. To reproduce it with aplay, you just need to add a thread and query the ALSA latency 20~30times every second. My system: XSCALE PXA300/zylonite, Linux2.6.25, alsa-lib-1.0.13(I tried alsa-lib-1.0.17, bugs too), codec: wm9713. Linux2.6.14 has this issue too, but I did not find it in 2.6.21 and 2.6.24. Does anyone has idea about it? Thanks! Jun Nie

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Re: [alsa-devel] [PATCH 1/4] ASOC: Blackfin driver for ALSA SoC framework
by Mark Brown 03 Sep '08

03 Sep '08

On Wed, Aug 27, 2008 at 05:39:25PM +0800, Bryan Wu wrote: > From: Cliff Cai <cliff.cai(a)analog.com> > > Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> > Signed-off-by: Bryan Wu <cooloney(a)kernel.org> As with the other patches in the series this looks good but it has been affected by changes in the ASoC APIs and needs to be updated to reflect current ALSA. The topic/asoc branch of Takashi's tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git is what's currently queued for 2.6.28. Other than that, checkpatch-style things and a few more minor issues the main things are the register write interface in /proc and the hardware parameters for the I2S driver. I've not flagged all the checkpatch stuff here. Can I also suggest splitting this up into a series of patches for easier review - for example, adding the SPORT support first, followed by the AC97 and I2S drivers and then the machine drivers? I'd certainly have appreciated having seen the SPORT support (which appears at the end of the patch) before looking at the AC97 and I2S stuff. > + depends on BLACKFIN && SND_SOC > + help > + Say Y or M if you want to add support for codecs attached to > + the Blackfin SPORT (synchronous serial ports) interface in slot 16 > + mode (pseudo AC97 interface). > + You will also need to select the audio interfaces to support below. > + > + Note: > + AC'97 codecs which do not implment the slot-16 mode will not function > + properly with this driver. This driver is known to work with the > + Analog Devices line of AC97 codecs. Your spelling of AC97 isn't consistent here. The kernel seems to prefer AC97, as do you. > + * File: sound/soc/blackfin/bf5xx-ac97-pcm.c > + * Author: Cliff Cai <Cliff.Cai(a)analog.com> > + * > + * Created: Tue June 06 2008 > + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC Hopefully the driver is more generic than that? > +static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + size_t size = bf5xx_pcm_hardware.buffer_bytes_max * sizeof(struct ac97_frame)/4; Checkpatch picks up on this and quite a few other places being over 80 columns. > +static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct sport_device *sport = runtime->private_data; > + int ret = 0; > + > + pr_debug("%s %s\n", substream->stream?"Capture":"Playback", \ > + cmd?" start":" stop"); This looks wrong - there's rather more options for cmd, for example, and you're making assumptions about the value of SNDRV_PMC_STREAM_PLAYBACK. > +/*Need to allocate local buffer when enable MMAP for SPORT working in TMD mode(include AC97).*/ Indentation? > diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h > new file mode 100644 > index 0000000..5831300 > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.h > +struct bf5xx_gpio { > + u32 sys; > + u32 rx; > + u32 tx; > + u32 clk; > + u32 frm; > +}; That should be a space before rx, not a tab. > diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c > new file mode 100644 > index 0000000..685a494 > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-ac97.c > +#include <sound/driver.h> This header is obsolte and should be removed. > +static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97, > + unsigned short reg) > +{ > + struct ac97_frame out_frame[2], in_frame[2]; > + > + pr_debug("%s enter 0x%x\n", __func__, reg); > + > + /* When dma descriptor is enabled, the register should not be read */ > + if (sport_handle->tx_run || sport_handle->rx_run) { > + printk(KERN_ERR "Could you send a mail to author " > + "to report this?\n"); > + return -EFAULT; > + } Might be nice to say what to report and who the author is - by itself the printout is going to be rather obscure :) > +static int bf5xx_ac97_resume(struct platform_device *pdev, > + struct snd_soc_cpu_dai *dai) struct snd_soc_cpu_dai and struct snd_soc_codec_dai have been merged into a single struct snd_soc_dai in current versions. > + ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } > + ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } Shouldn't this undo the previous sport_confix_rx() on error? Also, the use of blank lines is a bit funny here - I'd expect the blank before rather than after the function calls. > +static struct proc_dir_entry *ac_entry; > + > +/* For test purpose, read a register from codec */ > +static int proc_write(struct file *file, const char __user *buffer, > + unsigned long count, void *data) > +{ I think it's probably better to drop this from mainline - it doesn't seem like something that we should be supporting long term and obviously it could do damage if used. If you do want to put it in mainline then it ought to go in debugfs rather than proc. > + /*SPORT works in TDM mode to simulate AC97 transfers*/ > + ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } > + ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); > + > + if (ret) { > + printk(KERN_ERR "SPORT is busy!\n"); > + return -EBUSY; > + } Again, odd blank lines and presumably there needs to be something to undo things on error? > + * File: sound/soc/blackfin/bf5xx-ad1980.c > + * Author: Cliff Cai <Cliff.Cai(a)analog.com> > + * > + * Created: Tue June 06 2008 o> + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC This comment is wrong - it's for the AD1980. > +static int bf5xx_board_startup(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; > + > + pr_debug("%s enter\n", __func__); > + cpu_dai->private_data = sport_handle; > + return 0; > +} > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c > @@ -0,0 +1,293 @@ > +/* > + * File: sound/soc/blackfin/bf5xx-i2s-pcm.c > + * Author: Cliff Cai <Cliff.Cai(a)analog.com> > + * > + * Created: Tue June 06 2008 > + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC This comment is wrong too :) > +#include <sound/driver.h> Obsolete. > + pr_debug("%s %s\n", substream->stream?"Capture":"Playback", \ > + cmd?" start":" stop"); Same comment as for AC97. Shouldn't more of this code be shared with the AC97 DMA driver? > +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) > +{ > + pr_debug("%s enter\n", __func__); > + /*Nothing need to be cleared here*/ > + > + return 0; > +} You should just be able to remove this if it's empty. > +#include <sound/driver.h> Obsolete. > + /* TX and RX are not independent,they are enabled at the same time, > + * even if only one side is running.So,we need to configure both of them in advance. > + * CPU DAI format:I2S,word length:32 bit,slave mode. > + */ Are the RX and TX clocks independant? If not you should probably tell user space about the constraint in startup(), forcing the second stream that's opened to use the same configuration as the first - take a look at the fsl_ssi driver or the wm8903 driver for examples of how to do this. I'm also not seeing the code that configures the sample rate anywhere - but then it looks like the driver only support slave mode ATM? That's what the machine driver is using. There should still be a set_fmt() to document what's supported if nothing else. > +void decfrag(struct sport_device *sport, int *frag, int tx) > +{ > + --(*frag); > + if (tx == 1 && *frag == 0) > + *frag = sport->tx_frags; > + > + if (tx == 0 && *frag == 0) > + *frag = sport->rx_frags; > +} > +EXPORT_SYMBOL(decfrag); This symbol should be namespaced. > diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c > new file mode 100644 > index 0000000..41e161f > --- /dev/null > +++ b/sound/soc/blackfin/bf5xx-ssm2602.c This driver needs to follow the ssm2602 codec driver so either the series needs reordering or at least this bit needs to be split out into another patch. > +#include <sound/driver.h> Obsolete. > + * WARNING - TODO > + * > + * This code assumes there is a variable clocksource for the SSM2602. > + * i.e. it supplies MCLK depending on rate. > + * > + * If you are using a crystal source then modify the below case > + * statement with a static frequency. > + * > + * If you are using the SPORT to generate clocking then this is > + * where to do it. > + */ > + > + switch (params_rate(params)) { > + case 8000: > + case 16000: > + case 48000: > + case 96000: > + case 11025: > + case 22050: > + case 44100: > + clk = 12000000; > + break; > + } I'm not sure that comment quite agrees with the code here...

2 2

[alsa-devel] [PATCH 1/1] ALSA: add dummy function to support shared mmap in nommu Blackfin arch (v3)
by Bryan Wu 03 Sep '08

03 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v2-v3: - when it is mmu arch, we define the dummy function to NULL v1-v2: - only adding this dummy function in nommu arch Cc: Mike Frysinger <vapier.adi(a)gmail.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/core/pcm_native.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c49b9d9..e2068a4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3391,6 +3391,16 @@ out: } #endif /* CONFIG_SND_SUPPORT_OLD_API */ +#ifndef CONFIG_MMU +unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr, + unsigned long len, unsigned long pgoff, unsigned long flags) +{ + return 0; +} +#else +# define dummy_get_unmapped_area NULL +#endif + /* * Register section */ @@ -3407,6 +3417,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, + .get_unmapped_area = dummy_get_unmapped_area, }, { .owner = THIS_MODULE, @@ -3419,5 +3430,6 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, + .get_unmapped_area = dummy_get_unmapped_area, } }; -- 1.5.6

2 3

[alsa-devel] [PATCH 1/1] ALSA: add dummy function to support shared mmap in nommu Blackfin arch (v2)
by Bryan Wu 03 Sep '08

03 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v1-v2: - only adding this dummy function in nommu arch Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/core/pcm_native.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c49b9d9..1e88cde 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3391,6 +3391,14 @@ out: } #endif /* CONFIG_SND_SUPPORT_OLD_API */ +#ifndef CONFIG_MMU +unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr, + unsigned long len, unsigned long pgoff, unsigned long flags) +{ + return 0; +} +#endif + /* * Register section */ @@ -3407,6 +3415,9 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, +#ifndef CONFIG_MMU + .get_unmapped_area = dummy_get_unmapped_area, +#endif }, { .owner = THIS_MODULE, @@ -3419,5 +3430,8 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, +#ifndef CONFIG_MMU + .get_unmapped_area = dummy_get_unmapped_area, +#endif } }; -- 1.5.6

2 2

[alsa-devel] [PATCH] ASoC: Convert wm8510 to a new-style i2c driver (v2)
by Jean Delvare 02 Sep '08

02 Sep '08

Convert the wm8510 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Geoffrey Wossum <gwossum(a)acm.org> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- As noticed by Mark, my 1st patch didn't apply properly. Here's a fixed version which should apply fine. Sorry for the noise. Warning: this assumes that the playpaq has its codec chip on I2C bus 0, which may or may not be the case. Someone with this machine please test and report. sound/soc/at32/playpaq_wm8510.c | 1 sound/soc/codecs/wm8510.c | 110 ++++++++++++++++++--------------------- sound/soc/codecs/wm8510.h | 1 3 files changed, 55 insertions(+), 57 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/at32/playpaq_wm8510.c 2008-09-02 16:42:48.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/at32/playpaq_wm8510.c 2008-09-02 16:43:28.000000000 +0200 @@ -377,6 +377,7 @@ static struct snd_soc_machine snd_soc_ma static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, .i2c_address = 0x1a, }; --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8510.c 2008-09-02 16:42:48.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8510.c 2008-09-02 16:43:28.000000000 +0200 @@ -665,88 +665,86 @@ static struct snd_soc_device *wm8510_soc /* * WM8510 2 wire address is 0x1a */ -#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8510_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ - -static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8510_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8510_socdev; - struct wm8510_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8510_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8510\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8510_i2c_detach(struct i2c_client *client) +static int wm8510_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8510_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8510_codec_probe); -} +static const struct i2c_device_id wm8510_i2c_id[] = { + { "wm8510", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "WM8510 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8510, - .attach_adapter = wm8510_i2c_attach, - .detach_client = wm8510_i2c_detach, - .command = NULL, + .probe = wm8510_i2c_probe, + .remove = wm8510_i2c_remove, + .id_table = wm8510_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8510", - .driver = &wm8510_i2c_driver, -}; +static int wm8510_add_i2c_device(struct platform_device *pdev, + const struct wm8510_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8510", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8510_i2c_driver); + return -ENODEV; +} #endif static int wm8510_probe(struct platform_device *pdev) @@ -771,11 +769,8 @@ static int wm8510_probe(struct platform_ wm8510_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8510_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8510_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -798,6 +793,7 @@ static int wm8510_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); #endif kfree(codec); --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8510.h 2008-09-02 16:42:48.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8510.h 2008-09-02 16:43:28.000000000 +0200 @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int i2c_bus; unsigned short i2c_address; }; -- Jean Delvare

2 2

[alsa-devel] [PATCH 0/4] ARM: OMAP2: Add support for Gumstix Overo
by sakoman＠gmail.com 02 Sep '08

02 Sep '08

This patch series adds support for the Gumstix Overo board. Since it includes an ASoC driver I am copying both linux-omap and alsa-devel The first patch adds the basic board files, the second adds an ASoC driver for the TWL4030 codec, the third an ASoC machine driver for the Overo, and the final patch is the Overo defconfig

9 20

[alsa-devel] |PATCH] ASoC: Convert wm8510 to a new-style i2c driver
by Jean Delvare 02 Sep '08

02 Sep '08

Convert the wm8510 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Geoffrey Wossum <gwossum(a)acm.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Warning: this assumes that the playpaq has its codec chip on I2C bus 0, which may or may not be the case. Someone with this machine please test and report. sound/soc/at32/playpaq_wm8510.c | 1 sound/soc/codecs/wm8510.c | 108 +++++++++++++++++++-------------------- sound/soc/codecs/wm8510.h | 1 3 files changed, 55 insertions(+), 55 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/at32/playpaq_wm8510.c 2008-09-01 09:59:13.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/at32/playpaq_wm8510.c 2008-09-01 09:59:21.000000000 +0200 @@ -377,6 +377,7 @@ static struct snd_soc_machine snd_soc_ma static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, .i2c_address = 0x1a, }; --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8510.c 2008-09-01 09:59:13.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8510.c 2008-09-01 09:59:21.000000000 +0200 @@ -665,85 +665,85 @@ static struct snd_soc_device *wm8510_soc /* * WM8510 2 wire address is 0x1a */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8510_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ - -static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8510_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8510_socdev; - struct wm8510_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8510_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8510\n"); - goto err; - } - return ret; - -err: - kfree(i2c); return ret; } -static int wm8510_i2c_detach(struct i2c_client *client) +static int wm8510_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8510_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8510_codec_probe); -} +static const struct i2c_device_id wm8510_i2c_id[] = { + { "wm8510", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "WM8510 I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8510_i2c_attach, - .detach_client = wm8510_i2c_detach, - .command = NULL, + .probe = wm8510_i2c_probe, + .remove = wm8510_i2c_remove, + .id_table = wm8510_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8510", - .driver = &wm8510_i2c_driver, -}; +static int wm8510_add_i2c_device(struct platform_device *pdev, + const struct wm8510_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8510", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8510_i2c_driver); + return -ENODEV; +} #endif static int wm8510_probe(struct platform_device *pdev) @@ -768,11 +768,8 @@ static int wm8510_probe(struct platform_ wm8510_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8510_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8510_add_i2c_device(pdev, setup);; } #else /* Add other interfaces here */ @@ -795,6 +792,7 @@ static int wm8510_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); #endif kfree(codec); --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8510.h 2008-09-01 09:59:13.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8510.h 2008-09-01 09:59:21.000000000 +0200 @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int i2c_bus; unsigned short i2c_address; }; -- Jean Delvare

3 4

[alsa-devel] [PATCH 0/4] Blackfin supports for ALSA/ASOC
by Bryan Wu 02 Sep '08

02 Sep '08

Hi folks, It's very happy to release this ASOC Blackfin ports. Please review our drivers and feel free to ask us update it. In this patchset, we post ASOC Blackfin supports, SSM2602 audio codec driver and other 2 ALSA/ASOC patches. Thanks a lot -Bryan

8 21

[alsa-devel] get rid of controls with snd_ctl_remove
by Harald Radke 02 Sep '08

02 Sep '08

Hi there! Please be patient with me, I am quite inexperienced with alsa programming, so this might be a dumb question: We have a WM8750 working in our PDA, the codec is working so far but I want to remove some of the control since they aren't needed. Instead of copying the wm8750.c file and adjusting it, I want to write another module which basically get rid of those controls, something like: [...] static const char* unused_controls[] = {"AAAA", "BBBB,"CCCC"}; struct snd_kcontrol *ctl; list_for_each_entry(ctl, &card->controls, list) for(i=0; i < ARRAY_SIZE(unused_controls); i++) if (!strcmp(unused_controls[i],ctl->id.name)) { down_write(&card->controls_rwsem); snd_ctl_remove(card,ctl); up_write(&card->controls_rwsem); break; } unfortunatly this doesn't work (module segfaults...."of course" you will say *g*) My question: is it possible at all and I am just not deep nuff in understanding the sound system or do I have to adjust the original code? Thx Harry

4 14

[alsa-devel] [PATCH 2/3] ASoC: Convert wm8753 to a new-style i2c driver
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the wm8753 codec driver to the new (standard) i2c device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Ville Syrjala <syrjala(a)sci.fi> Cc: Frank Mandarino <fmandarino(a)endrelia.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Warning: this assumes that the neo1973 has its codec chip on I2C bus 0, which may or may not be the case. Someone with this machine please test and report. include/linux/i2c-id.h | 1 sound/soc/codecs/wm8753.c | 106 ++++++++++++++++++------------------ sound/soc/codecs/wm8753.h | 1 sound/soc/s3c24xx/neo1973_wm8753.c | 1 4 files changed, 55 insertions(+), 54 deletions(-) --- linux-2.6.27-rc5.orig/include/linux/i2c-id.h 2008-09-01 11:06:04.000000000 +0200 +++ linux-2.6.27-rc5/include/linux/i2c-id.h 2008-09-01 12:46:57.000000000 +0200 @@ -64,7 +64,6 @@ #define I2C_DRIVERID_SAA717X 80 /* saa717x video encoder */ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ -#define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ #define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8753.c 2008-08-29 09:32:25.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8753.c 2008-09-01 12:49:46.000000000 +0200 @@ -1637,84 +1637,86 @@ static struct snd_soc_device *wm8753_soc * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8753_i2c_driver; -static struct i2c_client client_template; - -static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8753_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8753_socdev; - struct wm8753_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (!i2c) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8753_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8753\n"); - goto err; - } - - return ret; -err: - kfree(i2c); return ret; } -static int wm8753_i2c_detach(struct i2c_client *client) +static int wm8753_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8753_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8753_codec_probe); -} +static const struct i2c_device_id wm8753_i2c_id[] = { + { "wm8753", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8753_i2c_driver = { .driver = { .name = "WM8753 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8753, - .attach_adapter = wm8753_i2c_attach, - .detach_client = wm8753_i2c_detach, - .command = NULL, + .probe = wm8753_i2c_probe, + .remove = wm8753_i2c_remove, + .id_table = wm8753_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8753", - .driver = &wm8753_i2c_driver, -}; +static int wm8753_add_i2c_device(struct platform_device *pdev, + const struct wm8753_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8753_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8753", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8753_i2c_driver); + return -ENODEV; +} #endif static int wm8753_probe(struct platform_device *pdev) @@ -1748,11 +1750,8 @@ static int wm8753_probe(struct platform_ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8753_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8753_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1796,6 +1795,7 @@ static int wm8753_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8753_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8753.h 2008-08-05 18:18:18.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8753.h 2008-09-01 12:44:52.000000000 +0200 @@ -79,6 +79,7 @@ #define WM8753_ADCTL2 0x3f struct wm8753_setup_data { + int i2c_bus; unsigned short i2c_address; }; --- linux-2.6.27-rc5.orig/sound/soc/s3c24xx/neo1973_wm8753.c 2008-08-23 22:18:36.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/s3c24xx/neo1973_wm8753.c 2008-09-01 12:45:10.000000000 +0200 @@ -586,6 +586,7 @@ static struct snd_soc_machine neo1973 = }; static struct wm8753_setup_data neo1973_wm8753_setup = { + .i2c_bus = 0, .i2c_address = 0x1a, }; -- Jean Delvare

2 1

[alsa-devel] [PATCH 3/3] ASoC: Convert neo1973/lm4857 to a new-style i2c driver
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the lm4857 driver in neo1973_wm8753 to the new (standard) i2c device driver binding model. I assumed that the LM4857 was always on the same I2C bus as the WM8753 codec. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Tim Niemeyer <reddog(a)mastersword.de> Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I couldn't even test-build that one. Tim, Graeme, can any of you please review and test this patch? Thanks. include/linux/i2c-id.h | 1 sound/soc/s3c24xx/neo1973_wm8753.c | 110 ++++++++++++++++++------------------ 2 files changed, 58 insertions(+), 53 deletions(-) --- linux-2.6.27-rc5.orig/include/linux/i2c-id.h 2008-09-01 14:31:28.000000000 +0200 +++ linux-2.6.27-rc5/include/linux/i2c-id.h 2008-09-01 14:46:19.000000000 +0200 @@ -64,7 +64,6 @@ #define I2C_DRIVERID_SAA717X 80 /* saa717x video encoder */ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ -#define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ #define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_M52790 95 /* Mitsubishi M52790SP/FP AV switch */ --- linux-2.6.27-rc5.orig/sound/soc/s3c24xx/neo1973_wm8753.c 2008-09-01 14:31:30.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/s3c24xx/neo1973_wm8753.c 2008-09-01 15:02:39.000000000 +0200 @@ -597,54 +597,20 @@ static struct snd_soc_device neo1973_snd .codec_data = &neo1973_wm8753_setup, }; -static struct i2c_client client_template; - -static const unsigned short normal_i2c[] = { 0x7C, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind) +static int lm4857_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) { - int ret; - DBG("Entered %s\n", __func__); - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - - ret = i2c_attach_client(i2c); - if (ret < 0) { - printk(KERN_ERR "LM4857 failed to attach at addr %x\n", addr); - goto exit_err; - } - lm4857_write_regs(); - return ret; - -exit_err: - kfree(i2c); - return ret; -} - -static int lm4857_i2c_detach(struct i2c_client *client) -{ - DBG("Entered %s\n", __func__); - - i2c_detach_client(client); - kfree(client); return 0; } -static int lm4857_i2c_attach(struct i2c_adapter *adap) +static int lm4857_i2c_remove(struct i2c_client *client) { DBG("Entered %s\n", __func__); - return i2c_probe(adap, &addr_data, lm4857_amp_probe); + return 0; } static u8 lm4857_state; @@ -682,27 +648,67 @@ static void lm4857_shutdown(struct i2c_c lm4857_write_regs(); } -/* corgi i2c codec control layer */ +static const struct i2c_device_id lm4857_i2c_id[] = { + { "neo1973_lm4857", 0 } + { } +}; + static struct i2c_driver lm4857_i2c_driver = { .driver = { .name = "LM4857 I2C Amp", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_LM4857, .suspend = lm4857_suspend, .resume = lm4857_resume, .shutdown = lm4857_shutdown, - .attach_adapter = lm4857_i2c_attach, - .detach_client = lm4857_i2c_detach, - .command = NULL, -}; - -static struct i2c_client client_template = { - .name = "LM4857", - .driver = &lm4857_i2c_driver, + .probe = lm4857_i2c_probe, + .remove = lm4857_i2c_remove, + .id_table = lm4857_i2c_id, }; static struct platform_device *neo1973_snd_device; +static struct i2c_client *lm4857_client; + +static int __init neo1973_add_lm4857_device(struct platform_device *pdev, + int i2c_bus, + unsigned short i2c_address) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&lm4857_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add lm4857 driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = i2c_address; + strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + lm4857_client = client; + return 0; + +err_driver: + i2c_del_driver(&lm4857_i2c_driver); + return -ENODEV; +} static int __init neo1973_init(void) { @@ -723,11 +729,10 @@ static int __init neo1973_init(void) return ret; } - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "can't add i2c driver"); + ret = neo1973_add_lm4857_device(neo1973_snd_device, + neo1973_wm8753_setup, 0x7C); + if (ret != 0) platform_device_unregister(neo1973_snd_device); - } return ret; } @@ -736,6 +741,7 @@ static void __exit neo1973_exit(void) { DBG("Entered %s\n", __func__); + i2c_unregister_device(lm4857_client); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } -- Jean Delvare

1 0

[alsa-devel] [PATCH 1/3] ASoC: Fix an error path in neo1973_wm8753
by Jean Delvare 01 Sep '08

01 Sep '08

The error handling in neo1973_init is incorrect: * If platform_device_add fails, we go on with the rest of the initialization instead of bailing out. Things will break when the module is removed (platform_device_unregister called on a device that wasn't registered.) * If i2c_add_driver fails, we return an error so the module will not load, but we don't unregister neo1973_snd_device, so we are leaking resources. Add the missing error handling. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Tim Niemeyer <reddog(a)mastersword.de> Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I couldn't even test-build that one. Tim, Graeme, can any of you please review and test this patch? Thanks. sound/soc/s3c24xx/neo1973_wm8753.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/s3c24xx/neo1973_wm8753.c 2008-09-01 15:13:17.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/s3c24xx/neo1973_wm8753.c 2008-09-01 15:22:42.000000000 +0200 @@ -717,12 +717,16 @@ static int __init neo1973_init(void) neo1973_snd_devdata.dev = &neo1973_snd_device->dev; ret = platform_device_add(neo1973_snd_device); - if (ret) + if (ret) { platform_device_put(neo1973_snd_device); + return ret; + } ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) + if (ret != 0) { printk(KERN_ERR "can't add i2c driver"); + platform_device_unregister(neo1973_snd_device); + } return ret; } -- Jean Delvare

1 0

[alsa-devel] [PATCH] ASoC: Convert wm8731 to a new-style i2c driver (testers wanted)
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the wm8731 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Warning: this assumes that the corgi, poodle and eti_b1 have their codec chips on I2C bus 0, which may or may not be the case. Someone with these machines please test and report. Note that you can safely ignore patch failures in i2c-id.h. include/linux/i2c-id.h | 1 sound/soc/at91/eti_b1_wm8731.c | 1 sound/soc/codecs/wm8731.c | 111 +++++++++++++++++++--------------------- sound/soc/codecs/wm8731.h | 1 sound/soc/pxa/corgi.c | 1 sound/soc/pxa/poodle.c | 1 6 files changed, 59 insertions(+), 57 deletions(-) --- linux-2.6.27-rc4.orig/include/linux/i2c-id.h 2008-08-26 16:41:12.000000000 +0200 +++ linux-2.6.27-rc4/include/linux/i2c-id.h 2008-08-26 17:02:29.000000000 +0200 @@ -64,7 +64,6 @@ #define I2C_DRIVERID_SAA717X 80 /* saa717x video encoder */ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ -#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ --- linux-2.6.27-rc4.orig/sound/soc/at91/eti_b1_wm8731.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/at91/eti_b1_wm8731.c 2008-08-26 17:02:20.000000000 +0200 @@ -243,6 +243,7 @@ static struct snd_soc_machine snd_soc_ma }; static struct wm8731_setup_data eti_b1_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1a, }; --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8731.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8731.c 2008-08-26 17:11:50.000000000 +0200 @@ -570,86 +570,87 @@ static struct snd_soc_device *wm8731_soc * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8731_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ - -static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8731_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8731_socdev; - struct wm8731_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } ret = wm8731_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8731\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8731_i2c_detach(struct i2c_client *client) +static int wm8731_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8731_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8731_codec_probe); -} +static const struct i2c_device_id wm8731_i2c_id[] = { + { "wm8731", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "WM8731 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8731, - .attach_adapter = wm8731_i2c_attach, - .detach_client = wm8731_i2c_detach, - .command = NULL, + .probe = wm8731_i2c_probe, + .remove = wm8731_i2c_remove, + .id_table = wm8731_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8731", - .driver = &wm8731_i2c_driver, -}; +static int wm8731_add_i2c_device(struct platform_device *pdev, + const struct wm8731_setup_data *setup) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8731_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + socdev->codec->control_data = client; + + return 0; + +err_driver: + i2c_del_driver(&wm8731_i2c_driver); + return -ENODEV; +} #endif static int wm8731_probe(struct platform_device *pdev) @@ -682,11 +683,8 @@ static int wm8731_probe(struct platform_ wm8731_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8731_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8731_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -711,6 +709,7 @@ static int wm8731_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8731_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8731.h 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8731.h 2008-08-26 17:02:20.000000000 +0200 @@ -35,6 +35,7 @@ #define WM8731_DAI 0 struct wm8731_setup_data { + int i2c_bus; unsigned short i2c_address; }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/corgi.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/corgi.c 2008-08-26 17:02:20.000000000 +0200 @@ -330,6 +330,7 @@ static struct snd_soc_machine snd_soc_ma /* corgi audio private data */ static struct wm8731_setup_data corgi_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/poodle.c 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/poodle.c 2008-08-26 17:02:20.000000000 +0200 @@ -284,6 +284,7 @@ static struct snd_soc_machine snd_soc_ma /* poodle audio private data */ static struct wm8731_setup_data poodle_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; -- Jean Delvare

5 9

[alsa-devel] [PATCH] hda: Distortion fix for dell_m6_core_init
by Matthew Ranostay 01 Sep '08

01 Sep '08

Added the EQ distortion fix to the dell_m6_core_init. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 6ad6ef3..9968ee4 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -606,8 +606,9 @@ static struct hda_verb dell_eq_core_init[] = { }; static struct hda_verb dell_m6_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, /* setup audio connections */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},

2 2

[alsa-devel] [PATCH] ASoC: Convert wm8990 to a new-style i2c driver
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the wm8990 codec driver to the new (standard) device driver binding model. After this change, WM8990 devices are no longer discovered automatically and must instead be instantiated explicitly. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 103 +++++++++++++++++++++++---------------------- sound/soc/codecs/wm8990.h | 1 2 files changed, 54 insertions(+), 50 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8990.c 2008-08-29 09:32:25.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8990.c 2008-09-01 12:21:20.000000000 +0200 @@ -1477,81 +1477,86 @@ static struct snd_soc_device *wm8990_soc * low = 0x34 * high = 0x36 */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8990_i2c_driver; -static struct i2c_client client_template; - -static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8990_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8990_socdev; - struct wm8990_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8990_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8990\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8990_i2c_detach(struct i2c_client *client) +static int wm8990_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8990_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8990_codec_probe); -} +static const struct i2c_device_id wm8990_i2c_id[] = { + { "wm8990", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); static struct i2c_driver wm8990_i2c_driver = { .driver = { .name = "WM8990 I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8990_i2c_attach, - .detach_client = wm8990_i2c_detach, - .command = NULL, + .probe = wm8990_i2c_probe, + .remove = wm8990_i2c_remove, + .id_table = wm8990_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8990", - .driver = &wm8990_i2c_driver, -}; +static int wm8990_add_i2c_device(struct platform_device *pdev, + const struct wm8990_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8990_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8990", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8990_i2c_driver); + return -ENODEV; +} #endif static int wm8990_probe(struct platform_device *pdev) @@ -1584,11 +1589,8 @@ static int wm8990_probe(struct platform_ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8990_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8990_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1612,6 +1614,7 @@ static int wm8990_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8990_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc5.orig/sound/soc/codecs/wm8990.h 2008-08-21 09:10:47.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/wm8990.h 2008-09-01 12:05:05.000000000 +0200 @@ -827,6 +827,7 @@ #define WM8990_AINRMUX_PWR_BIT 3 struct wm8990_setup_data { + unsigned i2c_bus; unsigned short i2c_address; }; -- Jean Delvare

2 1

[alsa-devel] [PATCH] ASoC: Convert tlv320aic3x to a new-style i2c driver
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the tlv320aic3x codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Vladimir Barinov <vbarinov(a)ru.mvista.com> Cc: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Warning: this assumes that the davinci-evm and n810 have their codec chips on I2C bus 0, which may or may not be the case. Someone with these machines please test and report. sound/soc/codecs/tlv320aic3x.c | 106 +++++++++++++++++++-------------------- sound/soc/codecs/tlv320aic3x.h | 1 sound/soc/davinci/davinci-evm.c | 1 sound/soc/omap/n810.c | 1 4 files changed, 57 insertions(+), 52 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/codecs/tlv320aic3x.c 2008-08-29 09:32:24.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/tlv320aic3x.c 2008-09-01 09:55:44.000000000 +0200 @@ -1172,71 +1172,39 @@ static struct snd_soc_device *aic3x_socd * AIC3X 2 wire address can be up to 4 devices with device addresses * 0x18, 0x19, 0x1A, 0x1B */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver aic3x_i2c_driver; -static struct i2c_client client_template; /* * If the i2c layer weren't so broken, we could pass this kind of data * around */ -static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int aic3x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = aic3x_socdev; - struct aic3x_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - printk(KERN_ERR "aic3x: failed to attach codec at addr %x\n", - addr); - goto err; - } - ret = aic3x_init(socdev); - if (ret < 0) { + if (ret < 0) printk(KERN_ERR "aic3x: failed to initialise AIC3X\n"); - goto err; - } - return ret; - -err: - kfree(i2c); return ret; } -static int aic3x_i2c_detach(struct i2c_client *client) +static int aic3x_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int aic3x_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, aic3x_codec_probe); -} +static const struct i2c_device_id aic3x_i2c_id[] = { + { "tlv320aic3x", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); /* machine i2c codec control layer */ static struct i2c_driver aic3x_i2c_driver = { @@ -1244,13 +1212,9 @@ static struct i2c_driver aic3x_i2c_drive .name = "aic3x I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = aic3x_i2c_attach, - .detach_client = aic3x_i2c_detach, -}; - -static struct i2c_client client_template = { - .name = "AIC3X", - .driver = &aic3x_i2c_driver, + .probe = aic3x_i2c_probe, + .remove = aic3x_i2c_remove, + .id_table = aic3x_i2c_id, }; static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) @@ -1258,6 +1222,46 @@ static int aic3x_i2c_read(struct i2c_cli value[0] = i2c_smbus_read_byte_data(client, value[0]); return (len == 1); } + +static int aic3x_add_i2c_device(struct platform_device *pdev, + const struct aic3x_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&aic3x_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "tlv320aic3x", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&aic3x_i2c_driver); + return -ENODEV; +} #endif static int aic3x_probe(struct platform_device *pdev) @@ -1290,12 +1294,9 @@ static int aic3x_probe(struct platform_d aic3x_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t) i2c_master_send; codec->hw_read = (hw_read_t) aic3x_i2c_read; - ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = aic3x_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1320,6 +1321,7 @@ static int aic3x_remove(struct platform_ snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&aic3x_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc5.orig/sound/soc/codecs/tlv320aic3x.h 2008-08-05 18:18:18.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/tlv320aic3x.h 2008-09-01 08:29:55.000000000 +0200 @@ -224,6 +224,7 @@ int aic3x_get_gpio(struct snd_soc_codec int aic3x_headset_detected(struct snd_soc_codec *codec); struct aic3x_setup_data { + int i2c_bus; unsigned short i2c_address; unsigned int gpio_func[2]; }; --- linux-2.6.27-rc5.orig/sound/soc/davinci/davinci-evm.c 2008-08-13 09:51:02.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/davinci/davinci-evm.c 2008-09-01 08:30:20.000000000 +0200 @@ -136,6 +136,7 @@ static struct snd_soc_machine snd_soc_ma /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; --- linux-2.6.27-rc5.orig/sound/soc/omap/n810.c 2008-08-29 09:32:25.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/omap/n810.c 2008-09-01 08:30:10.000000000 +0200 @@ -290,6 +290,7 @@ static struct snd_soc_machine snd_soc_ma /* Audio private data */ static struct aic3x_setup_data n810_aic33_setup = { + .i2c_bus = 0, .i2c_address = 0x18, .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, -- Jean Delvare

3 4

[alsa-devel] master branch on sound-2.6.git rebuilt
by Takashi Iwai 01 Sep '08

01 Sep '08

Hi, FYI, I rebuilt the master branch on sound-2.6.git tree now because of the conflict with sparc tree. Basically only topic/remove-assert is changed and topic/sgbuf-large-pages branch is rebased. Other topic branches are kept as is. The master is the straight merge results of all topics now. The old master branch before rebase is saved as old/master on sound-2.6.git. It'll be removed sometime later. If you've developed patches based on the master branch, try to rebase like the following: - mark your current branch % git branch my-master To be sure, keep it also as a backup branch in case you broke via rebase: % git branch backup-master - mark the old origin/master % git branch old-master origin/master - fetch the sound-2.6.git % git fetch origin (or whatever) - rebase via --onto option % git rebase --onto origin/master old-master my-master This will move the changes done on local branch onto origin/master, and make the result as the current branch (i.e. the new my-master). - remove temporary branches if needed % git branch -D old-master % git branch -D backup-master Takashi

1 0

[alsa-devel] [PATCH] ASoC: fix pxa2xx-i2s clk_get call
by Dmitry Baryshkov 01 Sep '08

01 Sep '08

pxa2xx-isa: probe actual device and use it for clk_get call thus fixing error during startup hook Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-i2s.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 files changed, 39 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 8548818..c796b18 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -16,6 +16,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/clk.h> +#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -81,7 +82,6 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - clk_i2s = clk_get(NULL, "I2SCLK"); if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); @@ -152,6 +152,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); + BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); pxa_i2s_wait(); @@ -317,6 +318,43 @@ struct snd_soc_dai pxa_i2s_dai = { EXPORT_SYMBOL_GPL(pxa_i2s_dai); +static int pxa2xx_i2s_probe(struct platform_device *dev) +{ + clk_i2s = clk_get(&dev->dev, "I2SCLK"); + return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0; +} + +static int __devexit pxa2xx_i2s_remove(struct platform_device *dev) +{ + clk_put(clk_i2s); + clk_i2s = ERR_PTR(-ENOENT); + return 0; +} + +static struct platform_driver pxa2xx_i2s_driver = { + .probe = pxa2xx_i2s_probe, + .remove = __devexit_p(pxa2xx_i2s_remove), + + .driver = { + .name = "pxa2xx-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa2xx_i2s_init(void) +{ + clk_i2s = ERR_PTR(-ENOENT); + return platform_driver_register(&pxa2xx_i2s_driver); +} + +static void __exit pxa2xx_i2s_exit(void) +{ + platform_driver_unregister(&pxa2xx_i2s_driver); +} + +module_init(pxa2xx_i2s_init); +module_exit(pxa2xx_i2s_exit); + /* Module information */ MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); -- 1.5.6.3

5 6

[alsa-devel] [PATCH] ASoC: Convert uda1380 to a new-style i2c driver
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the uda1380 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- sound/soc/codecs/uda1380.c | 109 +++++++++++++++++++++----------------------- sound/soc/codecs/uda1380.h | 1 2 files changed, 54 insertions(+), 56 deletions(-) --- linux-2.6.27-rc4.orig/sound/soc/codecs/uda1380.c 2008-08-26 16:12:35.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/uda1380.c 2008-08-27 22:53:04.000000000 +0200 @@ -701,87 +701,86 @@ static struct snd_soc_device *uda1380_so #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */ - -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver uda1380_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ - -static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int uda1380_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = uda1380_socdev; struct uda1380_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("uda1380: failed to attach codec at addr %x\n", addr); - goto err; - } - ret = uda1380_init(socdev, setup->dac_clk); - if (ret < 0) { + if (ret < 0) pr_err("uda1380: failed to initialise UDA1380\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int uda1380_i2c_detach(struct i2c_client *client) +static int uda1380_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int uda1380_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, uda1380_codec_probe); -} +static const struct i2c_device_id uda1380_i2c_id[] = { + { "uda1380", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); static struct i2c_driver uda1380_i2c_driver = { .driver = { .name = "UDA1380 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_UDA1380, - .attach_adapter = uda1380_i2c_attach, - .detach_client = uda1380_i2c_detach, - .command = NULL, + .probe = uda1380_i2c_probe, + .remove = uda1380_i2c_remove, + .id_table = uda1380_i2c_id, }; -static struct i2c_client client_template = { - .name = "UDA1380", - .driver = &uda1380_i2c_driver, -}; +static int uda1380_add_i2c_device(struct platform_device *pdev, + const struct uda1380_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "uda1380", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&uda1380_i2c_driver); + return -ENODEV; +} #endif static int uda1380_probe(struct platform_device *pdev) @@ -806,11 +805,8 @@ static int uda1380_probe(struct platform uda1380_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&uda1380_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = uda1380_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -833,6 +829,7 @@ static int uda1380_remove(struct platfor snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&uda1380_i2c_driver); #endif kfree(codec); --- linux-2.6.27-rc4.orig/sound/soc/codecs/uda1380.h 2008-08-26 16:12:35.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/uda1380.h 2008-08-27 22:52:32.000000000 +0200 @@ -73,6 +73,7 @@ #define R23_AGC_EN 0x0001 struct uda1380_setup_data { + int i2c_bus; unsigned short i2c_address; int dac_clk; #define UDA1380_DAC_CLK_SYSCLK 0 -- Jean Delvare

2 1

[alsa-devel] [PATCH] ASoC: Convert ak4535 to a new-style i2c driver
by Jean Delvare 01 Sep '08

01 Sep '08

Convert the ak4535 codec driver to the new (standard) device driver binding model. After this change, AK4535 devices are no longer discovered automatically and must instead be instantiated explicitly. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- sound/soc/codecs/ak4535.c | 110 +++++++++++++++++++++------------------------ sound/soc/codecs/ak4535.h | 1 2 files changed, 54 insertions(+), 57 deletions(-) --- linux-2.6.27-rc4.orig/sound/soc/codecs/ak4535.c 2008-08-26 16:12:35.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/ak4535.c 2008-08-27 23:01:26.000000000 +0200 @@ -535,87 +535,85 @@ static struct snd_soc_device *ak4535_soc #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ - -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver ak4535_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int ak4535_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = ak4535_socdev; - struct ak4535_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - printk(KERN_ERR "failed to attach codec at addr %x\n", addr); - goto err; - } - ret = ak4535_init(socdev); - if (ret < 0) { + if (ret < 0) printk(KERN_ERR "failed to initialise AK4535\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int ak4535_i2c_detach(struct i2c_client *client) +static int ak4535_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int ak4535_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, ak4535_codec_probe); -} +static const struct i2c_device_id ak4535_i2c_id[] = { + { "ak4535", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver ak4535_i2c_driver = { .driver = { .name = "AK4535 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_AK4535, - .attach_adapter = ak4535_i2c_attach, - .detach_client = ak4535_i2c_detach, - .command = NULL, + .probe = ak4535_i2c_probe, + .remove = ak4535_i2c_remove, + .id_table = ak4535_i2c_id, }; -static struct i2c_client client_template = { - .name = "AK4535", - .driver = &ak4535_i2c_driver, -}; +static int ak4535_add_i2c_device(struct platform_device *pdev, + const struct ak4535_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ak4535_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ak4535", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&ak4535_i2c_driver); + return -ENODEV; +} #endif static int ak4535_probe(struct platform_device *pdev) @@ -648,12 +646,9 @@ static int ak4535_probe(struct platform_ ak4535_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; codec->hw_read = (hw_read_t)i2c_master_recv; - ret = i2c_add_driver(&ak4535_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = ak4535_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -678,6 +673,7 @@ static int ak4535_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&ak4535_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc4.orig/sound/soc/codecs/ak4535.h 2008-08-26 16:12:35.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/ak4535.h 2008-08-27 23:01:04.000000000 +0200 @@ -37,6 +37,7 @@ #define AK4535_CACHEREGNUM 0x10 struct ak4535_setup_data { + int i2c_bus; unsigned short i2c_address; }; -- Jean Delvare

2 1

[alsa-devel] [PATCH] Automatically use ALC268_TOSHIBA for Toshiba L305 Laptop
by Travis Place 01 Sep '08

01 Sep '08

Changelog: Added Subsystem IDs (0x1179, 0xff64) for the Toshiba Satellite L305 laptop, so it automatically uses the ALC268_TOSHIBA quirk. Signed-off-by: Travis Place <wishie(a)wishie.net> Patch Attached. Thanks, Travis Place (wishie)

2 1

[alsa-devel] snd_pcm_avail_update does not return until I have 32768 bytes ready
by William Estrada 01 Sep '08

01 Sep '08

Hi group, I am having a little problem with ALSA. My problem is that snd_pcm_readi does not return until i have 32768 bytes available. I want to get an interrupt back when I have 80 bytes ( 1/100 of a second ). This delay is about 4.09 seconds which is much too long to want. My question is how do I tell ALSA to return from snd_pcm_avail_update when I have 80 bytes ready to read? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] snd-soc-cs4270: Convert to a new-style i2c driver (work in progress)
by Jean Delvare 31 Aug '08

31 Aug '08

Hi Tibur, I am in the process of converting your cs4270 codec driver from the legacy i2c model to the new (standard) one. This is work in progress. The patch below converts the cs4270 driver itself. However we also need to convert its users. As far as I can see there's only one user at this point: mpc8610_hpcd. The problem is that this driver doesn't look like the other codec drivers I have already converted. So, we need to add code to instantiate the cs4270 i2c device, but I don't know where this should happen. Given that the mpc8610_hpcd is apparently based on Open Firmware, I guess that the i2c device should be instantiated directly by the platform code. I see that the device is declared in mpc8610_hpcd.dts, so maybe it's already done and my patch should work already? What do you think? Thanks. * * * * * Convert the cs4270 codec driver to the new (standard) device driver binding model. After this change, CS4720 devices are no longer discovered automatically and must instead be instantiated explicitly. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 79 +++++++++------------------------------------ 1 file changed, 16 insertions(+), 63 deletions(-) --- linux-2.6.27-rc5.orig/sound/soc/codecs/cs4270.c 2008-08-31 14:23:34.000000000 +0200 +++ linux-2.6.27-rc5/sound/soc/codecs/cs4270.c 2008-08-31 15:16:43.000000000 +0200 @@ -272,17 +272,6 @@ static int cs4270_set_dai_fmt(struct snd } /* - * A list of addresses on which this CS4270 could use. I2C addresses are - * 7 bits. For the CS4270, the upper four bits are always 1001, and the - * lower three bits are determined via the AD2, AD1, and AD0 pins - * (respectively). - */ -static const unsigned short normal_i2c[] = { - 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END -}; -I2C_CLIENT_INSMOD; - -/* * Pre-fill the CS4270 register cache. * * We use the auto-increment feature of the CS4270 to read all registers in @@ -490,32 +479,18 @@ static int cs4270_mute(struct snd_soc_da #endif -static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); +static int cs4270_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id); -/* - * Notify the driver that a new I2C bus has been found. - * - * This function is called for each I2C bus in the system. The function - * then asks the I2C subsystem to probe that bus at the addresses on which - * our device (the CS4270) could exist. If a device is found at one of - * those addresses, then our probe function (cs4270_i2c_probe) is called. - */ -static int cs4270_i2c_attach(struct i2c_adapter *adapter) -{ - return i2c_probe(adapter, &addr_data, cs4270_i2c_probe); -} - -static int cs4270_i2c_detach(struct i2c_client *client) +static int cs4270_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); codec->control_data = NULL; kfree(codec->reg_cache); codec->reg_cache = NULL; - kfree(client); return 0; } @@ -525,14 +500,19 @@ static const struct snd_kcontrol_new cs4 CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; +static const struct i2c_device_id cs4270_i2c_id[] = { + { "cs4270", 0 }, + { } +}; + static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "CS4270 I2C", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_CS4270, - .attach_adapter = cs4270_i2c_attach, - .detach_client = cs4270_i2c_detach, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, + .id_table = cs4270_i2c_id, }; /* @@ -555,17 +535,14 @@ static struct snd_soc_device *cs4270_soc /* * Initialize the I2C interface of the CS4270 * - * This function is called for whenever the I2C subsystem finds a device - * at a particular address. - * * Note: snd_soc_new_pcms() must be called before this function can be called, * because of snd_ctl_add(). */ -static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) +static int cs4270_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = cs4270_socdev; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c_client = NULL; int i; int ret = 0; @@ -578,12 +555,6 @@ static int cs4270_i2c_probe(struct i2c_a /* Note: codec_dai->codec is NULL here */ - i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); - if (!i2c_client) { - printk(KERN_ERR "cs4270: could not allocate I2C client\n"); - return -ENOMEM; - } - codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); if (!codec->reg_cache) { printk(KERN_ERR "cs4270: could not allocate register cache\n"); @@ -592,11 +563,6 @@ static int cs4270_i2c_probe(struct i2c_a } i2c_set_clientdata(i2c_client, codec); - strcpy(i2c_client->name, "CS4270"); - - i2c_client->driver = &cs4270_i2c_driver; - i2c_client->adapter = adapter; - i2c_client->addr = addr; /* Verify that we have a CS4270 */ @@ -612,18 +578,10 @@ static int cs4270_i2c_probe(struct i2c_a goto error; } - printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr); + printk(KERN_INFO "cs4270: found device at I2C address %X\n", + i2c_client->addr); printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); - /* Tell the I2C layer a new client has arrived */ - - ret = i2c_attach_client(i2c_client); - if (ret) { - printk(KERN_ERR "cs4270: could not attach codec, " - "I2C address %x, error code %i\n", addr, ret); - goto error; - } - codec->control_data = i2c_client; codec->read = cs4270_read_reg_cache; codec->write = cs4270_i2c_write; @@ -651,17 +609,12 @@ static int cs4270_i2c_probe(struct i2c_a return 0; error: - if (codec->control_data) { - i2c_detach_client(i2c_client); - codec->control_data = NULL; - } + codec->control_data = NULL; kfree(codec->reg_cache); codec->reg_cache = NULL; codec->reg_cache_size = 0; - kfree(i2c_client); - return ret; } -- Jean Delvare

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[alsa-devel] [PATCH] alsa-lib: Don't call the simple mixer interface "legacy"
by Lennart Poettering 31 Aug '08

31 Aug '08

The simpler mixer interface is not legacy, so the header file shouldn't claim otherwise. Fixes bug 3569 Lennart --- include/mixer.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/mixer.h b/include/mixer.h index 59deb9f..df92164 100644 --- a/include/mixer.h +++ b/include/mixer.h @@ -88,7 +88,7 @@ typedef int (*snd_mixer_event_t)(snd_mixer_class_t *class_, unsigned int mask, /** Mixer element type */ typedef enum _snd_mixer_elem_type { - /* Simple (legacy) mixer elements */ + /* Simple mixer elements */ SND_MIXER_ELEM_SIMPLE, SND_MIXER_ELEM_LAST = SND_MIXER_ELEM_SIMPLE } snd_mixer_elem_type_t; @@ -166,7 +166,7 @@ int snd_mixer_class_set_compare(snd_mixer_class_t *class_, snd_mixer_compare_t c * \{ */ -/* Simple (legacy) mixer elements API */ +/* Simple mixer elements API */ /** Mixer simple element channel identifier */ typedef enum _snd_mixer_selem_channel_id { -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] alsa-lib: Reword doxygen doc for snd_pcm_delay()
by Lennart Poettering 31 Aug '08

31 Aug '08

snd_pcm_delay() is for synchronization purposes, it returns the overall latency of the stream, not just the latency induced by the hardware playback buffer. The documentation is a bit misleading about this fact, and some people already misunderstood it. So let's reword this to make the explanation clearer and explicit. This is a result of the discussions of the thread this mail belongs to: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008456.html Lennart --- src/pcm/pcm.c | 22 +++++++++++++++++----- 1 files changed, 17 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index ee2300e..ae89795 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -939,11 +939,23 @@ int snd_pcm_hwsync(snd_pcm_t *pcm) * \param delayp Returned delay in frames * \return 0 on success otherwise a negative error code * - * Delay is distance between current application frame position and - * sound frame position. - * It's positive and less than buffer size in normal situation, - * negative on playback underrun and greater than buffer size on - * capture overrun. + * For playback the delay is defined as the time that a frame that is written + * to the PCM stream shortly after this call will take to be actually + * audible. It is as such the overall latency from the write call to the final + * DAC. + * + * For capture the delay is defined as the time that a frame that was + * digitized by the audio device takes until it can be read from the PCM + * stream shortly after this call returns. It is as such the overall latency + * from the initial ADC to the read call. + * + * Please note that hence in case of a playback underrun this value will not + * necessarily got down to 0. + * + * If the application is interested in the fill level of the playback buffer + * of the device, it should use snd_pcm_avail_update(). The + * value returned by that call is not directly related to the delay, since the + * latter might include some additional, fixed latencies the former does not. * * Note this function does not update the actual r/w pointer * for applications. The function #snd_pcm_avail_update() -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] alsa-lib: Make sure SND_PCM_NO_xxx flags don't get lost when nonblocking mode is enabled
by Lennart Poettering 31 Aug '08

31 Aug '08

The plug PCM copies the 'mode' field from the slave PCM. If blocking mode is enabled for the plug PCM the mode is subsequently overwritten with the original requested 'mode'. If non-blocking mode is requested this does not happen. Because the hw PCM synthesizes the 'mode' from the actual file descriptor flags no SND_PCM_NO_xxx will ever be set for it. This has the effect that the 'mode' of the plug PCM will also not include those flags anymore -- unless they are overwritten as mentioned above. This basically means SND_PCM_NO_xxx is ignored for plug:hw:4711 style device strings opened in non-blocking mode. You can easily test this with "aplay --channels 7 --disable-channels -f S16_LE -r 44100 -D plug:hw:0" on a device that cannot do 7 channels. Normally this call should fail, however if you add "-N" to the command line this call will succeed. This patch simply copies the SND_PCM_NO_xxx flags back into the 'mode' field in case we don't overwrite it with the original anyway. Probably closes bug 3571 for good. Lennart --- src/pcm/pcm_hw.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 1e1889c..c8d9a62 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1507,7 +1507,13 @@ int _snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, /* revert to blocking mode for read/write access */ snd_pcm_hw_nonblock(*pcmp, 0); (*pcmp)->mode = mode; - } + } else + /* make sure the SND_PCM_NO_xxx flags don't get lost on the + * way */ + (*pcmp)->mode |= mode & (SND_PCM_NO_AUTO_RESAMPLE| + SND_PCM_NO_AUTO_CHANNELS| + SND_PCM_NO_AUTO_FORMAT| + SND_PCM_NO_SOFTVOL); hw = (*pcmp)->private_data; if (format != SND_PCM_FORMAT_UNKNOWN) -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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Re: [alsa-devel] ALSA, select() question.
by William Estrada 29 Aug '08

29 Aug '08

Jaroslav, I got you Email address from http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09605.html. <http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09605.html> I am trying to write a C program that simply reads from a ALSA device and then writes back to it. The program work with one major problem, there is a 4.1 second delay between the read and the write. This delay is caused by the select() not returning at the right time. In you post you said to use: > Yes, but you have to create revents for snd_seq_poll_descriptors_revents() > to get the right event then. Don't use values from select() directly. > > Jaroslav > Do you have some sample code that I could look at so that I could fix my problem? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] Issues with alsa-info.sh - Trying to track down the problem.
by Travis Place 29 Aug '08

29 Aug '08

Since the move to use alsa-project.org as the main place for uploads, some users have reported that no URL is being printed at the end of the script. I am trying to track down the problem, but would appreciate some help if possible. Thanks in advance, Travis Place (wishie)

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[alsa-devel] [PATCH] ice1724 MIDI on PHASE 22 and PHASE 28 and a few minor fixes
by Vedran Miletić 29 Aug '08

29 Aug '08

Changelog: * added info about PHASE 28 (as far as I could collect, I don't have a card) * added info about TS22PCI and new revisions of PHASE 22 * disable 192k on PHASE 22 as AK4524 I2S doesn't support it * enable midi for both PHASE 22 and PHASE 28 * enable SPDIF reciever on PHASE 22 MIDI works for me, I have tried loopback and it plays good. Thanks for help Pavel and Takashi. diff --git a/pci/ice1712/phase.c b/pci/ice1712/phase.c index f5acdee..6e596e9 100644 --- a/pci/ice1712/phase.c +++ b/pci/ice1712/phase.c @@ -24,7 +24,8 @@ /* PHASE 22 overview: * Audio controller: VIA Envy24HT-S (slightly trimmed down version of Envy24HT) * Analog chip: AK4524 (partially via Philip's 74HCT125) - * Digital receiver: CS8414-CS (not supported in this release) + * Digital receiver: CS8414-CS (supported in this release) + * PHASE 22 revision 2.0 and Terrasoniq/Musonik TS22PCI have CS8416 (support status unknown, please test and report) * * Envy connects to AK4524 * - CS directly from GPIO 10 @@ -33,6 +34,12 @@ * CDTI may be completely blocked by 74HCT125's gate #1 controlled by GPIO 3 */ +/* PHASE 28 overview: + * Audio controller: VIA Envy24HT (full untrimmed version, 8 ins 8 outs) + * Analog chip: WM8770 (8 channel DAC, 2 channel ADC) + * Digital receiver: CS8414-CS (supported in this release) + */ + #include <asm/io.h> #include <linux/delay.h> #include <linux/interrupt.h> @@ -161,9 +168,9 @@ static int __devinit phase22_add_controls(struct snd_ice1712 *ice) } static unsigned char phase22_eeprom[] __devinitdata = { - [ICE_EEP2_SYSCONF] = 0x00, /* 1xADC, 1xDACs */ + [ICE_EEP2_SYSCONF] = 0x28, /* clock 512, mpu 401, spdif-in/1xADC, 1xDACs */ [ICE_EEP2_ACLINK] = 0x80, /* I2S */ - [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit */ + [ICE_EEP2_I2S] = 0xf0, /* vol, 96k, 24bit */ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ [ICE_EEP2_GPIO_DIR] = 0xff, [ICE_EEP2_GPIO_DIR1] = 0xff, @@ -177,7 +184,7 @@ static unsigned char phase22_eeprom[] __devinitdata = { }; static unsigned char phase28_eeprom[] __devinitdata = { - [ICE_EEP2_SYSCONF] = 0x0b, /* clock 512, spdif-in/ADC, 4DACs */ + [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, spdif-in/1xADC, 4xDACs */ [ICE_EEP2_ACLINK] = 0x80, /* I2S */ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ -- Vedran Miletić

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[alsa-devel] [PATCH] ice1724: enable MIDI on Terratec Aureon 7.1 Universe
by Maximilian Rehkopf 29 Aug '08

29 Aug '08

This enables MIDI on Terratec Aureon 7.1 Universe cards specifically. Apparently the other envy24ht-based Aureon cards do not have MIDI ports, hence I added a Universe specific eeprom array. The newer cards (Aureon 7.1 PCI / 5.1 PCI/Fun) have MIDI but use a CMI DSP so this driver does not concern them at all. I am new to git, do I have to perform any git operations regarding the patch myself? Signed-off-by: Maximilian Rehkopf <otakon(a)gmx.net> -- Maximilian Rehkopf <otakon(a)gmx.net>

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[alsa-devel] PATCH - via82xx vt1618 7.1 Audio Support
by John L. Utz III 29 Aug '08

29 Aug '08

1. Implement 7.1 Output for vt1618 codec. 2. Fix typos in comments from my previous 1617a patch. Signed-off-by: John L. Utz III <john.utz(a)dmx.com> diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..89f3050 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..7d5abf5 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * noodling the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3498,7 +3498,7 @@ static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */ - /* grab our desirec bits, then mash them together in a manner + /* grab our desired bits, then mash them together in a manner * consistent with Table 6 on page 17 in the 1617a docs */ usSM51 = snd_ac97_read(pac97, 0x7a) >> 14; @@ -3576,6 +3576,201 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } +/* VIA VT1618 8 CHANNEL AC97 CODEC + * + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack and it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get 5.1 output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be applicable + * to them. + */ + +struct vt1618_uaj_item { + unsigned short mask; + unsigned short shift; + const char *items[4]; +}; + +/* This list reflects the vt1618 docs for Vendor Defined Register 0x60. */ + +static struct vt1618_uaj_item vt1618_uaj[3] = { + { + /* speaker jack */ + .mask = 0x03, + .shift = 0, + .items = { + "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In" + } + }, + { + /* line jack */ + .mask = 0x0c, + .shift = 2, + .items = { + "Surround Out", "DAC Unmixed Out", "Line In", "Mic In" + } + }, + { + /* mic jack */ + .mask = 0x30, + .shift = 4, + .items = { + "Center LFE Out", "DAC Unmixed Out", "Line In", "Mic In" + }, + }, +}; + +static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, + vt1618_uaj[kcontrol->private_value].items, + 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned short datpag, uaj; + + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + mutex_lock(&pac97->page_mutex); + + datpag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + uaj = snd_ac97_read(pac97, 0x60) & + vt1618_uaj[kcontrol->private_value].mask; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, datpag); + mutex_unlock(&pac97->page_mutex); + + ucontrol->value.enumerated.item[0] = uaj >> + vt1618_uaj[kcontrol->private_value].shift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return ac97_update_bits_page(snd_kcontrol_chip(kcontrol), 0x60, + vt1618_uaj[kcontrol->private_value].mask, + ucontrol->value.enumerated.item[0]<< + vt1618_uaj[kcontrol->private_value].shift, + 0); +} + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *txt_aux[] = {"Aux In", "Back Surr Out"}; + + return ac97_enum_text_info(kcontrol, uinfo, txt_aux, 2); +} + +static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = + (snd_ac97_read(snd_kcontrol_chip(kcontrol), 0x5c) & 0x0008)>>3; + return 0; +} + +static int snd_ac97_vt1618_aux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* toggle surround rear dac power */ + + snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); + + /* toggle aux in surround rear out jack */ + + return snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + AC97_SINGLE("Exchange Center/LFE", 0x5a, 8, 1, 0), + AC97_SINGLE("DC Offset", 0x5a, 10, 1, 0), + AC97_SINGLE("Soft Mute", 0x5c, 0, 1, 1), + AC97_SINGLE("Headphone Amp", 0x5c, 5, 1, 1), + AC97_DOUBLE("Back Surr Volume", 0x5e, 8, 0, 31, 1), + AC97_SINGLE("Back Surr Switch", 0x5e, 15, 1, 1), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack Mode", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Jack Mode", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Jack Mode", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 2 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Jack Mode", + .info = snd_ac97_vt1618_aux_info, + .get = snd_ac97_vt1618_aux_get, + .put = snd_ac97_vt1618_aux_put, + } +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97)

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.27-rc5-122-g1328f04
by Takashi Iwai 29 Aug '08

29 Aug '08

At Fri, 29 Aug 2008 11:35:25 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > commit 1328f04260822e45a2717b65a0e2bc0c9c1255be > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Fri Aug 29 11:29:39 2008 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Fri Aug 29 11:29:39 2008 +0200 > > ALSA: intel8x0: implement ac97_clock whitelist > > The AC97 clock detection is not accurate in some cases. This patch adds > an initial whitelist for audio devices gathered from RedHat's bugzilla: > > https://bugzilla.redhat.com/show_bug.cgi?id=441087 > > As a side effect, white-listing might speedup kernel booting (AC97 clock > measuring code is not activated). > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> (snip) > +struct intel8x0_clock_list { > + unsigned short subvendor; > + unsigned short subdevice; > + unsigned int rate; > +}; > + > +static struct intel8x0_clock_list intel8x0_clock_list[] __devinitdata = { > + { 0x1028, 0x00be, 44100 }, /* Analog Devices AD1885 */ > + { 0x1028, 0x0177, 48000 }, /* Analog Devices AD1980 */ > + { 0x1043, 0x80f3, 48000 }, /* Analog Devices AD1985 */ > + { 0x0000, 0x0000, 00000 } /* terminator */ > +}; > + > +static int __devinit intel8x0_in_clock_list(struct intel8x0 *chip) > +{ > + struct pci_dev *pci = chip->pci; > + struct intel8x0_clock_list *wl; > + > + for (wl = intel8x0_clock_list; wl->subvendor; wl++) { > + if (wl->subvendor == pci->subsystem_vendor && > + wl->subdevice == pci->subsystem_device) { > + printk(KERN_INFO "intel8x0: white list rate for %04x:%04x is %i\n", > + pci->subsystem_vendor, > + pci->subsystem_device, wl->rate); > + chip->ac97_bus->clock = wl->rate; > + return 1; > + } > + } > + return 0; > +} Use snd_pci_quirk_lookup() please. But, a better fix would be to fix the clock measurement. AFAIK, this seems to happen when the driver is built-in. Thus I guess it's rather a problem of time-keeping core side. Takashi

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Re: [alsa-devel] [PATCH] HDA: patch-realtek: toshiba dynabook SS RX1 which use ALC262
by Takashi Iwai 28 Aug '08

28 Aug '08

At Thu, 28 Aug 2008 13:40:23 +0000, Hiroshi Miura wrote: > > Recent patch for toshiba dynabook SS RX1 against recent snapshot; Thanks. Now applied and pushed to my tree. Takashi > Hiroshi > > 2008/8/26 Takashi Iwai <tiwai(a)suse.de>: > > Hi Miura-san, > > > > At Tue, 26 Aug 2008 17:39:17 +0900, > > 三浦 SPS wrote: > >> > >> Hi Takashi, > >> > >> When I test with recent tree on the Toshiba laptop, Toshiba dynabook SS RX1, > >> some mixer items are not controlled by ALSA. > >> Some mixer items appears as an OSS mixer control. > >> Chipset is HDA intel with Realtek ALC262. > >> > >> I make this patch by using as many datas and functions in existent model. > >> > >> Toshiba hda-intel ALC262 should be controlled as same model as fujitsu. > >> There is an only difference in nid of head phone jack. > >> > >> A related bugtrack id is #3386 and I also post it there. > >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3386 > > > > Thanks. The changes look OK, but your MUA broke the embedded patch. > > > > BTW, I merged other patches to add more support in patch_realtek.c, > > and one of them is the support for Toshiba S06. > > Could you check whether it's compatible with yours? > > It seems S06 uses the digital mic instead of analog mic, and this > > can be a difference. > > > > Signed-off-by: Hiroshi Miura <miurahr(a)acm.org> > --- > --- alsa-driver-unstable/alsa-kernel/pci/hda/patch_realtek.c.orig 2008-08-27 > 06:05:19.000000000 +0000 > +++ alsa-driver-unstable/alsa-kernel/pci/hda/patch_realtek.c 2008-08-27 > 11:28:04.000000000 +0000 > @@ -103,6 +103,7 @@ enum { > ALC262_LENOVO_3000, > ALC262_NEC, > ALC262_TOSHIBA_S06, > + ALC262_TOSHIBA_RX1, > ALC262_AUTO, > ALC262_MODEL_LAST /* last tag */ > }; > @@ -9705,6 +9706,25 @@ static struct snd_kcontrol_new alc262_le > { } /* end */ > }; > > +static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { > + HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), > + { > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Master Playback Switch", > + .info = snd_hda_mixer_amp_switch_info, > + .get = snd_hda_mixer_amp_switch_get, > + .put = alc262_sony_master_sw_put, > + .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), > + }, > + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), > + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), > + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), > + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), > + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), > + { } /* end */ > +}; > + > /* additional init verbs for Benq laptops */ > static struct hda_verb alc262_EAPD_verbs[] = { > {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, > @@ -10169,6 +10189,24 @@ static struct hda_verb alc262_HP_BPC_Wil > { } > }; > > +static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { > + > + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */ > + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, > + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, > + > + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */ > + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */ > + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, > + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, > + > + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */ > + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, > + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, > + {} > +}; > + > + > #ifdef CONFIG_SND_HDA_POWER_SAVE > #define alc262_loopbacks alc880_loopbacks > #endif > @@ -10258,6 +10296,8 @@ static const char *alc262_models[ALC262_ > [ALC262_ULTRA] = "ultra", > [ALC262_LENOVO_3000] = "lenovo-3000", > [ALC262_NEC] = "nec", > + [ALC262_TOSHIBA_S06] = "toshiba-s06", > + [ALC262_TOSHIBA_RX1] = "toshiba-rx1", > [ALC262_AUTO] = "auto", > }; > > @@ -10292,7 +10332,7 @@ static struct snd_pci_quirk alc262_cfg_t > SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), > SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), > SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", > - ALC262_SONY_ASSAMD), > + ALC262_TOSHIBA_RX1), > SND_PCI_QUIRK(0x1179, 0x0268, "Toshiba S06", ALC262_TOSHIBA_S06), > SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), > SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), > @@ -10500,6 +10540,18 @@ static struct alc_config_preset alc262_p > .unsol_event = alc262_toshiba_s06_unsol_event, > .init_hook = alc262_toshiba_s06_init_hook, > }, > + [ALC262_TOSHIBA_RX1] = { > + .mixers = { alc262_toshiba_rx1_mixer }, > + .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs }, > + .num_dacs = ARRAY_SIZE(alc262_dac_nids), > + .dac_nids = alc262_dac_nids, > + .hp_nid = 0x03, > + .num_channel_mode = ARRAY_SIZE(alc262_modes), > + .channel_mode = alc262_modes, > + .input_mux = &alc262_capture_source, > + .unsol_event = alc262_hippo_unsol_event, > + .init_hook = alc262_hippo_automute, > + }, > }; > > static int patch_alc262(struct hda_codec *codec) > > -- > HIroshi Miura > NTT DATA Corp. and IPA OSS center > （株）NTTデータ /（独）情報処理推進機構 > 三浦広志 >

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[alsa-devel] [PATCH] ASoC: Convert DAPM pop time configuration to debugfs
by Mark Brown 28 Aug '08

28 Aug '08

Pop time configuration is a debugging feature which shouldn't be used in normal operation so move it to debugfs where it is clearer. This also simplifies the code since debugfs provides standard facilities for exposing simple integer values to user space. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- It would be nice to apply this to 2.6.27 since the sysfs file was only added in the 2.6.27 cycle but it's not critical either way. sound/soc/soc-dapm.c | 53 ++++++++++++++++++------------------------------- 1 files changed, 20 insertions(+), 33 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7a88f76..c016426 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -38,6 +38,7 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -67,7 +68,9 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); -static unsigned int pop_time; +static struct dentry *asoc_debugfs; + +static u32 pop_time; static void pop_wait(void) { @@ -817,51 +820,35 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -/* pop/click delay times */ -static ssize_t dapm_pop_time_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - return sprintf(buf, "%d\n", pop_time); -} - -static ssize_t dapm_pop_time_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) - -{ - unsigned long val; - - if (strict_strtoul(buf, 10, &val) >= 0) - pop_time = val; - else - printk(KERN_ERR "Unable to parse pop_time setting\n"); - - return count; -} - -static DEVICE_ATTR(dapm_pop_time, 0744, dapm_pop_time_show, - dapm_pop_time_store); - int snd_soc_dapm_sys_add(struct device *dev) { int ret = 0; - if (dapm_status) { - ret = device_create_file(dev, &dev_attr_dapm_widget); + if (!dapm_status) + return 0; - if (ret == 0) - ret = device_create_file(dev, &dev_attr_dapm_pop_time); - } + ret = device_create_file(dev, &dev_attr_dapm_widget); + if (ret != 0) + return ret; - return ret; + asoc_debugfs = debugfs_create_dir("asoc", NULL); + if (!IS_ERR(asoc_debugfs)) + debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, + &pop_time); + else + asoc_debugfs = NULL; + + return 0; } static void snd_soc_dapm_sys_remove(struct device *dev) { if (dapm_status) { - device_remove_file(dev, &dev_attr_dapm_pop_time); device_remove_file(dev, &dev_attr_dapm_widget); } + + if (asoc_debugfs) + debugfs_remove_recursive(asoc_debugfs); } /* free all dapm widgets and resources */ -- 1.5.6.5

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[alsa-devel] modifying VIA8237.conf - does it currently work correctly for somebody?
by John L. Utz III 28 Aug '08

28 Aug '08

Hi; I am sorry for the vagueness of the questions, but now that i have some understanding of how these files are supposed to work i want to get this right! to support 4.0/5.1/7.1 for the vt1618 codec i have to change a different set of twiddlers than what is called out in the current version of the conf file. worse yet, the twiddlers called out in the current file dont even *exist* on the vt1618: { name "Surround Down Mix" preserve true value off lock true optional true } { name "Center/LFE Down Mix" preserve true value off lock true optional true } How do i add new stuff without breaking old stuff? is there a sub-identity that i could use? tnx! johnu

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[alsa-devel] Report of partially supported STAC92HD73* onboard a DELL Studio/Inspiron 1535
by Travis Place 28 Aug '08

28 Aug '08

Current status: Internal speakers: Working Internal microphone: Not working External microphone: Not working Headphones: Not working Card Information (alsa-info): http://www.alsa-project.org/db/?f=166dedf1effd5c5ba0ee91075a0f0ca4da65f3b4 Notes: i had the user upgrade to 1.0.17 of alsa-driver, but didnt suggest git code, as i couldnt find any immediate difference for his card in the later code. Thanks, Travis Place (wishie)

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[alsa-devel] i think ac97 needs an is_surback_on() function
by John L. Utz III 28 Aug '08

28 Aug '08

ac97_patch.c provides two helper functions that wrap the channel_mode variable, but they only take you out to 5.1 Audio: static inline int is_surround_on(struct snd_ac97 *ac97) { return ac97->channel_mode >= 1; } static inline int is_clfe_on(struct snd_ac97 *ac97) { return ac97->channel_mode >= 2; } For 7.1 Audio, i think there needs to be a test for 'Surround Back' or 'Back Surround' static inline int is_surback_on(struct snd_ac97 *ac97) { return ac97->channel_mode >= 4; } Am i correct? or am i mistaken? agreement or corrections appreciated! I think that i need this function for implementing vt1618_update_jacks() because vt1618 is 7.1 Audio: static void vt1618_update_jacks(struct snd_ac97 *ac97) { unsigned short usSft = 0, usMsk = 0x0003; if(is_surround_on(ac97)) ac97_update_bits_page(ac97, 0x60, 0x000C, ucontrol->value.enumerated.item[0]<<2, 0); if(is_clfe_on(ac97)) ac97_update_bits_page(ac97, 0x60, 0x0030, ucontrol->value.enumerated.item[0]<<4, 0); if(is_surback_on(ac97)) { /* toggle surround rear dac power */ snd_ac97_update_bits(ac97, 0x5c, 0x0008, ucontrol->value.enumerated.item[0] << 3); /* toggle aux in surround rear out jack */ snd_ac97_update_bits(ac97, 0x76, 0x0008, ucontrol->value.enumerated.item[0] << 3); } }

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[alsa-devel] [PATCH] ASoC: Convert wm8731 to a new-style i2c driver (v2, testers wanted)
by Jean Delvare 27 Aug '08

27 Aug '08

Convert the wm8731 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Tested-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- This second version fixes a bug reported by Manuel Lauss where codec->control_data was used before being initialized. Warning: this assumes that the corgi, poodle and eti_b1 have their codec chips on I2C bus 0, which may or may not be the case. Someone with these machines please test and report. include/linux/i2c-id.h | 1 sound/soc/at91/eti_b1_wm8731.c | 1 sound/soc/codecs/wm8731.c | 108 +++++++++++++++++++--------------------- sound/soc/codecs/wm8731.h | 1 sound/soc/pxa/corgi.c | 1 sound/soc/pxa/poodle.c | 1 6 files changed, 57 insertions(+), 56 deletions(-) --- linux-2.6.27-rc4.orig/include/linux/i2c-id.h 2008-08-27 22:27:01.000000000 +0200 +++ linux-2.6.27-rc4/include/linux/i2c-id.h 2008-08-27 22:38:47.000000000 +0200 @@ -64,7 +64,6 @@ #define I2C_DRIVERID_SAA717X 80 /* saa717x video encoder */ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ -#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ --- linux-2.6.27-rc4.orig/sound/soc/at91/eti_b1_wm8731.c 2008-08-27 22:27:01.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/at91/eti_b1_wm8731.c 2008-08-27 22:38:47.000000000 +0200 @@ -243,6 +243,7 @@ static struct snd_soc_machine snd_soc_ma }; static struct wm8731_setup_data eti_b1_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1a, }; --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8731.c 2008-08-27 22:27:01.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8731.c 2008-08-27 22:45:15.000000000 +0200 @@ -570,86 +570,86 @@ static struct snd_soc_device *wm8731_soc * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8731_i2c_driver; -static struct i2c_client client_template; - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ - -static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8731_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8731_socdev; - struct wm8731_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8731_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8731\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8731_i2c_detach(struct i2c_client *client) +static int wm8731_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8731_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8731_codec_probe); -} +static const struct i2c_device_id wm8731_i2c_id[] = { + { "wm8731", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "WM8731 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8731, - .attach_adapter = wm8731_i2c_attach, - .detach_client = wm8731_i2c_detach, - .command = NULL, + .probe = wm8731_i2c_probe, + .remove = wm8731_i2c_remove, + .id_table = wm8731_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8731", - .driver = &wm8731_i2c_driver, -}; +static int wm8731_add_i2c_device(struct platform_device *pdev, + const struct wm8731_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8731_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8731_i2c_driver); + return -ENODEV; +} #endif static int wm8731_probe(struct platform_device *pdev) @@ -682,11 +682,8 @@ static int wm8731_probe(struct platform_ wm8731_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8731_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8731_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -711,6 +708,7 @@ static int wm8731_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8731_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8731.h 2008-08-27 22:27:01.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8731.h 2008-08-27 22:38:47.000000000 +0200 @@ -35,6 +35,7 @@ #define WM8731_DAI 0 struct wm8731_setup_data { + int i2c_bus; unsigned short i2c_address; }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/corgi.c 2008-08-27 22:27:01.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/corgi.c 2008-08-27 22:38:47.000000000 +0200 @@ -330,6 +330,7 @@ static struct snd_soc_machine snd_soc_ma /* corgi audio private data */ static struct wm8731_setup_data corgi_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/poodle.c 2008-08-27 22:27:01.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/poodle.c 2008-08-27 22:38:47.000000000 +0200 @@ -284,6 +284,7 @@ static struct snd_soc_machine snd_soc_ma /* poodle audio private data */ static struct wm8731_setup_data poodle_wm8731_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; -- Jean Delvare

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[alsa-devel] [PATCH] ASoC: Convert wm8750 to a new-style i2c driver (v2, testers wanted)
by Jean Delvare 27 Aug '08

27 Aug '08

Convert the wm8750 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This second version fixes a bug reported by Manuel Lauss where codec->control_data was used before being initialized. Warning: this assumes that the spitz has its codec chip on I2C bus 0, which may or may not be the case. Someone with this machine please test and report. include/linux/i2c-id.h | 1 sound/soc/codecs/wm8750.c | 105 ++++++++++++++++++++++----------------------- sound/soc/codecs/wm8750.h | 1 sound/soc/pxa/spitz.c | 1 4 files changed, 55 insertions(+), 53 deletions(-) --- linux-2.6.27-rc4.orig/include/linux/i2c-id.h 2008-08-26 19:56:18.000000000 +0200 +++ linux-2.6.27-rc4/include/linux/i2c-id.h 2008-08-27 22:46:24.000000000 +0200 @@ -65,7 +65,6 @@ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ #define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ -#define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8750.c 2008-08-26 19:56:18.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8750.c 2008-08-27 22:45:24.000000000 +0200 @@ -846,83 +846,86 @@ static struct snd_soc_device *wm8750_soc * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8750_i2c_driver; -static struct i2c_client client_template; - -static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8750_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8750_socdev; - struct wm8750_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8750_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8750\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8750_i2c_detach(struct i2c_client *client) +static int wm8750_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8750_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8750_codec_probe); -} +static const struct i2c_device_id wm8750_i2c_id[] = { + { "wm8750", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8750_i2c_driver = { .driver = { .name = "WM8750 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8750, - .attach_adapter = wm8750_i2c_attach, - .detach_client = wm8750_i2c_detach, - .command = NULL, + .probe = wm8750_i2c_probe, + .remove = wm8750_i2c_remove, + .id_table = wm8750_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8750", - .driver = &wm8750_i2c_driver, -}; +static int wm8750_add_i2c_device(struct platform_device *pdev, + const struct wm8750_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8750", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8750_i2c_driver); + return -ENODEV; +} #endif static int wm8750_probe(struct platform_device *pdev) @@ -954,11 +957,8 @@ static int wm8750_probe(struct platform_ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8750_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1002,6 +1002,7 @@ static int wm8750_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8750.h 2008-08-26 19:56:18.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8750.h 2008-08-26 21:33:59.000000000 +0200 @@ -58,6 +58,7 @@ #define WM8750_SYSCLK 0 struct wm8750_setup_data { + int i2c_bus; unsigned short i2c_address; }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/spitz.c 2008-08-26 19:56:18.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/spitz.c 2008-08-26 21:33:59.000000000 +0200 @@ -337,6 +337,7 @@ static struct snd_soc_machine snd_soc_ma /* spitz audio private data */ static struct wm8750_setup_data spitz_wm8750_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; -- Jean Delvare

1 0

Re: [alsa-devel] Swissonic TS22PCI / Teratec Phase 22 / driver problems with ice1724
by Takashi Iwai 27 Aug '08

27 Aug '08

At Thu, 14 Aug 2008 16:07:13 +0200, Manuel Schneider wrote: > > Takashi Iwai schrieb, Am 14.08.2008 16:03: > > > At Thu, 14 Aug 2008 15:59:06 +0200, > > Manuel Schneider wrote: > >> Takashi Iwai schrieb, Am 14.08.2008 15:47: > >> > >>> Pass model=phase22 module option to snd-ice1724 driver. Then it > >>> handles the card as if it's phase22. > >> Thanks for this hin. Unfortunately it changes nothing - still kernel > >> error message, still no usable sound device. > > > > Are you sure? With the proper model name, at least you won't get > > EEPROM version error any more. > > > yes, pretty sure. I just reverted my patch and recompiled the driver to > check it out: > > ice1712: cannot initialize pro ac97, skipped > ice1724: No matching model found for ID 0x3b157b11 This means that you didn't pass the model option properly. > ice1724: Invalid EEPROM version 0 > AC'97 0 does not respond - RESET > AC'97 0 access is not valid [0xffffffff], removing mixer. > ice1712: cannot initialize pro ac97, skipped > > > >> I also tried the following kernel patch, which gives me an accessible > >> sound device and usable mixer (and no kernel error), but recording does > >> not work (just noise). > > > > Likely an incompatible hardware design for the recording... > > I guess it's another firmware... The board has no firmware to load. > I can't test playback as I only have > remote access to the server and it is only used for recording > (streaming) but I assume it does also not work. > I admit that this patch was just a hack to try it out. First you should check whether the exactly same codec chip, AK4524, on it. Takashi

2 5

Re: [alsa-devel] [PATCH RESEND] sound: make OSS sound core optional
by Takashi Iwai 27 Aug '08

27 Aug '08

At Wed, 27 Aug 2008 11:57:06 +0200, Tejun Heo wrote: > > Takashi Iwai wrote: > >> SOUND_OSS_CORE should be turned on if ALSA OSS emul is turned on or > >> arch-um hostsound is turned on, both of which can be enabled regardless > >> of SOUND_PRIME, no? > > > > Yes, sorry, I didn't cite clearly. I meant only the last line. > > sound/oss/dmasound/Kconfig is included and evaluated outside the block > > "if SOUND_PRIME". Thus, CONFIG_SOUND_OSS_CORE won't be enabled for > > drivers in sound/oss/dmasound. > > Ah.. okay, dmasound also requires SOUND_OSS_CORE but not in > SOUND_PRIME. I see. > > > My suggestion is to move that line source sound/oss/dmasound/Kconfig > > into the "if SOUND_PRIME" block first, then apply your patch. > > That would pull in sound_firmware. Most likely no big deal but > still... Ouch, another pain. sound_firmware.c should be removed and the relevant code should use request_firmware()... > How about just adding select SOUND_OSS_CORE to DMASOUND for > now? Looks good to me. Acked-by: Takashi Iwai <tiwai(a)suse.de> If no objection comes up, I'll merge it to sound tree later. thanks, Takashi > Thanks. > > Subject: [PATCH] sound: make OSS sound core optional > > sound/sound_core.c implements soundcore.ko and contains two parts - > sound_class which is shared by both ALSA and OSS and device > redirection support for OSS. It's always compiled when any sound > support is enabled although it's necessary only when OSS (the actual > one or emulation) is enabled. This is slightly wasteful and as device > redirection always registers character device region for major 14, it > prevents alternative implementation. > > This patch introduces a new config SOUND_OSS_CORE which is selected > iff OSS support is actually necessary and build the OSS core part > conditionally. > > If OSS is disabled, soundcore merely contains sound_class but leaving > it that way seems to be the simplest approach as otherwise sound_class > should be in ALSA core file if OSS is disabled but should be in > soundcore if OSS is enabled. Also, there's also the user confusion > factor. > > Signed-off-by: Tejun Heo <tj(a)kernel.org> > --- > arch/um/Kconfig.char | 4 ++ > sound/Kconfig | 5 ++ > sound/core/Kconfig | 1 > sound/oss/dmasound/Kconfig | 1 > sound/sound_core.c | 79 ++++++++++++++++++++++++++++++++++----------- > 5 files changed, 71 insertions(+), 19 deletions(-) > > Index: work/sound/Kconfig > =================================================================== > --- work.orig/sound/Kconfig > +++ work/sound/Kconfig > @@ -28,6 +28,10 @@ menuconfig SOUND > > if SOUND > > +config SOUND_OSS_CORE > + bool > + default n > + > source "sound/oss/dmasound/Kconfig" > > if !M68K > @@ -80,6 +84,7 @@ endif # SND > > menuconfig SOUND_PRIME > tristate "Open Sound System (DEPRECATED)" > + select SOUND_OSS_CORE > help > Say 'Y' or 'M' to enable Open Sound System drivers. > > Index: work/sound/core/Kconfig > =================================================================== > --- work.orig/sound/core/Kconfig > +++ work/sound/core/Kconfig > @@ -38,6 +38,7 @@ config SND_SEQ_DUMMY > will be called snd-seq-dummy. > > config SND_OSSEMUL > + select SOUND_OSS_CORE > bool > > config SND_MIXER_OSS > Index: work/sound/sound_core.c > =================================================================== > --- work.orig/sound/sound_core.c > +++ work/sound/sound_core.c > @@ -1,5 +1,61 @@ > /* > - * Sound core handling. Breaks out sound functions to submodules > + * Sound core. This file is composed of two parts. sound_class > + * which is common to both OSS and ALSA and OSS sound core which > + * is used OSS or emulation of it. > + */ > + > +/* > + * First, the common part. > + */ > +#include <linux/module.h> > +#include <linux/device.h> > +#include <linux/err.h> > + > +#ifdef CONFIG_SOUND_OSS_CORE > +static int __init init_oss_soundcore(void); > +static void __exit cleanup_oss_soundcore(void); > +#else > +static inline int init_oss_soundcore(void) { return 0; } > +static inline void cleanup_oss_soundcore(void) { } > +#endif > + > +struct class *sound_class; > +EXPORT_SYMBOL(sound_class); > + > +MODULE_DESCRIPTION("Core sound module"); > +MODULE_AUTHOR("Alan Cox"); > +MODULE_LICENSE("GPL"); > + > +static int __init init_soundcore(void) > +{ > + int rc; > + > + rc = init_oss_soundcore(); > + if (rc) > + return rc; > + > + sound_class = class_create(THIS_MODULE, "sound"); > + if (IS_ERR(sound_class)) { > + cleanup_oss_soundcore(); > + return PTR_ERR(sound_class); > + } > + > + return 0; > +} > + > +static void __exit cleanup_soundcore(void) > +{ > + cleanup_oss_soundcore(); > + class_destroy(sound_class); > +} > + > +module_init(init_soundcore); > +module_exit(cleanup_soundcore); > + > + > +#ifdef CONFIG_SOUND_OSS_CORE > +/* > + * OSS sound core handling. Breaks out sound functions to submodules > * > * Author: Alan Cox <alan.cox(a)linux.org> > * > @@ -34,21 +90,17 @@ > * locking at some point in 2.3.x. > */ > > -#include <linux/module.h> > #include <linux/init.h> > #include <linux/slab.h> > #include <linux/smp_lock.h> > #include <linux/types.h> > #include <linux/kernel.h> > -#include <linux/fs.h> > #include <linux/sound.h> > #include <linux/major.h> > #include <linux/kmod.h> > -#include <linux/device.h> > > #define SOUND_STEP 16 > > - > struct sound_unit > { > int unit_minor; > @@ -64,9 +116,6 @@ extern int msnd_classic_init(void); > extern int msnd_pinnacle_init(void); > #endif > > -struct class *sound_class; > -EXPORT_SYMBOL(sound_class); > - > /* > * Low level list operator. Scan the ordered list, find a hole and > * join into it. Called with the lock asserted > @@ -523,31 +572,23 @@ int soundcore_open(struct inode *inode, > return -ENODEV; > } > > -MODULE_DESCRIPTION("Core sound module"); > -MODULE_AUTHOR("Alan Cox"); > -MODULE_LICENSE("GPL"); > MODULE_ALIAS_CHARDEV_MAJOR(SOUND_MAJOR); > > -static void __exit cleanup_soundcore(void) > +static void __exit cleanup_oss_soundcore(void) > { > /* We have nothing to really do here - we know the lists must be > empty */ > unregister_chrdev(SOUND_MAJOR, "sound"); > - class_destroy(sound_class); > } > > -static int __init init_soundcore(void) > +static int __init init_oss_soundcore(void) > { > if (register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops)==-1) { > printk(KERN_ERR "soundcore: sound device already in use.\n"); > return -EBUSY; > } > - sound_class = class_create(THIS_MODULE, "sound"); > - if (IS_ERR(sound_class)) > - return PTR_ERR(sound_class); > > return 0; > } > > -module_init(init_soundcore); > -module_exit(cleanup_soundcore); > +#endif /* CONFIG_SOUND_OSS_CORE */ > Index: work/arch/um/Kconfig.char > =================================================================== > --- work.orig/arch/um/Kconfig.char > +++ work/arch/um/Kconfig.char > @@ -203,6 +203,10 @@ config SOUND > tristate > default UML_SOUND > > +config SOUND_OSS_CORE > + bool > + default UML_SOUND > + > config HOSTAUDIO > tristate > default UML_SOUND > Index: work/sound/oss/dmasound/Kconfig > =================================================================== > --- work.orig/sound/oss/dmasound/Kconfig > +++ work/sound/oss/dmasound/Kconfig > @@ -42,3 +42,4 @@ config DMASOUND_Q40 > > config DMASOUND > tristate > + select SOUND_OSS_CORE >

1 0

Re: [alsa-devel] [PATCH RESEND] sound: make OSS sound core optional
by Takashi Iwai 27 Aug '08

27 Aug '08

At Wed, 27 Aug 2008 11:37:44 +0200, Tejun Heo wrote: > > Hello, > > Takashi Iwai wrote: > >> +config SOUND_OSS_CORE > >> + bool > >> + default n > >> + > >> source "sound/oss/dmasound/Kconfig" > > > > This seems missing from the scope. > > But, I think this should be rather in the block "if SOUND_PRIME" below. > > SOUND_OSS_CORE should be turned on if ALSA OSS emul is turned on or > arch-um hostsound is turned on, both of which can be enabled regardless > of SOUND_PRIME, no? Yes, sorry, I didn't cite clearly. I meant only the last line. sound/oss/dmasound/Kconfig is included and evaluated outside the block "if SOUND_PRIME". Thus, CONFIG_SOUND_OSS_CORE won't be enabled for drivers in sound/oss/dmasound. My suggestion is to move that line source sound/oss/dmasound/Kconfig into the "if SOUND_PRIME" block first, then apply your patch. Anyway, there are some other minor issues in sound/Kconfig, for example, why "if !M68K" is there. We can clean up a bit more. thanks, Takashi

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Re: [alsa-devel] [PATCH RESEND] sound: make OSS sound core optional
by Takashi Iwai 27 Aug '08

27 Aug '08

At Wed, 27 Aug 2008 10:19:35 +0200, Tejun Heo wrote: > > sound/sound_core.c implements soundcore.ko contains two parts - > sound_class which is shared by both ALSA and OSS and device > redirection support for OSS. It's always compiled when any sound > support is enabled although it's necessary only when OSS (the actual > one or emulation) is enabled. This is slightly wasteful and as device > redirection always registers character device region for major 14, it > prevents alternative implementation. > > This patch introduces a new config SOUND_OSS_CORE which is selected > iff OSS support is actually necessary and build the OSS core part > conditionally. > > If OSS is disabled, soundcore merely contains sound_class but leaving > it that way seems to be the simplest approach as otherwise sound_class > should be in ALSA core file if OSS is disabled but should be in > soundcore if OSS is enabled and there's also the user confusion > factor. > > Signed-off-by: Tejun Heo <tj(a)kernel.org> Thanks for the patch. I like the idea. > +config SOUND_OSS_CORE > + bool > + default n > + > source "sound/oss/dmasound/Kconfig" This seems missing from the scope. But, I think this should be rather in the block "if SOUND_PRIME" below. > --- work.orig/sound/sound_core.c > +++ work/sound/sound_core.c (snip) > +MODULE_ALIAS_CHARDEV_MAJOR(SOUND_MAJOR); This alias should be in #ifdef CONFIG_SOUND_OSS_CORE. The module doesn't provide that major without CONFIG_SOUND_OSS_CORE. Takashi

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Re: [alsa-devel] [ALSA PATCH] alsa-git merge request
by Takashi Iwai 27 Aug '08

27 Aug '08

At Tue, 26 Aug 2008 23:10:55 +0200, Guillaume Chazarain wrote: > > On Tue, Aug 26, 2008 at 7:59 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > Are you sure both are the same kernel but just one commit reverted? > > I retested, yes. Thanks for confirmation! > With or without the revert, model=auto disables sound completely, and > I get this in dmesg: > > HDA Intel 0000:00:1b.0: PCI INT A disabled > HDA Intel 0000:00:1b.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20 > HDA Intel 0000:00:1b.0: setting latency timer to 64 > ALSA /home/g/linux-2.6/sound/pci/hda/hda_codec.c:2325: hda_codec: > model 'auto' is selected > ALSA /home/g/linux-2.6/sound/pci/hda/hda_codec.c:3021: autoconfig: > line_outs=1 (0x14/0x0/0x0/0x0/0x0) > ALSA /home/g/linux-2.6/sound/pci/hda/hda_codec.c:3025: > speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > ALSA /home/g/linux-2.6/sound/pci/hda/hda_codec.c:3029: hp_outs=1 > (0x15/0x0/0x0/0x0/0x0) > ALSA /home/g/linux-2.6/sound/pci/hda/hda_codec.c:3030: mono: mono_out=0x0 > ALSA /home/g/linux-2.6/sound/pci/hda/hda_codec.c:3038: inputs: > mic=0x18, fmic=0x0, line=0x1a, fline=0x0, cd=0x0, aux=0x0 > > I played with model=intel-mac-auto and model=intel-mac-v3 but it had > no effect, i.e. same as without the model= parameter. > > So, without model=auto the problem is that I get the output on both > the internal speakers and the headphones, and this is fixed by > reverting the commit. Then this is really a PCI SSID conflict, and reverting the patch is the right fix (and possibly addition of check of codec SID). I fixed it now on my tree, and will include it in the next pull request. thanks, Takashi

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[alsa-devel] [PATCH] ASoC: Convert wm8750 to a new-style i2c driver (testers wanted)
by Jean Delvare 26 Aug '08

26 Aug '08

Convert the wm8750 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Warning: this assumes that the spitz has its codec chip on I2C bus 0, which may or may not be the case. Someone with this machine please test and report. Note that you can safely ignore patch failures in i2c-id.h. include/linux/i2c-id.h | 1 sound/soc/codecs/wm8750.c | 108 ++++++++++++++++++++++----------------------- sound/soc/codecs/wm8750.h | 1 sound/soc/pxa/spitz.c | 1 4 files changed, 57 insertions(+), 54 deletions(-) --- linux-2.6.27-rc4.orig/include/linux/i2c-id.h 2008-08-26 16:40:44.000000000 +0200 +++ linux-2.6.27-rc4/include/linux/i2c-id.h 2008-08-26 16:41:12.000000000 +0200 @@ -65,7 +65,6 @@ #define I2C_DRIVERID_DS1672 81 /* Dallas/Maxim DS1672 RTC */ #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ #define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ -#define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8750.c 2008-08-26 15:39:54.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8750.c 2008-08-26 16:01:06.000000000 +0200 @@ -846,83 +846,87 @@ static struct snd_soc_device *wm8750_soc * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8750_i2c_driver; -static struct i2c_client client_template; - -static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8750_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8750_socdev; - struct wm8750_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) - return -ENOMEM; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = i2c_attach_client(i2c); - if (ret < 0) { - pr_err("failed to attach codec at addr %x\n", addr); - goto err; - } ret = wm8750_init(socdev); - if (ret < 0) { + if (ret < 0) pr_err("failed to initialise WM8750\n"); - goto err; - } - return ret; -err: - kfree(i2c); return ret; } -static int wm8750_i2c_detach(struct i2c_client *client) +static int wm8750_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8750_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8750_codec_probe); -} +static const struct i2c_device_id wm8750_i2c_id[] = { + { "wm8750", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8750_i2c_driver = { .driver = { .name = "WM8750 I2C Codec", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_WM8750, - .attach_adapter = wm8750_i2c_attach, - .detach_client = wm8750_i2c_detach, - .command = NULL, + .probe = wm8750_i2c_probe, + .remove = wm8750_i2c_remove, + .id_table = wm8750_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8750", - .driver = &wm8750_i2c_driver, -}; +static int wm8750_add_i2c_device(struct platform_device *pdev, + const struct wm8750_setup_data *setup) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8750", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + socdev->codec->control_data = client; + + return 0; + +err_driver: + i2c_del_driver(&wm8750_i2c_driver); + return -ENODEV; +} #endif static int wm8750_probe(struct platform_device *pdev) @@ -954,11 +958,8 @@ static int wm8750_probe(struct platform_ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8750_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1002,6 +1003,7 @@ static int wm8750_remove(struct platform snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif kfree(codec->private_data); --- linux-2.6.27-rc4.orig/sound/soc/codecs/wm8750.h 2008-08-26 15:39:54.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/codecs/wm8750.h 2008-08-26 15:48:15.000000000 +0200 @@ -58,6 +58,7 @@ #define WM8750_SYSCLK 0 struct wm8750_setup_data { + int i2c_bus; unsigned short i2c_address; }; --- linux-2.6.27-rc4.orig/sound/soc/pxa/spitz.c 2008-08-26 15:39:54.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/pxa/spitz.c 2008-08-26 15:48:15.000000000 +0200 @@ -337,6 +337,7 @@ static struct snd_soc_machine snd_soc_ma /* spitz audio private data */ static struct wm8750_setup_data spitz_wm8750_setup = { + .i2c_bus = 0, .i2c_address = 0x1b, }; -- Jean Delvare

2 2

[alsa-devel] alsa regression: snd_bt87x mute (not mixer related)
by Jorge Peixoto de Morais Neto 26 Aug '08

26 Aug '08

Hi. Under kernel 2.6.23, I could get sound from the alsa device created by the module snd_bt87x. In kernel 2.6.24, there is no sound. And yes, I have checked in alsamixer that the device is not muted. Also, in kernel 2.6.23 the device would only appear if I modprobed the module with the option load_all=1. In kernel 2.6.24, the option load_all=1 is unnecessary for the device to appear. This is probably because I have in 2007 sent an email to this list (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-August/002617.html) with information about my board, and it was responded by Takashi. I have tried the latest linux-2.6.27-rc4 hoping the problem could have been fixed. No improvement. Could you please look at this? It is an important regression. I have included extensive information about my system (under both linux-2.6.23, where my card works, and linux-2.6.27-rc4) in the tarball attached. If you need any other information please ask me. Regards, Jorge Peixoto de Morais Neto

1 0

[alsa-devel] ice1724 - TerraTec PHASE 22 no MIDI ports get created
by Vedran Miletić 26 Aug '08

26 Aug '08

Hi guys, There was "MIDI on ice1724" thread not-so-recently, and it had a patch attached. Changelog says that the patch was added to ALSA in 1.0.17. So I took out my dusty TerraTec PHASE 22 out of the closet (and removed the dust, of course :-) ), but I can't get MIDI to work there. Regardless of ALSA version (1.0.15 - 1.0.17, even 1.0.18rc1), I get no MIDI ports at all: vedran@kalopsia:~$ amidi -l Dir Device Name vedran@kalopsia:~$ Why don't MIDI ports get created? I tried looking at the source, but I couldn't figure out how this "creation of midi ports" happens (or doesn't happen, as in this case). -- Vedran Miletić

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[alsa-devel] OSS emulatior can not create enough device files for me
by Taipei Techwell 26 Aug '08

26 Aug '08

Hi, I am working on an ALSA audio driver for my device which is very similar to BT878. It will have at least 8 audio devices in a single card. Now, the audio driver is working correctly. I can use aplay to play the audio file which is created by arecord. To identify audio devices, I am using hw:1 ~ hw:8 dvice description for my audio devices, because hw:0 is the sound card. In order to make the old application work, I know that ALSA emulates all my devices to have OSS nodes respectively. I guess that It should have /dev/dsp to /dev/dsp8, and /dev/dsp is for sound card, and /dev/dsp1 to /dev/dsp8 are for my audio devices. However, I can not find /dev/dsp8 in my system. I guess ALSA can only emulate 8 audio device totally. That the reason I can not see the 8th audio device. Am I right? How can I create the /dev/dsp8 manually? By the way, if someting is wrong above, please correct me. Thank you.

2 1

[alsa-devel] [PATCH] ASoC: WM8903 I2C cleanups
by Mark Brown 26 Aug '08

26 Aug '08

Fix a few issues identified by Jean Delvare: - Add missing newline to error display. - Return an error when we fail to locate the I2C adaptor. - Remove the static wm8903_i2c_device Jean suggested, it's redundant. Cc: Jean Delvare <khali(a)linux-fr.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 73dfd5b..a3f54ec 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1706,8 +1706,6 @@ static struct i2c_driver wm8903_i2c_driver = { .id_table = wm8903_i2c_id, }; -static struct i2c_client *wm8903_i2c_device; - static int wm8903_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1716,6 +1714,7 @@ static int wm8903_probe(struct platform_device *pdev) struct wm8903_priv *wm8903; struct i2c_board_info board_info; struct i2c_adapter *adapter; + struct i2c_client *i2c_client; int ret = 0; setup = socdev->codec_data; @@ -1746,7 +1745,7 @@ static int wm8903_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = i2c_add_driver(&wm8903_i2c_driver); if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver"); + dev_err(&pdev->dev, "can't add i2c driver\n"); goto err_priv; } else { memset(&board_info, 0, sizeof(board_info)); @@ -1757,12 +1756,13 @@ static int wm8903_probe(struct platform_device *pdev) if (!adapter) { dev_err(&pdev->dev, "Can't get I2C bus %d\n", setup->i2c_bus); + ret = -ENODEV; goto err_adapter; } - wm8903_i2c_device = i2c_new_device(adapter, &board_info); + i2c_client = i2c_new_device(adapter, &board_info); i2c_put_adapter(adapter); - if (wm8903_i2c_device == NULL) { + if (i2c_client == NULL) { dev_err(&pdev->dev, "I2C driver registration failed\n"); ret = -ENODEV; @@ -1792,7 +1792,7 @@ static int wm8903_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - i2c_unregister_device(wm8903_i2c_device); + i2c_unregister_device(socdev->codec->control_data); i2c_del_driver(&wm8903_i2c_driver); kfree(codec->private_data); kfree(codec); -- 1.5.6.5

3 2

[alsa-devel] [PATCH] ASoC: Add missing new-line at end of strings
by Jean Delvare 26 Aug '08

26 Aug '08

Two warning messages lack their trailing new-line, add it. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- sound/soc/at91/at91-ssc.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- linux-2.6.27-rc4.orig/sound/soc/at91/at91-ssc.c 2008-08-13 09:51:02.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/at91/at91-ssc.c 2008-08-26 15:45:11.000000000 +0200 @@ -408,7 +408,7 @@ static int at91_ssc_hw_params(struct snd dma_params->pdc_xfer_size = 4; break; default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format"); + printk(KERN_WARNING "at91-ssc: unsupported PCM format\n"); return -EINVAL; } --- linux-2.6.27-rc4.orig/sound/soc/davinci/davinci-i2s.c 2008-08-05 18:18:18.000000000 +0200 +++ linux-2.6.27-rc4/sound/soc/davinci/davinci-i2s.c 2008-08-26 15:45:04.000000000 +0200 @@ -256,7 +256,7 @@ static int davinci_i2s_hw_params(struct mcbsp_word_length = DAVINCI_MCBSP_WORD_32; break; default: - printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } -- Jean Delvare

2 1

Re: [alsa-devel] [PATCH] ASoC: Add WM8903 CODEC driver
by Jean Delvare 26 Aug '08

26 Aug '08

Hi Mark, On Tue, 26 Aug 2008 13:05:27 +0100, Mark Brown wrote: > The WM8903 is a high performance ultra-low power stereo CODEC optimised > for portable audio applications. Features include: > > * 5mW power consumption for DAC to headphone playback > * Stereo DAC SNR 96dB typical, THD -86dB typical > * Stereo ADC SNR 93dB typical, THD -80dB typical > * Up to 3 single ended inputs per stereo channel > * Up to 2 pseudo differential inputs per stereo channel > * Up to 1 fully differential mic input per stereo channel > * Digital Dynamic Range Controller (compressor/limiter) > * Digital sidetone mixing > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/wm8903.c | 1813 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/wm8903.h | 1463 ++++++++++++++++++++++++++++++++++++ > 4 files changed, 3282 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/wm8903.c > create mode 100644 sound/soc/codecs/wm8903.h Here are minor issues I spotted in the i2c part of your driver: > (...) > +static int wm8903_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) > +{ > + struct snd_soc_device *socdev = wm8903_socdev; > + struct snd_soc_codec *codec = socdev->codec; > + int ret; > + > + i2c_set_clientdata(i2c, codec); > + codec->control_data = i2c; > + > + ret = wm8903_init(socdev); > + if (ret < 0) > + dev_err(&i2c->dev, "Device initialisation failed\n"); > + > + return ret; > +} > + > +static int wm8903_i2c_remove(struct i2c_client *client) > +{ > + struct snd_soc_codec *codec = i2c_get_clientdata(client); > + kfree(codec->reg_cache); > + return 0; > +} > + > +/* i2c codec control layer */ > +static const struct i2c_device_id wm8903_i2c_id[] = { > + { "wm8903", 0 }, > + { } > +}; > +MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); > + > +static struct i2c_driver wm8903_i2c_driver = { > + .driver = { > + .name = "WM8903", > + .owner = THIS_MODULE, > + }, > + .probe = wm8903_i2c_probe, > + .remove = wm8903_i2c_remove, > + .id_table = wm8903_i2c_id, > +}; > + > +static struct i2c_client *wm8903_i2c_device; I know that I am the one who came up with this idea, but on second thought, wm8903_i2c_device will always be equal to wm8903_socdev->codec->control_data, so we can easily do without it. > + > +static int wm8903_probe(struct platform_device *pdev) > +{ > + struct snd_soc_device *socdev = platform_get_drvdata(pdev); > + struct wm8903_setup_data *setup; > + struct snd_soc_codec *codec; > + struct wm8903_priv *wm8903; > + struct i2c_board_info board_info; > + struct i2c_adapter *adapter; > + int ret = 0; > + > + setup = socdev->codec_data; > + > + if (!setup->i2c_address) { > + dev_err(&pdev->dev, "No codec address provided\n"); > + return -ENODEV; > + } > + > + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); > + if (codec == NULL) > + return -ENOMEM; > + > + wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); > + if (wm8903 == NULL) { > + ret = -ENOMEM; > + goto err_codec; > + } > + > + codec->private_data = wm8903; > + socdev->codec = codec; > + mutex_init(&codec->mutex); > + INIT_LIST_HEAD(&codec->dapm_widgets); > + INIT_LIST_HEAD(&codec->dapm_paths); > + > + wm8903_socdev = socdev; > + > + codec->hw_write = (hw_write_t)i2c_master_send; > + ret = i2c_add_driver(&wm8903_i2c_driver); > + if (ret != 0) { > + dev_err(&pdev->dev, "can't add i2c driver"); Missing \n at end of string (looks like many other drivers have this problem - I found 11 other cases under sound/soc, I can send a patch fixing these if you want.) > + goto err_priv; > + } else { > + memset(&board_info, 0, sizeof(board_info)); > + strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE); > + board_info.addr = setup->i2c_address; > + > + adapter = i2c_get_adapter(setup->i2c_bus); > + if (!adapter) { > + dev_err(&pdev->dev, "Can't get I2C bus %d\n", > + setup->i2c_bus); > + goto err_adapter; You are jumping to the error path but ret has value 0. This means that the function returns success when you really mean failure, and trouble is likely to arise after that. > + } > + > + wm8903_i2c_device = i2c_new_device(adapter, &board_info); > + i2c_put_adapter(adapter); > + if (wm8903_i2c_device == NULL) { > + dev_err(&pdev->dev, > + "I2C driver registration failed\n"); > + ret = -ENODEV; > + goto err_adapter; > + } > + } > + > + return ret; > + > +err_adapter: > + i2c_del_driver(&wm8903_i2c_driver); > +err_priv: > + kfree(codec->private_data); > +err_codec: > + kfree(codec); > + return ret; > +} -- Jean Delvare

3 3

[alsa-devel] [PATCH] ASoC: Fix error paths in N810 machine driver init and release clocks at exit
by Jarkko Nikula 26 Aug '08

26 Aug '08

Thanks to Felipe Balbi <felipe.balbi(a)nokia.com> by noticing that if clk_get to sys_clkout2_src fails, then n810_snd_device is never released. Add also sys_clkout2_src release into error path, error code return and release the clocks at exit. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 18 +++++++++++++----- 1 files changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 7694621..87d0ed0 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -329,12 +329,14 @@ static int __init n810_soc_init(void) sys_clkout2_src = clk_get(dev, "sys_clkout2_src"); if (IS_ERR(sys_clkout2_src)) { dev_err(dev, "Could not get sys_clkout2_src clock\n"); - return -ENODEV; + err = PTR_ERR(sys_clkout2_src); + goto err2; } sys_clkout2 = clk_get(dev, "sys_clkout2"); if (IS_ERR(sys_clkout2)) { dev_err(dev, "Could not get sys_clkout2\n"); - goto err1; + err = PTR_ERR(sys_clkout2); + goto err3; } /* * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use @@ -343,7 +345,8 @@ static int __init n810_soc_init(void) func96m_clk = clk_get(dev, "func_96m_ck"); if (IS_ERR(func96m_clk)) { dev_err(dev, "Could not get func 96M clock\n"); - goto err2; + err = PTR_ERR(func96m_clk); + goto err4; } clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); @@ -356,20 +359,25 @@ static int __init n810_soc_init(void) gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; -err2: +err4: clk_put(sys_clkout2); +err3: + clk_put(sys_clkout2_src); +err2: platform_device_del(n810_snd_device); err1: platform_device_put(n810_snd_device); return err; - } static void __exit n810_soc_exit(void) { gpio_free(N810_SPEAKER_AMP_GPIO); gpio_free(N810_HEADSET_AMP_GPIO); + clk_put(sys_clkout2_src); + clk_put(sys_clkout2); + clk_put(func96m_clk); platform_device_unregister(n810_snd_device); } -- 1.5.6.3

3 2

[alsa-devel] [PATCH] ASoC: Fix double free and memory leak in many codec drivers
by Mark Brown 26 Aug '08

26 Aug '08

From: Jean Delvare <khali(a)linux-fr.org> Many SoC audio codec drivers have improper freeing of memory in error paths. * codec is allocated in the platform device probe function, but is not freed there in case of error. Instead it is freed in the i2c device probe function's error path. However the success or failure of both functions is not linked, so this could result in a double free (if the platform device is successfully probed, the i2c device probing fails and then the platform driver is unregistered.) * codec->private_data is allocated in many platform device probe functions but not freed in their error paths. This patch hopefully solves all these problems. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4535.c | 11 +++++++---- sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- sound/soc/codecs/uda1380.c | 9 +++++---- sound/soc/codecs/wm8510.c | 9 +++++---- sound/soc/codecs/wm8731.c | 11 +++++++---- sound/soc/codecs/wm8750.c | 10 ++++++---- sound/soc/codecs/wm8753.c | 11 +++++++---- sound/soc/codecs/wm8990.c | 11 +++++++---- 8 files changed, 51 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index b26003c..7da9f46 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -562,10 +562,9 @@ static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -583,7 +582,6 @@ static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -660,6 +658,11 @@ static int ak4535_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b1dce5f..5f9abb1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1199,10 +1199,9 @@ static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1221,7 +1220,6 @@ static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -1302,6 +1300,11 @@ static int aic3x_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a52d6d9..807318f 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -729,10 +729,9 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -750,7 +749,6 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -817,6 +815,9 @@ static int uda1380_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) + kfree(codec); return ret; } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 67325fd..3d998e6 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -693,10 +693,9 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -714,7 +713,6 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -782,6 +780,9 @@ static int wm8510_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) + kfree(codec); return ret; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 369d39c..9402fca 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -596,10 +596,9 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -617,7 +616,6 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -693,6 +691,11 @@ static int wm8731_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index c6a8edf..dd1f554 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -869,10 +869,9 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -890,7 +889,6 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -966,6 +964,10 @@ static int wm8750_probe(struct platform_device *pdev) /* Add other interfaces here */ #endif + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index dc7b18f..5761164 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1660,10 +1660,9 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (!i2c) { - kfree(codec); + if (!i2c) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1682,7 +1681,6 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -1759,6 +1757,11 @@ static int wm8753_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index e44153f..dd995ef 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1500,10 +1500,9 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1521,7 +1520,6 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -1595,6 +1593,11 @@ static int wm8990_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } -- 1.5.6.3

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Re: [alsa-devel] [Alsa-user] Sound card recording support @ 96kHz or above
by Vedran Miletić 26 Aug '08

26 Aug '08

Terrasoniq TS22 (newer revision of TerraTec PHASE 22) might do it for you. Almost anything Envy24-series based will offer what you want. Not sure how you would check that it actually works, though. 2008/8/26 Ben Gould <ben.gould(a)gmail.com>: > These cards don't seem to be in production any more. > > Any suggestions for more recent off-the-shelf hardware that will capture > a single stereo input at 96kHz (using ALSA)? > > -- ben > > Sergei Steshenko wrote: >> >> -----Original Message----- >> From: Ben Gould <ben.gould(a)gmail.com> >> To: alsa-user(a)lists.sourceforge.net >> Date: Thu, 21 Aug 2008 16:57:43 +0059 >> Subject: [Alsa-user] Sound card recording support @ 96kHz or above >> >>> Hi all, >>> >>> A project I'm working on using linux & alsa requires a higher than >>> normal sampling rate, not just up-sampling from 48kHz. >>> >>> I wonder whether anyone here can recommend a sound card that's capable >>> of recording a single stereo input at 96kHz sample rate (16 bits would >>> be sufficient). >>> >>> Obviously it needs to be properly supported by alsa... >>> >>> Thanks in advance >>> >>> -- ben >>> >> >> M-Audio Revolution 7.1, 5.1. >> >> Regards, >> Sergei. >> >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > -- Vedran Miletić

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[alsa-devel] Commit 2a3fdf2e7bbeae557b4a1ee64c071ae4c3c561e5 makes unable to compile ca106 module
by Vedran Miletić 26 Aug '08

26 Aug '08

Here's a proper patch: --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -249,11 +249,12 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , - /* Another MSI K8N Diamond MB, which has apprently a different SSID */ + /* MSI K8N Diamond PLUS MB */ { .serial = 0x10091102, .name = "MSI K8N Diamond MB", .gpio_type = 2, - .i2c_adc = 1 } , + .i2c_adc = 1, + .spi_dac = 2 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". The comma was missing after .spi_dac line. -- Vedran Miletić

2 1

Re: [alsa-devel] Alsa-driver compile fail on Ubuntu powerpc
by nguyen tritai 26 Aug '08

26 Aug '08

Takashi, Now I can build and install driver for ca0106 sb Audigy but I got another issue when play an example file. >> aplay -vv child.wav Playing WAVE '/root/child.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 22050 exact rate : 22050 (22050/1) msbits : 8 buffer_size : 3763 period_size : 470 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 470 period_event : 0 start_threshold : 3763 stop_threshold : 3763 silence_threshold: 0 silence_size : 0 boundary : 493223936 Slave: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 ######### + | 35%Segmentation fault Kernel dump message >> dmesg Unable to handle kernel paging request for data at address 0x00000000 Faulting instruction address: 0xc0033fd4 Oops: Kernel access of bad area, sig: 11 [#7] Canyonlands Modules linked in: snd_seq_midi snd_ca0106 snd_rawmidi snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_ac97_codec snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd_page_alloc snd ac97_bus NIP: c0033fd4 LR: c004ac94 CTR: e901a6c8 REGS: dfbf3d40 TRAP: 0300 Tainted: G D (2.6.26-00015-gc0b03c1-dirty) MSR: 00029000 <EE,ME> CR: 48022444 XER: 20000000 DEAR: 00000000, ESR: 00000000 TASK = dfa57000[15245] 'aplay' THREAD: dfbf2000 GPR00: c004ac94 dfbf3df0 dfa57000 00000000 c0b2c180 00000000 3f10c149 00000000 GPR08: 00000000 00000000 9e370001 c030b10c 00000149 10025478 1000bd14 bff7ab68 GPR16: c0320000 c030ae74 c0320000 00000000 00000000 00000000 00000000 de14c340 GPR24: 4814b000 00000000 dfbf2000 de00c2c0 c0b2c180 00000000 00000000 c0b2c180 NIP [c0033fd4] __wake_up_bit+0x10/0x40 LR [c004ac94] unlock_page+0x48/0x5c Call Trace: [dfbf3e10] [c004ac94] unlock_page+0x48/0x5c [dfbf3e20] [c005a2d8] __do_fault+0x254/0x618 [dfbf3e80] [c000faf0] do_page_fault+0x2d8/0x518 [dfbf3f40] [c000d134] handle_page_fault+0xc/0x80 Instruction dump: 7d6b4a14 800b0280 812b0278 20000020 7c630430 54631838 7c634a14 4e800020 7c0802a6 9421ffe0 90810008 90010024 <80030000> 90a1000c 7f801800 419e0014 ---[ end trace a0b9d3bfbd44991b ]--- I tried with another example file.wav but the problem is the same. The issue seems from accessing memory. I'm using AMCC 460EX board with distribution Ubuntu-7.04 Feel free to help me. Thanks, TaiNguyen. I tried with another wav On Mon, Aug 18, 2008 at 9:52 AM, nguyen tritai <nguyentritai(a)gmail.com>wrote: > Hi Takashi, > > Thank for your help. I checked in the config.log as you told me and see > that the issue is from the linux-header . > I'm now working on it. > > Thanks, > > TaiNguyen > > > On Thu, Aug 14, 2008 at 8:49 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> At Thu, 14 Aug 2008 16:25:24 +0700, >> nguyen tritai wrote: >> > >> > Hi all, >> > >> > I'm trying to bring up audio on powerpc AMCC 440EPX board with the >> soundcard >> > Sound Blaster Audigy CA0106. >> > The board is runing Ubuntu 7.04. >> > I configured: ./configure --with-cards=ca0106 --with-sequencer=yes >> > --with-kernel=/usr/src/linux >> > Then I make and get issue: >> > >> > make[2]: Leaving directory `/root/alsa-driver-1.0.17/usb' >> > make[2]: Entering directory `/root/alsa-driver-1.0.17/misc' >> > make[2]: Leaving directory `/root/alsa-driver-1.0.17/misc' >> > make[1]: Leaving directory `/root/alsa-driver-1.0.17' >> > make -C /usr/src/linux SUBDIRS=/root/alsa-driver-1.0.17 CPP="gcc -E" >> > CC="gcc" modules >> > make[1]: Entering directory `/usr/src/linux' >> > CC [M] /root/alsa-driver-1.0.17/acore/memory_wrapper.o >> > make[1]: Leaving directory `/usr/src/linux' >> > root@sequoia:~/alsa-driver-1.0.17# vim log >> > In file included from >> /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: >> > /root/alsa-driver-1.0.17/include/adriver.h:278:1: warning: "GFP_DMA32" >> > redefined >> > In file included from include/linux/slab.h:14, >> > from include/linux/percpu.h:5, >> > from include/linux/rcupdate.h:41, >> > from include/linux/pid.h:4, >> > from include/linux/sched.h:76, >> > from /root/alsa-driver-1.0.17/include/asm/elf.h:5, >> > from include/linux/elf.h:7, >> > from include/linux/module.h:14, >> > from /root/alsa-driver-1.0.17/include/adriver.h:26, >> > from >> /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: >> > include/linux/gfp.h:105:1: warning: this is the location of the previous >> > definition >> >> This means that configure script didn't detect the things correctly. >> Doesn't the snapshot version work, too? >> If no, check config.log about the relevant part checking DMA32. >> >> >> Takashi >> > >

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Re: [alsa-devel] Online code snapshots on ALSA server
by Jaroslav Kysela 26 Aug '08

26 Aug '08

On Mon, 25 Aug 2008, Sean McNamara wrote: > Glad to hear it, Jaroslav! That's much easier than checking out from GIT. > > If these are automatically updated, any idea how frequently the > packages will be regenerated? Is it after every commit, or only at > particular time intervals? You can grab the latest code immediately after commit to GIT repository (but generation of new package takes awhile as mentioned on page when it's not built on server already). The package generation is triggered from the web server by users requesting latest package immediately. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] Compilation of Alsa 1.0.17 on Suse10 crashes
by Takashi Iwai 26 Aug '08

26 Aug '08

At Fri, 22 Aug 2008 19:08:29 +0200 (CEST), Vaclav Peroutka wrote: > > Hello Mr. Takashi, > > I tried to compile alsa-driver-snapshot from the page you advised me. But the problem was the same. I am sending you configure.log and make.log files. Hopefully they will clarify the problem. > > I still have problem with mts64.c. OK, I see the problem, too. As a temporary workaround, you can change alsa-driver/kconfig-vers. This is a list of the least kernel version for some drivers, and mts64 is there, too. Change the version to 2.6.26 or such, so that it won't be built for your kernel. Meanwhile, I'll try to fix the problem in the upstream. thanks, Takashi > > Can you help me please ? > > Thank you in advance, > Vaclav > > > ------------ Původní zpráva ------------ > > Od: Takashi Iwai <tiwai(a)suse.de> > > Předmět: Re: [alsa-devel] Compilation of Alsa 1.0.17 on Suse10 crashes > > Datum: 22.8.2008 08:27:13 > > ---------------------------------------- > > At Thu, 21 Aug 2008 23:59:14 +0200 (CEST), > > Vaclav Peroutka wrote: > > > > > > Hello all, > > > > > > I downloaded alsa-driver-1.0.17 and tried to compile it. I started with > > "./configure --with-suse=auto --with-oss=yes --with-cards=all,mts64=n" and then > > "make". > > > > > > But I always get following: > > > ... > > > CC [M] /tmp/alsa-driver-1.0.17/drivers/mts64.o > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c: In function ‘snd_mts64_attach’: > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:896: error: implicit declaration of > > function ‘platform_device_alloc’ > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:896: warning: assignment makes pointer > > from integer without a cast > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:903: error: implicit declaration of > > function ‘platform_device_add’ > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:904: error: implicit declaration of > > function ‘platform_device_put’ > > > make[4]: *** [/tmp/alsa-driver-1.0.17/drivers/mts64.o] Error 1 > > > make[3]: *** [/tmp/alsa-driver-1.0.17/drivers] Error 2 > > > make[2]: *** [_module_/tmp/alsa-driver-1.0.17] Error 2 > > > make[1]: *** [modules] Error 2 > > > make[1]: Leaving directory `/usr/src/linux-2.6.13-15' > > > > > > My GCC: > > > > gcc --version > > > gcc (GCC) 4.0.2 20050901 (prerelease) (SUSE Linux) > > > > > > Is there something missing ? What should I do now ? > > > > Use the later version, at best, the daily snapshot tarball > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > > > Or, you can try alsa-driver-kmp on standard SUSE systems. > > http://download.opensuse.org/repositories/multimedia:/audio:/KMP/ > > > > Check openSUSE Wiki SDB:AudioTroubleshooting for details. > > > > > > Takashi > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > >

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Re: [alsa-devel] [ALSA PATCH] alsa-git merge request
by Takashi Iwai 26 Aug '08

26 Aug '08

At Tue, 26 Aug 2008 00:43:59 +0200, Guillaume Chazarain wrote: > > On Mon, Aug 25, 2008 at 11:36 PM, Guillaume Chazarain <guichaz(a)gmail.com> wrote: > > On Mon, Aug 25, 2008 at 7:50 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> I guess it's fixed as well by passing model=auto? > > > > Indeed that works too. > > I spoke too soon, with model=auto I get back the "Front" mixer that I > used during the git bisection to find the culprit, but I get no sound > at all in the plugged in headphones or the internal speakers. This is odd. Are you sure both are the same kernel but just one commit reverted? Takashi

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[alsa-devel] The state of Linux audio
by Fred . 26 Aug '08

26 Aug '08

What is the state of audio in Linux? How does it compare against other operating systems? Is the Linux audio functionality good or bad? What could be better? What is planned for the future?

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[alsa-devel] Sound Blaster Audigy (snd_ca0106 Creative Labs|SB Audigy LS): How to use the mixer?
by Colin Guthrie 25 Aug '08

25 Aug '08

Hi, Just trying to help a user with a problem and it seems to be down to the fact that his SB Audigy LS does not have a "Master" or "PCM" mixer control. I asked Lennart about this and he told that Takashi said he wanted to make sure that all devices have at least a Master vol control. This one doesn't! Please let me know what you need in order to fix this. This is with 1.0.17... I can't see anything in the changelog that would suggest this is fixed in the 1.0.18rc1. Please let me know if you need e.g. PCI ids etc. as I can ask the user to provide these. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] Simple mixer control 'Line in',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 255 [100%] [24.00dB] Front Right: Capture 255 [100%] [24.00dB] Simple mixer control 'Mic',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 255 [100%] [24.00dB] Front Right: Capture 255 [100%] [24.00dB] Simple mixer control 'Phone',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 207 [81%] [0.00dB] Front Right: Capture 207 [81%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Center/LFE',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'IEC958 Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'IEC958 Rear',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'IEC958 Unknown',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'Aux',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 207 [81%] [0.00dB] Front Right: Capture 207 [81%] [0.00dB] Simple mixer control 'Analog Center/LFE',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Front',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Rear',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Side',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Source',0 Capabilities: cenum Items: 'Phone' 'Mic' 'Line in' 'Aux' Item0: 'Line in' Simple mixer control 'CAPTURE feedback',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'Digital Source',0 Capabilities: cenum Items: 'IEC958 out' 'i2s mixer out' 'IEC958 in' 'i2s in' 'AC97 in' 'SRC out' Item0: 'i2s in' Simple mixer control 'Shared Mic/Line in',0 Capabilities: cenum Items: 'Line in' 'Mic in' Item0: 'Line in'

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[alsa-devel] Online code snapshots on ALSA server
by Jaroslav Kysela 25 Aug '08

25 Aug '08

Hi all, finally, I made possible to download a very fresh ALSA code in tarballs from our repository - http://www.alsa-project.org/snapshot/ . Users can get the recent code immediately now. There is no reason to wait a day or so after commit to the GIT repository now. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] PATCH - via82xx vt1618 7.1 Audio Support
by John L. Utz III 25 Aug '08

25 Aug '08

Makes Audio > 2.0 work on the VIA vt1618 7.1 Channel AC97 Codec Patched against ALSA-GIT Fri Aug 22 16:24:52 PDT 2008 Authored-by: John L. Utz III john.utz(a)dmx.com Signed-off-by: John L. Utz III john.utz(a)dmx.com diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..89f3050 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..24f1e98 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * noodling the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3498,7 +3498,7 @@ static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */ - /* grab our desirec bits, then mash them together in a manner + /* grab our desired bits, then mash them together in a manner * consistent with Table 6 on page 17 in the 1617a docs */ usSM51 = snd_ac97_read(pac97, 0x7a) >> 14; @@ -3576,6 +3576,200 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } +/* VIA VT1618 8 CHANNEL AC97 CODEC */ + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *acTxtAux[] = {"Aux In", "Back Surr Out"}; + + return ac97_enum_text_info(kcontrol, uinfo, acTxtAux, 2); +} + +static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = + (snd_ac97_read(snd_kcontrol_chip(kcontrol), 0x5c) & 0x0008)>>3; + return 0; +} + +static int snd_ac97_vt1618_aux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* toggle surround rear dac power */ + + snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); + + /* toggle aux in surround rear out jack */ + + return snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack and it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get surround output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be + * applicable to them. + */ + +static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + /* cribbed from ac97_surround_jack_mode_info() ordering in these lists + * reflects vt1618 docs for Vendor Defined Register 0x60 */ + + static const char *acTxtUAJ0[] = + { "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In"}; + + static const char *acTxtUAJ1[] = + { "Surround Out", "DAC Unmixed Out", "Line In", "Mic In"}; + + static const char *acTxtUAJ2[] = + { "Center LFE Out", "DAC Unmixed Out", "Line In", "Mic In"}; + + const char **ppacTxt = acTxtUAJ0; + + if (0x000C == kcontrol->private_value) + ppacTxt = acTxtUAJ1; + + if (0x0030 == kcontrol->private_value) + ppacTxt = acTxtUAJ2; + + return ac97_enum_text_info(kcontrol, uinfo, ppacTxt, 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned short usDatPag, usUAJ, usShift = 0; + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + if (0x000C == kcontrol->private_value) + usShift = 2; + + if (0x0030 == kcontrol->private_value) + usShift = 4; + + mutex_lock(&pac97->page_mutex); + + usDatPag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + usUAJ = snd_ac97_read(pac97, 0x60) & kcontrol->private_value; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, usDatPag); + mutex_unlock(&pac97->page_mutex); + + ucontrol->value.enumerated.item[0] = usUAJ >> usShift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned short usSft = 0; + + if (0x000C == kcontrol->private_value) + usSft = 2; + + if (0x0030 == kcontrol->private_value) + usSft = 4; + + /* UAJ1 and UAJ2 are not supposed to have 00 written to them?? i + * dunno, because thats something that i have to do to get 5.1 out to + * work. */ + + return ac97_update_bits_page(snd_kcontrol_chip(kcontrol), + 0x60, kcontrol->private_value, + ucontrol->value.enumerated.item[0]<<usSft, + 0); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + AC97_SINGLE("Exchange Center/LFE", 0x5a, 8, 1, 0), + AC97_SINGLE("DC Offset", 0x5a, 10, 1, 0), + AC97_SINGLE("Soft Mute", 0x5c, 0, 1, 1), + AC97_SINGLE("Headphone Amp", 0x5c, 5, 1, 1), + AC97_DOUBLE("Back Surr Volume", 0x5e, 8, 0, 31, 1), + AC97_SINGLE("Back Surr Switch", 0x5e, 15, 1, 1), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Jack", + .info = snd_ac97_vt1618_aux_info, + .get = snd_ac97_vt1618_aux_get, + .put = snd_ac97_vt1618_aux_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0x0003 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Jack", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0x000C + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Jack", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0x0030 + }, +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97)

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[alsa-devel] ac97_channel_mode_info() needs a "0ch" added.
by John L. Utz III 25 Aug '08

25 Aug '08

When attempting to leverage existing ALSA AC97 infrastructure, i realized that the existing infrastructure does not account for the 'prosumer-y' concept of repurposing ALL of the jacks as input, including the speaker. The vt1618 can use any of it's jacks as mic in or line in. if all of them are set to be inputs, then ac97_channel_mode_info() needs to return "Och" because there would be no output channels. that would imply that ac97->channel_mode would equal 0, yes? tnx for any input you can provide.

2 1

[alsa-devel] [PATCH] CA0106 on MSI K8N Diamond PLUS Motherboard
by Travis Place 25 Aug '08

25 Aug '08

Previously i posted a patch for a new ssid for the onboard ca0106 on the K8N Diamond motherboard. Now, it is apparently really called a 'K8N Diamond PLUS', and also requires the gpio_type to be changed too. Attached is a patch for this. Changelog: Correct a previous patch for the ca0106 onboard the MSI K8N Diamond PLUS motherboard. Confirmed to have Line/Mic/Aux working for input, and sound output working as expected. Signed-off-by: Travis Place <wishie(a)wishie.net>

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[alsa-devel] [PATCH 001/002] snd-powermac: mixers for PowerMac G4 AGP
by Risto Suominen 25 Aug '08

25 Aug '08

2 1

[alsa-devel] [PATCH] wss_lib: do not mess mixer settings during probe
by Krzysztof Helt 25 Aug '08

25 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss_dout function which does not mess shadowed register values during chip probing. Otherwise, user ends up with stupid mixer settings after driver loading. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Especially, the recording settings are messed up like left channel recording from different source than right channel. diff -urp linux-ref/sound/isa/wss/wss_lib.c linux-2.6/sound/isa/wss/wss_lib.c --- linux-ref/sound/isa/wss/wss_lib.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/wss/wss_lib.c 2008-08-25 21:13:14.000000000 +0200 @@ -1162,9 +1162,9 @@ static int snd_ad1848_probe(struct snd_w spin_lock_irqsave(&chip->reg_lock, flags); /* set CS423x MODE 1 */ - snd_wss_out(chip, CS4231_MISC_INFO, 0); + snd_wss_dout(chip, CS4231_MISC_INFO, 0); - snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */ + snd_wss_dout(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */ r = snd_wss_in(chip, CS4231_RIGHT_INPUT); if (r != 0x45) { /* RMGE always high on AD1847 */ @@ -1174,7 +1174,7 @@ static int snd_ad1848_probe(struct snd_w } hardware = WSS_HW_AD1847; } else { - snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); + snd_wss_dout(chip, CS4231_LEFT_INPUT, 0xaa); r = snd_wss_in(chip, CS4231_LEFT_INPUT); /* L/RMGE always low on AT2320 */ if ((r | CS4231_ENABLE_MIC_GAIN) != 0xaa) { @@ -1199,7 +1199,7 @@ static int snd_ad1848_probe(struct snd_w r = snd_wss_in(chip, CS4231_MISC_INFO); /* set CS423x MODE 2 */ - snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + snd_wss_dout(chip, CS4231_MISC_INFO, CS4231_MODE2); for (i = 0; i < 16; i++) { if (snd_wss_in(chip, i) != snd_wss_in(chip, 16 + i)) { /* we have more than 16 registers: check ID */ @@ -1221,7 +1221,7 @@ static int snd_ad1848_probe(struct snd_w else chip->hardware = WSS_HW_AD1848; out_mode: - snd_wss_out(chip, CS4231_MISC_INFO, 0); + snd_wss_dout(chip, CS4231_MISC_INFO, 0); out: spin_unlock_irqrestore(&chip->reg_lock, flags); return err; ---------------------------------------------------------------------- Zobacz galerie - tak wygladaja wampiry! >> link http://link.interia.pl/f1eee

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[alsa-devel] [ALSA] remove unused #include <version.h>
by Huang Weiyi 25 Aug '08

25 Aug '08

The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/mips/au1x00.c sound/soc/at32/playpaq_wm8510.c sound/soc/at91/eti_b1_wm8731.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index ee0741f..fbef38a 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -38,7 +38,6 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <linux/slab.h> -#include <linux/version.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 3f32621..fd2b2f3 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -22,7 +22,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/errno.h> #include <linux/clk.h> diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index b081e83..b81d6b2 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -22,7 +22,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h>

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[alsa-devel] alsa-utils,alsa-plugins v1.0.18rc2
by Jaroslav Kysela 25 Aug '08

25 Aug '08

Hello all, I released alsa-utils and alsa-plugins v1.0.18rc2 packages. The changelog is available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.18rc1_v1.0.18rc2 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [ALSA PATCH] alsa-git merge request
by Jaroslav Kysela 25 Aug '08

25 Aug '08

Linus, please pull from: git pull git://git.alsa-project.org/alsa-kernel.git for-linus gitweb interface: http://git.alsa-project.org/?p=alsa-kernel.git;a=shortlog;h=for-linus The GNU patch is available at: ftp://ftp.alsa-project.org/pub/kernel-patches/alsa-git-for-linus-2008-07-14… The following files will be updated: Documentation/sound/alsa/ALSA-Configuration.txt | 17 +- .../sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 4 +- include/asm-mips/mach-au1x00/au1xxx_psc.h | 8 + include/sound/ad1843.h | 46 + include/sound/control.h | 3 - include/sound/core.h | 8 +- include/sound/cs4231-regs.h | 8 + include/sound/cs4231.h | 3 + include/sound/emu10k1.h | 1 + include/sound/seq_kernel.h | 2 +- include/sound/soc-dapm.h | 42 +- include/sound/soc.h | 175 ++- include/sound/uda1341.h | 2 - include/sound/version.h | 4 +- sound/Kconfig | 34 +- sound/aoa/Kconfig | 11 +- sound/aoa/codecs/Kconfig | 4 - sound/aoa/fabrics/Kconfig | 1 - sound/aoa/soundbus/Kconfig | 1 - sound/arm/Kconfig | 21 +- sound/arm/sa11xx-uda1341.c | 2 - sound/core/Kconfig | 29 +- sound/core/control.c | 7 +- sound/core/init.c | 67 +- sound/core/memalloc.c | 62 - sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_device.c | 6 +- sound/core/sound.c | 8 +- sound/core/timer.c | 6 +- sound/drivers/Kconfig | 91 +- sound/drivers/vx/vx_hwdep.c | 2 +- sound/i2c/cs8427.c | 6 +- sound/i2c/l3/uda1341.c | 2 - sound/isa/Kconfig | 61 +- sound/isa/cs423x/cs4231_lib.c | 118 ++- sound/isa/opti9xx/opti92x-ad1848.c | 1126 +------------- sound/isa/sb/Makefile | 2 - sound/isa/wavefront/wavefront_synth.c | 2 +- sound/mips/Kconfig | 27 +- sound/mips/Makefile | 4 + sound/mips/ad1843.c | 561 +++++++ sound/mips/hal2.c | 947 ++++++++++++ sound/mips/hal2.h | 245 +++ sound/mips/sgio2audio.c | 1006 ++++++++++++ sound/oss/Kconfig | 49 +- sound/oss/dmasound/dmasound_core.c | 7 +- sound/oss/dmasound/dmasound_paula.c | 2 +- sound/oss/dmasound/dmasound_q40.c | 2 +- sound/oss/msnd.c | 2 - sound/oss/msnd.h | 2 - sound/oss/msnd_classic.h | 2 - sound/oss/msnd_pinnacle.c | 5 - sound/oss/msnd_pinnacle.h | 2 - sound/parisc/Kconfig | 13 +- sound/pci/Kconfig | 104 +-- sound/pci/Makefile | 2 +- sound/pci/ac97/Makefile | 12 +- sound/pci/ac97/ac97_codec.c | 11 +- sound/pci/ac97/ac97_patch.c | 81 +- sound/pci/{ac97 => }/ak4531_codec.c | 34 +- sound/pci/au88x0/au88x0_game.c | 2 - sound/pci/azt3328.c | 1235 ++++++++++----- sound/pci/azt3328.h | 207 +++- sound/pci/ca0106/ca0106_main.c | 5 + sound/pci/emu10k1/emu10k1_main.c | 1 + sound/pci/emu10k1/emumixer.c | 13 +- sound/pci/emu10k1/memory.c | 69 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 306 +++- sound/pci/hda/hda_proc.c | 5 +- sound/pci/hda/patch_analog.c | 38 +- sound/pci/hda/patch_conexant.c | 33 +- sound/pci/hda/patch_realtek.c | 548 +++++++- sound/pci/hda/patch_sigmatel.c | 71 +- sound/pci/ice1712/envy24ht.h | 10 +- sound/pci/ice1712/ice1712.h | 2 + sound/pci/ice1712/ice1724.c | 213 ++- sound/pci/maestro3.c | 42 +- sound/pci/nm256/nm256.c | 4 +- sound/pci/oxygen/hifier.c | 33 +- sound/pci/oxygen/oxygen.c | 76 +- sound/pci/oxygen/oxygen.h | 14 + sound/pci/oxygen/oxygen_io.c | 22 +- sound/pci/oxygen/oxygen_lib.c | 106 ++- sound/pci/oxygen/oxygen_pcm.c | 53 +- sound/pci/oxygen/virtuoso.c | 252 ++-- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/pcxhr/pcxhr_core.c | 18 +- sound/pci/trident/trident_main.c | 5 +- sound/pci/trident/trident_memory.c | 178 --- sound/pci/via82xx.c | 6 + sound/pci/ymfpci/ymfpci_main.c | 2 + sound/pcmcia/Kconfig | 15 +- sound/pcmcia/vx/vxp_ops.c | 2 +- sound/ppc/Kconfig | 26 +- sound/ppc/daca.c | 2 - sound/ppc/tumbler.c | 2 - sound/sh/Kconfig | 16 +- sound/soc/Kconfig | 19 +- sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 + sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 ++++++ sound/soc/at32/at32-pcm.h | 79 + sound/soc/at32/at32-ssc.c | 849 ++++++++++ sound/soc/at32/at32-ssc.h | 59 + sound/soc/at32/playpaq_wm8510.c | 522 +++++++ sound/soc/at91/Kconfig | 2 +- sound/soc/at91/at91-pcm.c | 6 +- sound/soc/at91/at91-ssc.c | 12 +- sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 53 +- sound/soc/au1x/Kconfig | 32 + sound/soc/au1x/Makefile | 13 + sound/soc/au1x/dbdma2.c | 421 +++++ sound/soc/au1x/psc-ac97.c | 387 +++++ sound/soc/au1x/psc-i2s.c | 414 +++++ sound/soc/au1x/psc.h | 53 + sound/soc/au1x/sample-ac97.c | 144 ++ sound/soc/codecs/Kconfig | 22 +- sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ac97.c | 31 +- sound/soc/codecs/ac97.h | 2 +- sound/soc/codecs/ak4535.c | 696 +++++++++ sound/soc/codecs/ak4535.h | 46 + sound/soc/codecs/cs4270.c | 8 +- sound/soc/codecs/cs4270.h | 2 +- sound/soc/codecs/tlv320aic3x.c | 384 +++-- sound/soc/codecs/tlv320aic3x.h | 55 +- sound/soc/codecs/uda1380.c | 852 ++++++++++ sound/soc/codecs/uda1380.h | 89 ++ sound/soc/codecs/wm8510.c | 817 ++++++++++ sound/soc/codecs/wm8510.h | 103 ++ sound/soc/codecs/wm8731.c | 79 +- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8750.c | 87 +- sound/soc/codecs/wm8750.h | 2 +- sound/soc/codecs/wm8753.c | 183 ++-- sound/soc/codecs/wm8753.h | 2 +- sound/soc/codecs/wm8990.c | 1626 ++++++++++++++++++++ sound/soc/codecs/wm8990.h | 832 ++++++++++ sound/soc/codecs/wm9712.c | 53 +- sound/soc/codecs/wm9712.h | 2 +- sound/soc/codecs/wm9713.c | 79 +- sound/soc/codecs/wm9713.h | 2 +- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-evm.c | 40 +- sound/soc/davinci/davinci-i2s.c | 16 +- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/fsl/Kconfig | 6 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_dma.h | 2 +- sound/soc/fsl/fsl_ssi.c | 24 +- sound/soc/fsl/fsl_ssi.h | 4 +- sound/soc/fsl/mpc8610_hpcd.c | 72 +- sound/soc/omap/Kconfig | 4 - sound/soc/omap/n810.c | 106 +- sound/soc/omap/omap-mcbsp.c | 16 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/Kconfig | 11 +- sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/corgi.c | 70 +- sound/soc/pxa/em-x270.c | 102 ++ sound/soc/pxa/poodle.c | 50 +- sound/soc/pxa/pxa2xx-ac97.c | 18 +- sound/soc/pxa/pxa2xx-ac97.h | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 17 +- sound/soc/pxa/pxa2xx-i2s.h | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 91 +- sound/soc/pxa/tosa.c | 47 +- sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/s3c24xx/neo1973_wm8753.c | 237 ++-- sound/soc/s3c24xx/s3c2412-i2s.c | 15 +- sound/soc/s3c24xx/s3c2412-i2s.h | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 15 +- sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 25 +- sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 6 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 3 - sound/soc/sh/Kconfig | 5 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/sh7760-ac97.c | 4 +- sound/soc/sh/ssi.c | 8 +- sound/soc/soc-core.c | 443 ++++-- sound/soc/soc-dapm.c | 344 ++++- sound/sparc/Kconfig | 17 +- sound/sparc/dbri.c | 2 +- sound/spi/Kconfig | 13 +- sound/usb/Kconfig | 16 +- sound/usb/caiaq/caiaq-audio.c | 1 + sound/usb/caiaq/caiaq-device.c | 12 +- sound/usb/caiaq/caiaq-device.h | 1 + sound/usb/usbaudio.c | 4 - sound/usb/usbquirks.h | 38 + 201 files changed, 16473 insertions(+), 3893 deletions(-) rename sound/pci/{ac97 => }/ak4531_codec.c (96%) Adrian Bunk (6): [ALSA] remove CVS keywords sound: sound/oss/: remove CVS keywords [ALSA] remove SND_GUS_SYNTH [ALSA] usbaudio.c: remove #ifndef CONFIG_USB_EHCI_SPLIT_ISO code ALSA: make sparc/dbri.c:snd_dbri_proc() static sound: sound/oss/dmasound/: cleanups Akio Idehara (1): [ALSA] hda - Add Toshiba dynabook SS RX1 support Andreas Mohr (2): [ALSA] PCI168 snd-azt3328 Linux driver: another huge update ALSA: PCI168 snd-azt3328: some more fixups Andy Green (1): ALSA: ASoC: Don't block system resume Chris Mennie (1): [ALSA] usb-audio - Support for Roland SonicCell sound module Clemens Ladisch (12): [ALSA] oxygen: fix version in MODULE_LICENSE [ALSA] oxygen: add symbol for I/O space size [ALSA] oxygen: save register writes [ALSA] oxygen: simplify DAC volume initialization [ALSA] oxygen: separate out hardware initialization code [ALSA] virtuoso: add xonar_enable_output() [ALSA] oxygen: add PM support [ALSA] oxygen: add symbols for buffer/period size constraints [ALSA] virtuoso: restrict period time to less than 10 s [ALSA] ice1724: fix MIDI ALSA: usb-audio: fix Yamaha KX quirk ALSA: usb-audio: add some Yamaha USB MIDI quirks Daniel Jacobowitz (1): [ALSA] ac97 - Add virtual master control to VT1616/VT1617A codec. Daniel Mack (3): [ALSA] soc - tlv320aic3x - revisit clock setup [ALSA] soc - tlv320aic3x - add GPIO support [ALSA] snd_usb_caiaq: add support for 'Session I/O' interface David Howells (2): ALSA: Fix a const pointer usage warning in the Digigram VX soundcard driver ALSA: Fix a const to non-const assignment in the Digigram VXpocket sound driver Dmitry Baryshkov (2): ALSA: tosa: fix compilation with new DAPM API ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration Geoffrey Wossum (1): [ALSA] Revised AT32 ASoC Patch Graeme Gregory (1): [ALSA] soc - DAPM - add hook to read state of DAPM widget Harvey Harrison (2): [ALSA] i2c: cs8427.c use put_unaligned helper alsa: add annotations to bitwise type snd_pcm_hw_param_t Jarkko Nikula (8): [ALSA] ASoC: Convert N810 machine driver to use gpiolib ALSA: ASoC: Cover also Nokia N810 WiMAX Edition in N810 machine driver ALSA: ASoC: Add support for generic DAPM register modifier widget ALSA: ASoC: TLV320AIC3X: Use register modifier widget for mic bias ALSA: ASoC: TLV320AIC3X: Modify only interface related bits in aic3x_set_dai_fmt ALSA: ASoC: TLV320AIC3X: Add support for digital microphone input ALSA: ASoC: Add digital mic configuration to N810 machine driver ALSA: ASoC: TLV320AIC3X: Add mixer control for ADC highpass filter Jaroslav Kysela (8): ALSA: Release v1.0.17rc1 ALSA: Release 1.0.17rc2 ALSA: Remove duplicate MODULE_AUTHOR/DESCRIPTION/LICENCE from snd-ens1370.ko ALSA: emu10k1 - fix possible memory leak in memory allocation routines [ALSA] Revert "alsa: add annotations to bitwise type snd_pcm_hw_param_t" ALSA: Release v1.0.17rc3 ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model ALSA: Release v1.0.17 Jiang zhe (1): [ALSA] hda - support intel DG33 motherboards Johannes Berg (1): ALSA: remove CONFIG_KMOD from sound Juergen Beisert (1): [ALSA] ac97: add support for wm9711 master left inv switch Kailang Yang (6): [ALSA] hda - Add support of Teradici controller [ALSA] hda - Add ICH9 controller support (8086:2911) [ALSA] hda - Add ALC663 support [ALSA] hda - Fix vref pincap check in alc882 auto-detection [ALSA] hda - show correct codec chip in PCM stream names [ALSA] hda - Fix EAPD and COEF setups for realtek codecs Krzysztof Helt (3): [ALSA] opti93x: fix sound ouput for Opti930 ALSA: opti93x: add support for Opti93x codec in cs4231-lib ALSA: opti93x: use cs4231 lib Liam Girdwood (16): ALSA: asoc: core - refactored DAPM pin control API. ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: omap - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: pxa - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: sh - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. ALSA: asoc: machines - add Digital Audio Interface (DAI) control functions. ALSA: asoc: n810 - fix build error. ALSA: asoc: at32 - DAI struct merge and enable_pin() change. ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform. Manuel Lauss (2): ALSA: ASoC: AC97 codec PM ALSA: ASoC: Au12x0/Au1550 PSC Audio support Mark Brown (30): [ALSA] soc - DAPM - Add bulk control registration [ALSA] soc - DAPM - Bulk route registration [ALSA] soc - tlv320aic3x - Convert to use bulk registration APIs [ALSA] soc - Zaurus - Convert to bulk DAPM registration APIs [ALSA] soc - eti_b1_wm8731 - Convert to use bulk DAPM control registration [ALSA] soc - neo1973_wm8753 - Convert to bulk DAPM registration APIs [ALSA] soc - davinci-evm - Update for bulk DAPM registration APIs [ALSA] soc - n810 - Update for bulk DAPM registration APIs [ALSA] ASoC: Remove in-code changelogs [ALSA] ASoC: Make CPU and codec DAI operations have same type [ALSA] ASoC: Clarify API for bias configuration [ALSA] ASoC: core checkpatch cleanups [ALSA] soc - Convert Wolfson codec drivers to use bulk DAPM registration [ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type [ALSA] ASoC: Add WM8510 driver [ALSA] ASoC: Add WM8990 driver [ALSA] ASoC: Fix default mono mixer configuration for WM8510 ALSA: ASoC: Fix register cache sizes for Wolfson codecs ALSA: ASoC: Fix register cache size for UDA1380 ALSA: ASoC: Tweak tlv320aicx reg_cache_size ALSA: ASoC: Add missing includes ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove ALSA: ASoC: Fix WM9713 voice PCM slave mode configuration ALSA: ASoC: Check for exact register match in wm97xx_reset() ALSA: ASoC: Replace custom debug macros with pr_ equivalents ALSA: ASoC: Advertise 16000Hz rate for WM9713 PCM interface ALSA: ASoC: Make pop/click debug wait times dynamically configurable ALSA: ASoC: Fix warning from strict_strtoul() ALSA: ASoC: Switch DAPM to use of standard DEBUG macro ALSA: ASoC: Dump DAPM state for non-stream changes Matthew Ranostay (3): ALSA: hda: Add support for 92HD73xxx codecs ALSA: hda - removed redundant gpio_mask ALSA: hda: 92hd71bxx PC Beep Mike Montour (3): ALSA: ASoC: Add TLV information to remaining WM8753 controls ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type ALSA: ASoC: Add TLV information to the LM4857 controls on the GTA01 Mike Rapoport (2): [ALSA] Add EM-X270 ASoC driver ALSA: ac97 - fix patch_ucb1400 for proper resume Milton Miller (1): ALSA: correct kcalloc usage Philipp Zabel (1): [ALSA] ASoC: Add UDA1380 driver Pierre Ossman (1): ALSA: trident - pause s/pdif output Randy Dunlap (1): ALSA: ASoC: fix PM=n build Richard Chan (1): ALSA: Add Yamaha KX49 (USB MIDI controller) to usbquirks.h Richard Purdie (1): ALSA: ASoC: Add AK4535 driver Stephen Rothwell (1): [ALSA] sound: fix export symbol typo Takashi Iwai (44): [ALSA] Remove unneeded ugly hack for i386 in memalloc.c [ALSA] hda - Fix DMA position inaccuracy sound: Convert to menuconfig [ALSA] Fix AC97 kconfig items [ALSA] Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE [ALSA] Improve the slots option handling [ALSA] Clean up sound/pci/ac97/Makefile [ALSA] Make ak4531 local to ens1370 driver [ALSA] trident - clean up obsolete synth codes sound: Clean up sound/oss/Kconfig [ALSA] trivial clean up of sound/isa/sb/Makefile [ALSA] emu10k1 - simplify page allocation for synth [ALSA] emu10k1 - Fix inverted Analog/Digital mixer switch on Audigy2 [ALSA] hda - Fix PLL gating control on Realtek codecs [ALSA] hda - increase max_codecs of ICH to 4 [ALSA] ac97 - Fix power_save option value as time-out ALSA: emu10k1 - Fix page allocation with GFP_DMA ALSA: hda - Add bdl_pos_adj option ALSA: hda - remove position_fix=3 ALSA: hda - bdl_pos_adj option to each instance ALSA: hda - Fix bdl_pos_adj value for ATI SB chipsets ALSA: hda - Add description of bdl_pos_adj option ALSA: hda - Add a warning if pending IRQ is found ALSA: hda - Fix stac9205_cfg_tbl ALSA: hda - Remove unused mutex ALSA: make snd_ctl_elem_read() and snd_ctl_elem_write() static ALSA: hda - bdl_pos_adj=32 as default ALSA: hda - use upper_32_bits() ALSA: emu10k1 - simplify the last fix ALSA: hda - Fix wrong volumes in AD1988 auto-probe mode ALSA: ca0106 - Add entry for another MSI K8N Diamond MB ALSA: hda - Fix digital converter proc output ALSA: hda - Add MacBook 3.1 support ALSA: ymfpci - fix initial volume for 44.1kHz output ALSA: hda - disable amp override on non-HP machines ALSA: Fix AC97 power down ALSA: ALSA: hda - Fix ALC883 medion model ALSA: hda - Add missing Thinkpad Z60m support ALSA: ALSA: hda - Fix speaker output on Toshiba P105 ALSA: hda - Add support for Lenovo 3000 N200 ALSA: wavefront - add const ALSA: hda - Fix internal mic vref pin setup ALSA: hda - Fix missing init for unsol events on micsense model ALSA: hda - Fix FSC V5505 model Thomas Bogendoerfer (2): ALSA: ALSA driver for SGI HAL2 audio device ALSA: ALSA driver for SGI O2 audio board Tim Niemeyer (1): [ALSA] soc - Patch to add debug messages to the neo1973_wm8753 (GTA01) sound driver Timur Tabi (1): ALSA: Fix register programming in Freescale MPC8610 HPCD sound driver Travis Place (2): ALSA: hda - Added model selection for iMac 24" ALSA: hda - Added SSID for 'Fujitsu Siemens Amilo M1451G' laptop Ville Syrjälä (1): [ALSA] maestro3: Fix hw volume on HP OmniBook Walter Sheets (1): ALSA: via82xx - Add VIA audio device #1841 to ac97_quirk list Werner Almesberger (1): [ALSA] soc - Fix s3c24xx-i2s LR sync while timer ticks are disabled ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] repost - snd_via82xx input channels problem
by Vaclav Peroutka 24 Aug '08

24 Aug '08

thank you for your answer. Finally I found, that backporting is not so necessary for me as I needed just three drivers - bt87x, via82xx and ice1724. Those drivers do not use newer functions I mentioned before, so the compilation has successfully ended. But I noticed that even with the newset driver the alc655 line-in inputs do not work well. Before ( alsa 1.0.9 ) I had the right channel always connected to the Mic input. Now (Alsa 2008-08-22 ) in the right channel there is a copy of the left channel. Thus I have monophonic input instead of stereo. I just downloaded the datasheet of ALC655 codec. I will read it hopefully tomorrow. But can you just advice me where to look to try to fix the problem ? It should be somewhere in snd_via82xx module ? Or somewhere else ? Inputs in snd_ice1724 (my second sound card) work properly. But I have never worked with kernel modules or kernel programming. I am just interested how to fix the problem I have. Thank you in advance, Vaclav

1 0

[alsa-devel] [PATCH 002/002] snd-powermac: HP detection for 1st iMac G3 SL
by Risto Suominen 23 Aug '08

23 Aug '08

1 0

[PATCH] ca0106 playback 44100Hz support to SPDIF and playback format & rate constraints
by Ben Stanley 23 Aug '08

23 Aug '08

Squashed commit of the following: commit 8bfbd000b4b602b517b5ab1edc5610ef39dced1e Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sat Aug 23 23:19:57 2008 +1000 Correct style and log messages according to Takashi's review comments. commit eb973ebecfac64b6f0e60df90eef8885ccc64aaf Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:56:15 2008 +1000 Annotate bug in comments. commit df725ab6af87e1c94d1d41e31e3865437b1999e0 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:44:12 2008 +1000 Checkpatch.pl round 4. commit c277bfa1773595b98c5447a7a085f21ccead110f Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:39:39 2008 +1000 Checkpatch.pl changes round 3. commit a3f9b1faae250aa0786ef68c3c53c3915e42972b Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:18:36 2008 +1000 Second round of checkpatch.pl changes. commit 3f45e4895fc8419a4a90eddf29af4f8547aa105d Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sun Aug 17 00:06:47 2008 +1000 Satisfy checkpatch.pl pass 1 commit 88424052a25359aa7ffa840e0515a7a93278f931 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sat Aug 16 22:32:25 2008 +1000 44100Hz playback support to SPDIF and playback format & rate constraints commit e0795c3bc756f07986eaae7c4ab54a78d3fecc80 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sat Aug 16 22:25:38 2008 +1000 Remove use of snd_assert. commit 097b03e9ff07432c45a9865bd02208b20b09beb3 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Thu Jul 24 02:59:38 2008 +1000 * fix cosmetic bug commit 75769f154ea53b4215de37d314351c4e45bd2543 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Thu Jul 3 20:47:31 2008 +1000 44100Hz playback support to SPDIF and playback format & rate constraints --- pci/ca0106/ca0106.h | 7 +- pci/ca0106/ca0106_main.c | 265 ++++++++++++++++++++++++++++++++++++++-------- 2 files changed, 228 insertions(+), 44 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..376fa9c 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.22 + * Version: 0.0.23 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -49,6 +49,8 @@ * Implement support for Line-in capture on SB Live 24bit. * 0.0.22 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.23 + * Add support for playback sampling rate and format constraints. * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -644,6 +646,8 @@ #include "ca_midi.h" +#define DRVNAME "snd-ca0106" + struct snd_ca0106; struct snd_ca0106_channel { @@ -659,6 +663,7 @@ struct snd_ca0106_pcm { struct snd_pcm_substream *substream; int channel_id; unsigned short running; + unsigned short hw_reserved; }; struct snd_ca0106_details { diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 2f8b28a..b06f7a9 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.25 + * Version: 0.0.26 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -83,9 +83,14 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * 0.0.25 * Powerdown SPI DAC channels when not in use + * 0.0.26 Ben Stanley + * Added support for output at 44100Hz rate (SPDIF only). + * Implemented constraints system for output rate and format. * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. + * Some programs fail to produce sound output (tested on SPDIF). See + * http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 * -- * * TODO: @@ -145,6 +150,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/ac97_codec.h> #include <sound/info.h> @@ -284,9 +290,9 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000), - .rate_min = 48000, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, @@ -318,6 +324,110 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; +static unsigned int all_spdif_playback_rates[] = + {44100, 48000, 96000, 192000}; + +static int hw_rule_playback_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int chi, any_44100 = 0, any_non_44100 = 0, mask; + struct snd_ca0106_channel *chp; + struct snd_pcm_runtime *runtime; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (chip->spdif_enable) { + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printdd("snd_hw_rule_playback_rate: ch=%d, " + "rate=%d.\n", chi, runtime->rate); + any_44100 += runtime->rate == 44100; + any_non_44100 += runtime->rate != 44100; + } + if (snd_BUG_ON(any_44100 && any_non_44100)) + return -EINVAL; + /* Compute the mask applied to all_spdif_playback_rates */ + if (any_44100) + mask = 0x1; + else if (any_non_44100) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC (FIXME Why?) */ + mask = 0xE; + } + snd_printdd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + any_44100, any_non_44100, mask, chip->spdif_enable); + return snd_interval_list(hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE), + ARRAY_SIZE(all_spdif_playback_rates), + all_spdif_playback_rates, mask); +} + +static int hw_rule_playback_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int chi, any_S16 = 0, any_S32 = 0; + struct snd_ca0106_channel *chp; + struct snd_pcm_runtime *runtime; + struct snd_mask fmt, *f = hw_param_mask( + params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printdd("snd_hw_rule_playback_format: ch=%d, format=%d.\n", + chi, runtime->format); + any_S16 += runtime->format == SNDRV_PCM_FORMAT_S16_LE; + any_S32 += runtime->format == SNDRV_PCM_FORMAT_S32_LE; + } + if (snd_BUG_ON(any_S16 && any_S32)) + return -EINVAL; + if (any_S16) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (any_S32) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + else { + /* No format yet chosen, so both formats are available. */ + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + } + result = snd_mask_refine(f, &fmt); + snd_printdd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + any_S16, any_S32, f->bits[0], fmt.bits[0], result); + return result; +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -508,10 +618,24 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + err = snd_pcm_hw_constraint_integer( + runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + err = snd_pcm_hw_constraint_step( + runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_playback_rate, (void *)chip, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_playback_format, (void *)chip, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -646,6 +770,9 @@ static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream /* hw_free callback */ static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + epcm->hw_reserved = 0; return snd_pcm_lib_free_pages(substream); } @@ -667,53 +794,103 @@ static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime, *runtimei; struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; + struct snd_ca0106_channel *chp; + int channel = epcm->channel_id, chi, any_44100 = 0, any_non_44100 = 0; u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; u32 hcfg_set = 0x00000000; u32 hcfg; - u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_mask = 0xFF0000; u32 reg40_set = 0; u32 reg40; - /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask; + u32 reg71_shift; u32 reg71_set = 0; u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); - /* Rate can be set per channel. */ - /* reg40 control host to fifo */ - /* reg71 controls DAC rate. */ - switch (runtime->rate) { - case 44100: - reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; - break; - case 48000: - reg40_set = 0; - reg71_set = 0; - break; - case 96000: - reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; - break; - case 192000: - reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; - break; - default: - reg40_set = 0; - reg71_set = 0; - break; + epcm->hw_reserved = 1; + /* FIXME CLEAN UP IF spdif_enable IS CHANGED WHILE CHANNELS ARE OPENED + * OR PREVENT THIS FROM HAPPENING. */ + if (emu->spdif_enable) + reg71_shift = 24; /* SPDIF Output Rate */ + else + reg71_shift = 16; /* I2S Output Rate */ + reg71_mask = 0x3 << reg71_shift; + + /*printk(KERN_DEBUG DRVNAME ": prepare_playback: " + "channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld,period_size=%ld, periods=%u, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1));*/ + /*printk("dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr,runtime->dma_area, table_base);*/ + /*printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr,emu->buffer.area, emu->buffer.bytes);*/ + /* We are forced to build the settings for all the channels. */ + for (chi = 0; chi < 4; ++chi) { + chp = &(emu->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (chi != channel && !chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtimei = chp->epcm->substream->runtime; + any_44100 += runtimei->rate == 44100; + any_non_44100 += runtimei->rate != 44100; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* We only support 44100 to spdif, not to DAC. + (FIXME WHY?)*/ + if (emu->spdif_enable) { + /* When using 44100, *all* channels + must be set to that rate. */ + reg40_set |= 0x550000; + reg71_set |= 0x1 << reg71_shift; + break; + } else { + printk(KERN_ERR DRVNAME + "prepare_playback: " + "44100Hz is invalid for DAC.\n"); + } + case 48000: + /* reg40_set &= !(0x1 << (chi<<1)); */ + /* reg71_set &= !(0x1 << reg71_shift); */ + break; + case 96000: + reg40_set |= 0x20000 << (chi<<1); + reg71_set |= 0x2 << reg71_shift; + break; + case 192000: + reg40_set |= 0x30000 << (chi<<1); + reg71_set |= 0x3 << reg71_shift; + break; + default: + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %d.\n", + runtimei->rate); + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + snd_printdd(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + any_44100, any_non_44100, emu->spdif_enable); + /* Format is a global setting. */ + /* Only the first channel accessed can set this + (enforced by constraints). */ switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: hcfg_set = 0; @@ -1363,8 +1540,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); @@ -1378,7 +1555,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } chip->details = c; if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + printk(KERN_INFO DRVNAME + ": Sound card name=%s, subsystem=0x%x. " + "Forced to subsystem=0x%x\n", c->name, chip->serial, subsystem[dev]); } -- 1.5.4.3 --=-lsMZJ4IxY7skpnVaSzUT--

1 0

[alsa-devel] [PATCH] ALSA: ASoC V2: report error if DMA doesn't start in FSL MPC8610 sound drivers
by Timur Tabi 23 Aug '08

23 Aug '08

Return an error if the DMA's current pointer is not within the bounds of the DMA buffer. This can happen if the DMA controller is not programmed correctly, or if the SSI doesn't communicate with the DMA controller correctly. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2. sound/soc/fsl/fsl_dma.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index cb1e5ae..b4ae486 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -689,6 +689,15 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "DMA(%u,%u) pointer is out of range, halting stream\n", + dma_private->dma_info->controller_id, + dma_private->dma_info->channel_id); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* -- 1.5.5

2 6

[alsa-devel] Compilation of Alsa 1.0.17 on Suse10 crashes
by Vaclav Peroutka 23 Aug '08

23 Aug '08

Hello all, I downloaded alsa-driver-1.0.17 and tried to compile it. I started with "./configure --with-suse=auto --with-oss=yes --with-cards=all,mts64=n" and then "make". But I always get following: ... CC [M] /tmp/alsa-driver-1.0.17/drivers/mts64.o /tmp/alsa-driver-1.0.17/drivers/mts64.c: In function ‘snd_mts64_attach’: /tmp/alsa-driver-1.0.17/drivers/mts64.c:896: error: implicit declaration of function ‘platform_device_alloc’ /tmp/alsa-driver-1.0.17/drivers/mts64.c:896: warning: assignment makes pointer from integer without a cast /tmp/alsa-driver-1.0.17/drivers/mts64.c:903: error: implicit declaration of function ‘platform_device_add’ /tmp/alsa-driver-1.0.17/drivers/mts64.c:904: error: implicit declaration of function ‘platform_device_put’ make[4]: *** [/tmp/alsa-driver-1.0.17/drivers/mts64.o] Error 1 make[3]: *** [/tmp/alsa-driver-1.0.17/drivers] Error 2 make[2]: *** [_module_/tmp/alsa-driver-1.0.17] Error 2 make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.13-15' My GCC: > gcc --version gcc (GCC) 4.0.2 20050901 (prerelease) (SUSE Linux) Is there something missing ? What should I do now ? Thank you in advance, Vaclav

3 3

[alsa-devel] Does anybody reading this list have an MB with a vt1618 with all four jacks?
by John L. Utz III 23 Aug '08

23 Aug '08

Our vt1618 motherboards for our product only has 3 of the 4 possible jacks, so i cant test my changes to support the Aux In/Surround Back Out Jack. I have reason to think that i have done it correctly, but it would be nice to get a blessing from somebody that actually has them. tnx! johnu

1 0

[alsa-devel] [PATCH] alsa-info.sh - Add checks for device permissions
by Travis Place 22 Aug '08

22 Aug '08

Changelog: 1) Show if the script was run as root, or as a user. 2) Check the group permissions of the ALSA device nodes, and check to see if the user is in that group, and display a warning if they are not. 3) Bumped version to 0.4.52 Signed-off-by: Travis Place <wishie(a)wishie.net> Notes: Im sure this code is far from perfect, and welcome any changes to it.

1 0

[alsa-devel] [PATCH 2/2] ca0106 playback 44100Hz support to SPDIF and playback format & rate constraints
by Ben Stanley 22 Aug '08

22 Aug '08

Signed-off-by: Ben Stanley Ben.Stanley(a)exemail.com.au This is the patch against alsa-driver. --- pci/ca0106/ca0106_main.patch | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/pci/ca0106/ca0106_main.patch b/pci/ca0106/ca0106_main.patch index 03820b3..d661011 100644 --- a/pci/ca0106/ca0106_main.patch +++ b/pci/ca0106/ca0106_main.patch @@ -14,12 +14,12 @@ static struct snd_ca0106_details ca0106_chip_details[] = { /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ -@@ -1352,7 +1353,7 @@ +@@ -1542,7 +1543,7 @@ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, -- pci->revision, chip->serial); -+ snd_pci_revision(pci), chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", +- chip->model, pci->revision, chip->serial); ++ chip->model, snd_pci_revision(pci), chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); -- 1.5.4.3

1 0

[alsa-devel] [PATCH - asihpi 4/6] Mem buffer alloc inside adapter mutex.
by linux＠audioscience.com 22 Aug '08

22 Aug '08

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/hpioctl.c b/pci/asihpi/hpioctl.c index 8222178..cf7c702 100644 --- a/pci/asihpi/hpioctl.c +++ b/pci/asihpi/hpioctl.c @@ -139,7 +139,20 @@ long asihpi_hpi_ioctl( pa = &adapters[hm.wAdapterIndex]; hr.wSize = 0; if (hm.wObject == HPI_OBJ_SUBSYSTEM) { - HPI_MessageF(&hm, &hr, file); + switch (hm.wFunction) { + case HPI_SUBSYS_CREATE_ADAPTER: + case HPI_SUBSYS_DELETE_ADAPTER: + /* Application must not use these functions! */ + hr.wSize = sizeof(struct hpi_response_header); + hr.wError = HPI_ERROR_INVALID_OPERATION; + hr.wFunction = hm.wFunction; + uncopied_bytes = copy_to_user(phr, &hr, hr.wSize); + if (uncopied_bytes) + return -EFAULT; + return 0; + default: + HPI_MessageF(&hm, &hr, file); + } } else { u16 __user *ptr = NULL; u32 size = 0; @@ -164,16 +177,6 @@ long asihpi_hpi_ioctl( /* Dig out any pointers embedded in the message. */ switch (hm.wFunction) { - case HPI_SUBSYS_CREATE_ADAPTER: - case HPI_SUBSYS_DELETE_ADAPTER: - /* Application must not use these functions! */ - hr.wSize = sizeof(struct hpi_response_header); - hr.wError = HPI_ERROR_INVALID_OPERATION; - hr.wFunction = hm.wFunction; - uncopied_bytes = copy_to_user(phr, &hr, hr.wSize); - if (uncopied_bytes) - return -EFAULT; - return 0; case HPI_OSTREAM_WRITE: case HPI_ISTREAM_READ: /* Yes, sparse, this is correct. */ @@ -202,6 +205,9 @@ long asihpi_hpi_ioctl( "HPI could not allocate " "stream buffer size %d\n", size); + + mutex_unlock(&adapters[nAdapter]. + mutex); return -EINVAL; } @@ -216,10 +222,11 @@ long asihpi_hpi_ioctl( break; default: + size = 0; break; } - if (wrflag == 0) { + if (size && (wrflag == 0)) { uncopied_bytes = copy_from_user(pa->pBuffer, ptr, size); if (uncopied_bytes) @@ -231,7 +238,7 @@ long asihpi_hpi_ioctl( HPI_MessageF(&hm, &hr, file); - if (wrflag == 1) { + if (size && (wrflag == 1)) { uncopied_bytes = copy_to_user(ptr, pa->pBuffer, size); if (uncopied_bytes) HPI_DEBUG_LOG(WARNING, -- 1.5.4.3

2 3

[alsa-devel] [PATCH] ALS4000 driver work, step 2
by Andreas Mohr 22 Aug '08

22 Aug '08

Hi, - more register naming work - finally figured out that weird CR register stuff (and did I mention that I hate _really_ undecipherable open-coded values?) - fix handling of IRQ sharing in interrupt handler (hopefully properly, otherwise I'd be grateful to hear your pedantic comments ;) - add handy SPECS_PAGE references wherever useful - comments, cleanup - add me as module author Patch created on 2.6.27-rc1-git4 (diffed against my earlier submission), pushed through checkpatch.pl, verified on my ALS4000 card (listening to it right now of course). Thanks! (especially for your very fast handling) Next patch _will_ contain more useful things, I promise ;) Oh, and apologies to several people for my previous asbestos-requiring comments! Seems Bart's mail address is still valid, hopefully (and found that he's also a Winchip freak, heh). Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- sound/pci/als4000.c.sent 2008-08-18 22:12:27.000000000 +0200 +++ sound/pci/als4000.c 2008-08-21 21:30:10.000000000 +0200 @@ -2,7 +2,7 @@ * card-als4000.c - driver for Avance Logic ALS4000 based soundcards. * Copyright (C) 2000 by Bart Hartgers <bart(a)etpmod.phys.tue.nl>, * Jaroslav Kysela <perex(a)perex.cz> - * Copyright (C) 2002 by Andreas Mohr <hw7oshyuv3001(a)sneakemail.com> + * Copyright (C) 2002, 2008 by Andreas Mohr <hw7oshyuv3001(a)sneakemail.com> * * Framework borrowed from Massimo Piccioni's card-als100.c. * @@ -27,8 +27,10 @@ * bought an ALS4000 based soundcard, I was forced to base this driver * on reverse engineering. * - * Note: this is no longer true. Pretty verbose chip docu (ALS4000a.PDF) - * can be found on the ALSA web site. + * Note: this is no longer true (thank you!): + * pretty verbose chip docu (ALS4000a.PDF) can be found on the ALSA web site. + * Page numbers stated anywhere below with the "SPECS_PAGE:" tag + * refer to: ALS4000a.PDF specs Ver 1.0, May 28th, 1998. * * The ALS4000 seems to be the PCI-cousin of the ALS100. It contains an * ALS100-like SB DSP/mixer, an OPL3 synth, a MPU401 and a gameport @@ -59,7 +61,6 @@ * - value -> some port 0x0c0d * * ToDo: - * - Proper shared IRQ handling? * - by default, don't enable legacy game and use PCI game I/O * - power management? (card can do voice wakeup according to datasheet!!) */ @@ -79,7 +80,7 @@ #include <sound/sb.h> #include <sound/initval.h> -MODULE_AUTHOR("Bart Hartgers <bart(a)etpmod.phys.tue.nl>"); +MODULE_AUTHOR("Bart Hartgers <bart(a)etpmod.phys.tue.nl>, Andreas Mohr"); MODULE_DESCRIPTION("Avance Logic ALS4000"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS4000}}"); @@ -124,22 +125,22 @@ MODULE_DEVICE_TABLE(pci, snd_als4000_ids); enum als4k_iobase_t { - /* IOx: B == Byte, W = Word, D = DWord */ + /* IOx: B == Byte, W = Word, D = DWord; SPECS_PAGE: 37 */ ALS4K_IOD_00_AC97_ACCESS = 0x00, ALS4K_IOW_04_AC97_READ = 0x04, ALS4K_IOB_06_AC97_STATUS = 0x06, ALS4K_IOB_07_IRQSTATUS = 0x07, ALS4K_IOD_08_GCR_DATA = 0x08, ALS4K_IOB_0C_GCR_INDEX = 0x0c, - ALS4K_IOB_0E_SB_MPU_IRQ = 0x0e, + ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU = 0x0e, ALS4K_IOB_10_ADLIB_ADDR0 = 0x10, ALS4K_IOB_11_ADLIB_ADDR1 = 0x11, ALS4K_IOB_12_ADLIB_ADDR2 = 0x12, ALS4K_IOB_13_ADLIB_ADDR3 = 0x13, ALS4K_IOB_14_MIXER_INDEX = 0x14, ALS4K_IOB_15_MIXER_DATA = 0x15, - ALS4K_IOB_16_ESP_RST_PORT = 0x16, - ALS4K_IOB_16_CR1E_ACK_PORT = 0x16, /* 2nd function */ + ALS4K_IOB_16_ESP_RESET = 0x16, + ALS4K_IOB_16_ACK_FOR_CR1E = 0x16, /* 2nd function */ ALS4K_IOB_18_OPL_ADDR0 = 0x18, ALS4K_IOB_19_OPL_ADDR1 = 0x19, ALS4K_IOB_1A_ESP_RD_DATA = 0x1a, @@ -154,62 +155,137 @@ ALS4K_IOB_31_MIDI_COMMAND = 0x31, /* 2nd function */ }; -enum als4k_gcr_t { - /* all registers 32bit wide */ - ALS4K_GCR_8C_MISC_CTRL = 0x8c, - ALS4K_GCR_90_TEST_MODE_REG = 0x90, - ALS4K_GCR_91_DMA0_ADDR = 0x91, - ALS4K_GCR_92_DMA0_MODE_COUNT = 0x92, - ALS4K_GCR_93_DMA1_ADDR = 0x93, - ALS4K_GCR_94_DMA1_MODE_COUNT = 0x94, - ALS4K_GCR_95_DMA3_ADDR = 0x95, - ALS4K_GCR_96_DMA3_MODE_COUNT = 0x96, - ALS4K_GCR_99_DMA_EMULATION_CTRL = 0x99, - ALS4K_GCR_A0_FIFO1_CURRENT_ADDR = 0xa0, - ALS4K_GCR_A1_FIFO1_STATUS_BYTECOUNT = 0xa1, - ALS4K_GCR_A2_FIFO2_PCIADDR = 0xa2, - ALS4K_GCR_A3_FIFO2_COUNT = 0xa3, - ALS4K_GCR_A4_FIFO2_CURRENT_ADDR = 0xa4, - ALS4K_GCR_A5_FIFO1_STATUS_BYTECOUNT = 0xa5, - ALS4K_GCR_A6_PM_CTRL = 0xa6, - ALS4K_GCR_A7_PCI_ACCESS_STORAGE = 0xa7, - ALS4K_GCR_A8_LEGACY_CFG1 = 0xa8, - ALS4K_GCR_A9_LEGACY_CFG2 = 0xa9, - ALS4K_GCR_FF_DUMMY_SCRATCH = 0xff, -}; - -enum als4k_gcr_8c_t { - ALS4K_GCR_8C_IRQ_MASK_CTRL_ENABLE = 0x8000, - ALS4K_GCR_8C_CHIP_REV_MASK = 0xf0000 +enum als4k_iobase_0e_t { + ALS4K_IOB_0E_MPU_IRQ = 0x10, + ALS4K_IOB_0E_CR1E_IRQ = 0x40, + ALS4K_IOB_0E_SB_DMA_IRQ = 0x80, }; -static inline void snd_als4000_gcr_write_addr(unsigned long iobase, - enum als4k_gcr_t reg, - u32 val) +enum als4k_gcr_t { /* all registers 32bit wide; SPECS_PAGE: 38 to 42 */ + ALS4K_GCR8C_MISC_CTRL = 0x8c, + ALS4K_GCR90_TEST_MODE_REG = 0x90, + ALS4K_GCR91_DMA0_ADDR = 0x91, + ALS4K_GCR92_DMA0_MODE_COUNT = 0x92, + ALS4K_GCR93_DMA1_ADDR = 0x93, + ALS4K_GCR94_DMA1_MODE_COUNT = 0x94, + ALS4K_GCR95_DMA3_ADDR = 0x95, + ALS4K_GCR96_DMA3_MODE_COUNT = 0x96, + ALS4K_GCR99_DMA_EMULATION_CTRL = 0x99, + ALS4K_GCRA0_FIFO1_CURRENT_ADDR = 0xa0, + ALS4K_GCRA1_FIFO1_STATUS_BYTECOUNT = 0xa1, + ALS4K_GCRA2_FIFO2_PCIADDR = 0xa2, + ALS4K_GCRA3_FIFO2_COUNT = 0xa3, + ALS4K_GCRA4_FIFO2_CURRENT_ADDR = 0xa4, + ALS4K_GCRA5_FIFO1_STATUS_BYTECOUNT = 0xa5, + ALS4K_GCRA6_PM_CTRL = 0xa6, + ALS4K_GCRA7_PCI_ACCESS_STORAGE = 0xa7, + ALS4K_GCRA8_LEGACY_CFG1 = 0xa8, + ALS4K_GCRA9_LEGACY_CFG2 = 0xa9, + ALS4K_GCRFF_DUMMY_SCRATCH = 0xff, +}; + +enum als4k_gcr8c_t { + ALS4K_GCR8C_IRQ_MASK_CTRL_ENABLE = 0x8000, + ALS4K_GCR8C_CHIP_REV_MASK = 0xf0000 +}; + +static inline void snd_als4k_iobase_writeb(unsigned long iobase, + enum als4k_iobase_t reg, + u8 val) +{ + outb(val, iobase + reg); +} + +static inline void snd_als4k_iobase_writel(unsigned long iobase, + enum als4k_iobase_t reg, + u32 val) +{ + outl(val, iobase + reg); +} + +static inline u8 snd_als4k_iobase_readb(unsigned long iobase, + enum als4k_iobase_t reg) +{ + return inb(iobase + reg); +} + +static inline u32 snd_als4k_iobase_readl(unsigned long iobase, + enum als4k_iobase_t reg) { - outb(reg, iobase + ALS4K_IOB_0C_GCR_INDEX); - outl(val, iobase + ALS4K_IOD_08_GCR_DATA); + return inl(iobase + reg); } -static inline void snd_als4000_gcr_write(struct snd_sb *sb, +static inline void snd_als4k_gcr_write_addr(unsigned long iobase, enum als4k_gcr_t reg, u32 val) { - snd_als4000_gcr_write_addr(sb->alt_port, reg, val); + snd_als4k_iobase_writeb(iobase, ALS4K_IOB_0C_GCR_INDEX, reg); + snd_als4k_iobase_writel(iobase, ALS4K_IOD_08_GCR_DATA, val); +} + +static inline void snd_als4k_gcr_write(struct snd_sb *sb, + enum als4k_gcr_t reg, + u32 val) +{ + snd_als4k_gcr_write_addr(sb->alt_port, reg, val); } -static inline u32 snd_als4000_gcr_read_addr(unsigned long iobase, +static inline u32 snd_als4k_gcr_read_addr(unsigned long iobase, enum als4k_gcr_t reg) { - outb(reg, iobase + ALS4K_IOB_0C_GCR_INDEX); - return inl(iobase + ALS4K_IOD_08_GCR_DATA); + /* SPECS_PAGE: 37/38 */ + snd_als4k_iobase_writeb(iobase, ALS4K_IOB_0C_GCR_INDEX, reg); + return snd_als4k_iobase_readl(iobase, ALS4K_IOD_08_GCR_DATA); +} + +static inline u32 snd_als4k_gcr_read(struct snd_sb *sb, enum als4k_gcr_t reg) +{ + return snd_als4k_gcr_read_addr(sb->alt_port, reg); +} + +enum als4k_cr_t { /* all registers 8bit wide; SPECS_PAGE: 20 to 23 */ + ALS4K_CR0_SB_CONFIG = 0x00, + ALS4K_CR2_MISC_CONTROL = 0x02, + ALS4K_CR3_CONFIGURATION = 0x03, + ALS4K_CR17_FIFO_STATUS = 0x17, + ALS4K_CR18_ESP_MAJOR_VERSION = 0x18, + ALS4K_CR19_ESP_MINOR_VERSION = 0x19, + ALS4K_CR1A_MPU401_UART_MODE_CONTROL = 0x1a, + ALS4K_CR1C_FIFO2_BLOCK_LENGTH_LO = 0x1c, + ALS4K_CR1D_FIFO2_BLOCK_LENGTH_HI = 0x1d, + ALS4K_CR1E_FIFO2_CONTROL = 0x1e, /* secondary PCM FIFO (recording) */ + ALS4K_CR3A_MISC_CONTROL = 0x3a, + ALS4K_CR3B_CRC32_BYTE0 = 0x3b, /* for testing, activate via CR3A */ + ALS4K_CR3C_CRC32_BYTE1 = 0x3c, + ALS4K_CR3D_CRC32_BYTE2 = 0x3d, + ALS4K_CR3E_CRC32_BYTE3 = 0x3e, +}; + +enum als4k_cr0_t { + ALS4K_CR0_DMA_CONTIN_MODE_CTRL = 0x02, /* IRQ/FIFO controlled for 0/1 */ + ALS4K_CR0_DMA_90H_MODE_CTRL = 0x04, /* IRQ/FIFO controlled for 0/1 */ + ALS4K_CR0_MX80_81_REG_WRITE_ENABLE = 0x80, +}; + +static inline void snd_als4_cr_write(struct snd_sb *chip, + enum als4k_cr_t reg, + u8 data) +{ + /* Control Register is reg | 0xc0 (bit 7, 6 set) on sbmixer_index + * NOTE: assumes chip->mixer_lock to be locked externally already! + * SPECS_PAGE: 6 */ + snd_sbmixer_write(chip, reg | 0xc0, data); } -static inline u32 snd_als4000_gcr_read(struct snd_sb *sb, enum als4k_gcr_t reg) +static inline u8 snd_als4_cr_read(struct snd_sb *chip, + enum als4k_cr_t reg) { - return snd_als4000_gcr_read_addr(sb->alt_port, reg); + /* NOTE: assumes chip->mixer_lock to be locked externally already! */ + return snd_sbmixer_read(chip, reg | 0xc0); } + + static void snd_als4000_set_rate(struct snd_sb *chip, unsigned int rate) { if (!(chip->mode & SB_RATE_LOCK)) { @@ -222,16 +298,18 @@ static inline void snd_als4000_set_capture_dma(struct snd_sb *chip, dma_addr_t addr, unsigned size) { - snd_als4000_gcr_write(chip, ALS4K_GCR_A2_FIFO2_PCIADDR, addr); - snd_als4000_gcr_write(chip, ALS4K_GCR_A3_FIFO2_COUNT, (size-1)); + /* SPECS_PAGE: 40 */ + snd_als4k_gcr_write(chip, ALS4K_GCRA2_FIFO2_PCIADDR, addr); + snd_als4k_gcr_write(chip, ALS4K_GCRA3_FIFO2_COUNT, (size-1)); } static inline void snd_als4000_set_playback_dma(struct snd_sb *chip, dma_addr_t addr, unsigned size) { - snd_als4000_gcr_write(chip, ALS4K_GCR_91_DMA0_ADDR, addr); - snd_als4000_gcr_write(chip, ALS4K_GCR_92_DMA0_MODE_COUNT, + /* SPECS_PAGE: 38 */ + snd_als4k_gcr_write(chip, ALS4K_GCR91_DMA0_ADDR, addr); + snd_als4k_gcr_write(chip, ALS4K_GCR92_DMA0_MODE_COUNT, (size-1)|0x180000); } @@ -316,7 +394,7 @@ count = snd_pcm_lib_period_bytes(substream); if (chip->capture_format & ALS4000_FORMAT_16BIT) - count >>=1; + count >>= 1; count--; spin_lock_irq(&chip->reg_lock); @@ -324,8 +402,8 @@ snd_als4000_set_capture_dma(chip, runtime->dma_addr, size); spin_unlock_irq(&chip->reg_lock); spin_lock_irq(&chip->mixer_lock); - snd_sbmixer_write(chip, 0xdc, count); - snd_sbmixer_write(chip, 0xdd, count>>8); + snd_als4_cr_write(chip, ALS4K_CR1C_FIFO2_BLOCK_LENGTH_LO, count & 0xff); + snd_als4_cr_write(chip, ALS4K_CR1D_FIFO2_BLOCK_LENGTH_HI, count >> 8); spin_unlock_irq(&chip->mixer_lock); return 0; } @@ -343,7 +421,7 @@ count = snd_pcm_lib_period_bytes(substream); if (chip->playback_format & ALS4000_FORMAT_16BIT) - count >>=1; + count >>= 1; count--; /* FIXME: from second playback on, there's a lot more clicks and pops @@ -360,8 +438,8 @@ /* snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); */ snd_sbdsp_command(chip, playback_cmd(chip).dsp_cmd); snd_sbdsp_command(chip, playback_cmd(chip).format); - snd_sbdsp_command(chip, count); - snd_sbdsp_command(chip, count>>8); + snd_sbdsp_command(chip, count & 0xff); + snd_sbdsp_command(chip, count >> 8); snd_sbdsp_command(chip, playback_cmd(chip).dma_off); spin_unlock_irq(&chip->reg_lock); @@ -384,12 +462,14 @@ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: chip->mode |= SB_RATE_LOCK_CAPTURE; - snd_sbmixer_write(chip, 0xde, capture_cmd(chip)); + snd_als4_cr_write(chip, ALS4K_CR1E_FIFO2_CONTROL, + capture_cmd(chip)); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: chip->mode &= ~SB_RATE_LOCK_CAPTURE; - snd_sbmixer_write(chip, 0xde, 0); + snd_als4_cr_write(chip, ALS4K_CR1E_FIFO2_CONTROL, + capture_cmd(chip)); break; default: result = -EINVAL; @@ -430,9 +510,9 @@ unsigned int result; spin_lock(&chip->reg_lock); - result = snd_als4000_gcr_read(chip, ALS4K_GCR_A4_FIFO2_CURRENT_ADDR); - result &= 0xffff; + result = snd_als4k_gcr_read(chip, ALS4K_GCRA4_FIFO2_CURRENT_ADDR); spin_unlock(&chip->reg_lock); + result &= 0xffff; return bytes_to_frames( substream->runtime, result ); } @@ -442,9 +522,9 @@ unsigned result; spin_lock(&chip->reg_lock); - result = snd_als4000_gcr_read(chip, ALS4K_GCR_A0_FIFO1_CURRENT_ADDR); - result &= 0xffff; + result = snd_als4k_gcr_read(chip, ALS4K_GCRA0_FIFO1_CURRENT_ADDR); spin_unlock(&chip->reg_lock); + result &= 0xffff; return bytes_to_frames( substream->runtime, result ); } @@ -452,46 +532,63 @@ * return IRQ_HANDLED no matter whether we actually had an IRQ flag or not). * ALS4000a.PDF writes that while ACKing IRQ in PCI block will *not* ACK * the IRQ in the SB core, ACKing IRQ in SB block *will* ACK the PCI IRQ - * register (alt_port + ALS4K_IOB_0E_SB_MPU_IRQ). Probably something + * register (alt_port + ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU). Probably something * could be optimized here to query/write one register only... * And even if both registers need to be queried, then there's still the * question of whether it's actually correct to ACK PCI IRQ before reading * SB IRQ like we do now, since ALS4000a.PDF mentions that PCI IRQ will *clear* * SB IRQ status. - * (hmm, page 38 mentions it the other way around!) + * (hmm, SPECS_PAGE: 38 mentions it the other way around!) * And do we *really* need the lock here for *reading* SB_DSP4_IRQSTATUS?? * */ static irqreturn_t snd_als4000_interrupt(int irq, void *dev_id) { struct snd_sb *chip = dev_id; - unsigned gcr_status; - unsigned sb_status; - - /* find out which bit of the ALS4000 produced the interrupt */ - gcr_status = inb(chip->alt_port + ALS4K_IOB_0E_SB_MPU_IRQ); + unsigned pci_irqstatus; + unsigned sb_irqstatus; - if ((gcr_status & 0x80) && (chip->playback_substream)) /* playback */ + /* find out which bit of the ALS4000 PCI block produced the interrupt, + SPECS_PAGE: 38, 5 */ + pci_irqstatus = snd_als4k_iobase_readb(chip->alt_port, + ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU); + if ((pci_irqstatus & ALS4K_IOB_0E_SB_DMA_IRQ) + && (chip->playback_substream)) /* playback */ snd_pcm_period_elapsed(chip->playback_substream); - if ((gcr_status & 0x40) && (chip->capture_substream)) /* capturing */ + if ((pci_irqstatus & ALS4K_IOB_0E_CR1E_IRQ) + && (chip->capture_substream)) /* capturing */ snd_pcm_period_elapsed(chip->capture_substream); - if ((gcr_status & 0x10) && (chip->rmidi)) /* MPU401 interrupt */ + if ((pci_irqstatus & ALS4K_IOB_0E_MPU_IRQ) + && (chip->rmidi)) /* MPU401 interrupt */ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); - /* release the gcr */ - outb(gcr_status, chip->alt_port + ALS4K_IOB_0E_SB_MPU_IRQ); + /* ACK the PCI block IRQ */ + snd_als4k_iobase_writeb(chip->alt_port, + ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU, pci_irqstatus); spin_lock(&chip->mixer_lock); - sb_status = snd_sbmixer_read(chip, SB_DSP4_IRQSTATUS); + /* SPECS_PAGE: 20 */ + sb_irqstatus = snd_sbmixer_read(chip, SB_DSP4_IRQSTATUS); spin_unlock(&chip->mixer_lock); - if (sb_status & SB_IRQTYPE_8BIT) + if (sb_irqstatus & SB_IRQTYPE_8BIT) snd_sb_ack_8bit(chip); - if (sb_status & SB_IRQTYPE_16BIT) + if (sb_irqstatus & SB_IRQTYPE_16BIT) snd_sb_ack_16bit(chip); - if (sb_status & SB_IRQTYPE_MPUIN) + if (sb_irqstatus & SB_IRQTYPE_MPUIN) inb(chip->mpu_port); - if (sb_status & 0x20) - inb(SBP(chip, RESET)); - return IRQ_HANDLED; + if (sb_irqstatus & ALS4K_IRQTYPE_CR1E_DMA) + snd_als4k_iobase_readb(chip->alt_port, + ALS4K_IOB_16_ACK_FOR_CR1E); + + /* printk(KERN_INFO "als4000: irq 0x%04x 0x%04x\n", + pci_irqstatus, sb_irqstatus); */ + + /* only ack the things we actually handled above */ + return IRQ_RETVAL( + (pci_irqstatus & (ALS4K_IOB_0E_SB_DMA_IRQ|ALS4K_IOB_0E_CR1E_IRQ| + ALS4K_IOB_0E_MPU_IRQ)) + || (sb_irqstatus & (SB_IRQTYPE_8BIT|SB_IRQTYPE_16BIT| + SB_IRQTYPE_MPUIN|ALS4K_IRQTYPE_CR1E_DMA)) + ); } /*****************************************************************/ @@ -603,7 +700,8 @@ struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "ALS4000 DSP", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(chip->card, "ALS4000 DSP", device, 1, 1, &pcm); + if (err < 0) return err; pcm->private_data = chip; pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -637,34 +735,38 @@ cfg1 |= (game_io | 1) << 16; if (opl_io > 0) cfg1 |= (opl_io | 1); - snd_als4000_gcr_write_addr(iobase, ALS4K_GCR_A8_LEGACY_CFG1, cfg1); - snd_als4000_gcr_write_addr(iobase, ALS4K_GCR_A9_LEGACY_CFG2, cfg2); + snd_als4k_gcr_write_addr(iobase, ALS4K_GCRA8_LEGACY_CFG1, cfg1); + snd_als4k_gcr_write_addr(iobase, ALS4K_GCRA9_LEGACY_CFG2, cfg2); } static void snd_als4000_configure(struct snd_sb *chip) { - unsigned tmp; + u8 tmp; int i; /* do some more configuration */ spin_lock_irq(&chip->mixer_lock); - tmp = snd_sbmixer_read(chip, 0xc0); - snd_sbmixer_write(chip, 0xc0, tmp|0x80); - /* always select DMA channel 0, since we do not actually use DMA */ + tmp = snd_als4_cr_read(chip, ALS4K_CR0_SB_CONFIG); + snd_als4_cr_write(chip, ALS4K_CR0_SB_CONFIG, + tmp|ALS4K_CR0_MX80_81_REG_WRITE_ENABLE); + /* always select DMA channel 0, since we do not actually use DMA + * SPECS_PAGE: 19/20 */ snd_sbmixer_write(chip, SB_DSP4_DMASETUP, SB_DMASETUP_DMA0); - snd_sbmixer_write(chip, 0xc0, tmp&0x7f); + snd_als4_cr_write(chip, ALS4K_CR0_SB_CONFIG, + tmp & ~ALS4K_CR0_MX80_81_REG_WRITE_ENABLE); spin_unlock_irq(&chip->mixer_lock); spin_lock_irq(&chip->reg_lock); /* enable interrupts */ - snd_als4000_gcr_write(chip, ALS4K_GCR_8C_MISC_CTRL, - ALS4K_GCR_8C_IRQ_MASK_CTRL_ENABLE); + snd_als4k_gcr_write(chip, ALS4K_GCR8C_MISC_CTRL, + ALS4K_GCR8C_IRQ_MASK_CTRL_ENABLE); - for (i = ALS4K_GCR_91_DMA0_ADDR; i <= ALS4K_GCR_96_DMA3_MODE_COUNT; ++i) - snd_als4000_gcr_write(chip, i, 0); + /* SPECS_PAGE: 39 */ + for (i = ALS4K_GCR91_DMA0_ADDR; i <= ALS4K_GCR96_DMA3_MODE_COUNT; ++i) + snd_als4k_gcr_write(chip, i, 0); - snd_als4000_gcr_write(chip, ALS4K_GCR_99_DMA_EMULATION_CTRL, - snd_als4000_gcr_read(chip, ALS4K_GCR_99_DMA_EMULATION_CTRL)); + snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL, + snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL)); spin_unlock_irq(&chip->reg_lock); } @@ -739,7 +841,7 @@ struct snd_card_als4000 *acard = card->private_data; /* make sure that interrupts are disabled */ - snd_als4000_gcr_write_addr(acard->iobase, ALS4K_GCR_8C_MISC_CTRL, 0); + snd_als4k_gcr_write_addr(acard->iobase, ALS4K_GCR8C_MISC_CTRL, 0); /* free resources */ snd_als4000_free_gameport(acard); pci_release_regions(acard->pci); @@ -788,7 +890,7 @@ pci_set_master(pci); card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof( struct snd_card_als4000 ) ); + sizeof(*acard) /* private_data: acard */); if (card == NULL) { pci_release_regions(pci); pci_disable_device(pci); @@ -806,6 +908,7 @@ if ((err = snd_sbdsp_create(card, iobase + ALS4K_IOB_10_ADLIB_ADDR0, pci->irq, + /* internally registered as IRQF_SHARED in case of ALS4000 SB */ snd_als4000_interrupt, -1, -1, @@ -835,8 +938,10 @@ goto out_err; } /* FIXME: ALS4000 has interesting MPU401 configuration features - * at CR 0x1A (pass-thru / UART switching, fast MIDI clock, etc.), - * however there doesn't seem to be an ALSA API for this... */ + * at ALS4K_CR1A_MPU401_UART_MODE_CONTROL + * (pass-thru / UART switching, fast MIDI clock, etc.), + * however there doesn't seem to be an ALSA API for this... + * SPECS_PAGE: 21 */ if ((err = snd_als4000_pcm(chip, 0)) < 0) { goto out_err; --- include/sound/sb.h.orig 2008-08-21 21:33:01.000000000 +0200 +++ include/sound/sb.h 2008-08-20 20:22:45.000000000 +0200 @@ -240,11 +240,15 @@ #define SB_DT019X_CAP_MAIN 0x07 #define SB_ALS4000_MONO_IO_CTRL 0x4b +#define SB_ALS4000_OUT_MIXER_CTRL_2 0x4c #define SB_ALS4000_MIC_IN_GAIN 0x4d +#define SB_ALS4000_ANALOG_REFRNC_VOLT_CTRL 0x4e #define SB_ALS4000_FMDAC 0x4f #define SB_ALS4000_3D_SND_FX 0x50 #define SB_ALS4000_3D_TIME_DELAY 0x51 #define SB_ALS4000_3D_AUTO_MUTE 0x52 +#define SB_ALS4000_ANALOG_BLOCK_CTRL 0x53 +#define SB_ALS4000_3D_DELAYLINE_PATTERN 0x54 #define SB_ALS4000_QSOUND 0xdb /* IRQ setting bitmap */ @@ -257,6 +261,7 @@ #define SB_IRQTYPE_8BIT 0x01 #define SB_IRQTYPE_16BIT 0x02 #define SB_IRQTYPE_MPUIN 0x04 +#define ALS4K_IRQTYPE_CR1E_DMA 0x20 /* DMA setting bitmap */ #define SB_DMASETUP_DMA0 0x01

2 1

[alsa-devel] [PATCH] ALSA: snd-ad1816a: add different TerraTec EWS64 S model.
by Rene Herman 22 Aug '08

22 Aug '08

2 1

[alsa-devel] [PATCH - asihpi 7/6] Turn off verbose logging
by linux＠audioscience.com 22 Aug '08

22 Aug '08

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/asihpi.c b/pci/asihpi/asihpi.c index fa0718f..0f49fbb 100644 --- a/pci/asihpi/asihpi.c +++ b/pci/asihpi/asihpi.c @@ -22,7 +22,7 @@ * for any purpose including commercial applications. */ /* >0: print Hw params, timer vars. >1: print stream write/copy sizes */ -#define REALLY_VERBOSE_LOGGING 2 +#define REALLY_VERBOSE_LOGGING 0 #if REALLY_VERBOSE_LOGGING #define VPRINTK1 printk -- 1.5.4.3

1 0

[PATCH] ALSA: snd-ad1816a: add different TerraTec EWS64 S model.
by Rene Herman 22 Aug '08

22 Aug '08

Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/ad1816a/ad1816a.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 68f1260..7752424 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -83,8 +83,10 @@ static struct pnp_card_device_id snd_ad1816a_pnpids[] = { { .id = "MDK1605", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } }, /* Shark Predator ISA - added by Ken Arromdee */ { .id = "SMM7180", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } }, - /* Analog Devices AD1816A - Terratec AudioSystem EWS64S */ + /* Analog Devices AD1816A - Terratec AudioSystem EWS64 S */ { .id = "TER1112", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } }, + /* Analog Devices AD1816A - Terratec AudioSystem EWS64 S */ + { .id = "TER1112", .devs = { { .id = "TER1100" }, { .id = "TER1101" } } }, /* Analog Devices AD1816A - Terratec Base 64 */ { .id = "TER1411", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } }, /* end */ -- 1.5.5

1 0

[alsa-devel] About ipc_key, ipc_gid and ipc_perm
by Simon Tremblay 21 Aug '08

21 Aug '08

Hi, someone knows to what is related ipc_key, ipc_gid and ipc_perm settings in /usr/share/alsa/alsa.conf file? I ask that because even if I don't have an audio group created on my embedded system, I can play a wave file with aplay. But since I want to use -D plug:dmix option, I can't. It asks me to create audio group. But after that, I have a strange error I can't explain: audio open error: Operation not permitted. I think ipc_perm has something related to chmod 0660 file? But what is ipc_key ? Does I have to set it somewhere since I installed manually alsa on my system ? Also, On SuSE, I see a + at the end of permission listing of each device file. What does it mean? Like this: crw-rw----+ 1 dax audio 116, 7 2008-08-13 14:06 controlC0 crw-rw----+ 1 dax audio 116, 3 2008-08-13 14:06 midiC0D0 crw-rw----+ 1 dax audio 116, 6 2008-08-13 14:06 pcmC0D0c crw-rw----+ 1 dax audio 116, 5 2008-08-13 14:06 pcmC0D0p crw-rw----+ 1 dax audio 116, 4 2008-08-13 14:06 pcmC0D1p I don't have that on my embedded system, is it something related only to SuSE ? Thanks, Simon

1 0

[alsa-devel] alsa driver on latest Mac Pro (MacPro3,1)
by S. Aguinaga 21 Aug '08

21 Aug '08

Additionally, I noticed that the results file from 'alsa-info.sh' the section about Module is does now show "macpro" under the model section. Might I be doing something wrong? < snippet > .... !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N .... < snippet > // Salvador // Northwestern Univeristy ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, August 19, 2008 5:48:47 AM Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Mon, 18 Aug 2008 10:26:42 -0700 (PDT), S. Aguinaga wrote: > > ============================================== > Trying to figure out where to add this "model=macpro" > and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ > > but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? You can create any file, typically, /etc/modprobe.d/sound or so. Put the following line there options snd-hda-intel model=macpro Then reload the module, check the sysfs file /sys/modules/snd_hda_intel/parameters/macpro to see whether the specified option was really passed properly. Takashi > below are the only files I have. > > /etc/modprobe.d/ > > anaconda > blacklist-visor > i915modeset > blacklist > floppy-pnp > modprobe.conf.dist > > > ============================================== > Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 > > At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), > S. Aguinaga wrote: > > > > S. Aguinaga wrote: > > > > > > Hello Stan, > > > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > > 2. aplay does not produce sound, period! > > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > > > sudo cat /var/log/messages | grep ALSA > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > (snip) > > These messages are harmless. It's just information from the driver, > enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > > > I think your problem is a misidentification of your sound chip. The > > model being used for your 889A appears to be incompatible, thus all > > the errors above. It appears that the macpro ALC885 is being selected > > before it gets to the choice for the ALC889A. Perhaps a developer can > > look into this. > > Well, ALC882, ALC885 and ALC889A are almost compatible. In the > driver, they are all handled in the same routine. > > There are a few models for intel macs with ALC88x. Try to specify > either model=macpro, model=mbp3 or model=imac24 as the module option > of snd-hda-intel driver. > > > Takashi > > > > >

1 0

[alsa-devel] alsa drivers running under latest Mac Pro hw (MacPro 3, 1, quad core)
by S. Aguinaga 21 Aug '08

21 Aug '08

Stan & Takashi, Still no luck when adding the mode=macpro option to the file: cat /etc/modprobe.d/sound options snd-hda-intel model=macpro The sys module: /sys/module/snd_hda_intel/parameters/model shows: cat model macpro,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> Attached are screen-captures of the volume meter showing no options and the Volume control showing no streams available. // Salvador ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, August 19, 2008 5:48:47 AM Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Mon, 18 Aug 2008 10:26:42 -0700 (PDT), S. Aguinaga wrote: > > ============================================== > Trying to figure out where to add this "model=macpro" > and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ > > but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? You can create any file, typically, /etc/modprobe.d/sound or so. Put the following line there options snd-hda-intel model=macpro Then reload the module, check the sysfs file /sys/modules/snd_hda_intel/parameters/macpro to see whether the specified option was really passed properly. Takashi > below are the only files I have. > > /etc/modprobe.d/ > > anaconda > blacklist-visor > i915modeset > blacklist > floppy-pnp > modprobe.conf.dist > > > ============================================== > Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 > > At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), > S. Aguinaga wrote: > > > > S. Aguinaga wrote: > > > > > > Hello Stan, > > > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > > 2. aplay does not produce sound, period! > > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > > > sudo cat /var/log/messages | grep ALSA > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > (snip) > > These messages are harmless. It's just information from the driver, > enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > > > I think your problem is a misidentification of your sound chip. The > > model being used for your 889A appears to be incompatible, thus all > > the errors above. It appears that the macpro ALC885 is being selected > > before it gets to the choice for the ALC889A. Perhaps a developer can > > look into this. > > Well, ALC882, ALC885 and ALC889A are almost compatible. In the > driver, they are all handled in the same routine. > > There are a few models for intel macs with ALC88x. Try to specify > either model=macpro, model=mbp3 or model=imac24 as the module option > of snd-hda-intel driver. > > > Takashi > > > > >

1 0

Re: [alsa-devel] [snd-bt87x]: seems cannot powerup after powerdown...
by Takashi Iwai 21 Aug '08

21 Aug '08

Added Clemens and alsa-devel to Cc. At Mon, 18 Aug 2008 19:22:20 +0400, Dmitry Butskoy wrote: > > Hi, > > Recently snd-bt87x driver has started to use CTL_A_PWRDN bit, see > http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.26.y.git;a=commi… > > Unfortunately, it produces a regression, at least in my case. > > I have "AVerMedia TVPhone 98" card, lspci is: > > 03:05.0 Multimedia video controller: Brooktree Corporation Bt878 Video > > Capture (rev 11) > > 03:05.1 Multimedia controller: Brooktree Corporation Bt878 Audio > > Capture (rev 11) > My card has no additional chip for digital sound, hence only analog is > available. > > After upgrading from 2.6.23 to 2.6.25, I have discovered no more sound > from my snd-bt87x device. Considering the code changes in the dirver, > I've guessed that it could be some power issues. > > Compiling the newest driver without CTL_A_PWRDN (ie. with that changes > reverted back) I got the sound back again. > > Note, that it is not enough to rmmod the old module and insmod the new. > I have to reboot computer completely with the new driver to take changes > in effect. This is a nasty problem. Reloading the module doesn't fix means that the PCI enable/disable process also doesn't suffice... Do you see any difference in PCI register space between the fresh boot and the silent state? thanks, Takashi > > > I can perform any tests, debug etc. if needed. > > > Regards, > Dmitry Butskoy > http://www.fedoraproject.org/wiki/DmitryButskoy >

2 1

[alsa-devel] [PATCH] revo51: add headphone output
by Alexander Beregalov 21 Aug '08

21 Aug '08

It is ALSA bug #1754. D6 bit in 02 register is PW4. Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/i2c/other/ak4xxx-adda.c | 4 ++-- sound/pci/ice1712/revo.c | 5 +++-- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 288926d..4c94458 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -233,8 +233,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) 0x01, 0x02, /* 1: reset and soft-mute */ 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect, * disable DZF, sharp roll-off, RSTN#=0 */ - 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */ - // 0x02, 0x2e, /* quad speed */ + 0x02, 0x4e, /* 2: DA's power up, normal speed, RSTN#=0 */ + // 0x02, 0x6e, /* quad speed */ 0x03, 0x01, /* 3: de-emphasis off */ 0x04, 0x00, /* 4: LOUT1 volume muted */ 0x05, 0x00, /* 5: ROUT1 volume muted */ diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 4d26314..30b5e83 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -216,6 +216,7 @@ static const struct snd_akm4xxx_dac_channel revo51_dac[] = { AK_DAC("PCM Center Playback Volume", 1), AK_DAC("PCM LFE Playback Volume", 1), AK_DAC("PCM Rear Playback Volume", 2), + AK_DAC("PCM Headphone Volume", 2), }; static const char *revo51_adc_input_names[] = { @@ -279,7 +280,7 @@ static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = { static struct snd_akm4xxx akm_revo51 __devinitdata = { .type = SND_AK4358, - .num_dacs = 6, + .num_dacs = 8, .ops = { .set_rate_val = revo_set_rate_val }, @@ -508,7 +509,7 @@ static int __devinit revo_init(struct snd_ice1712 *ice) ice->gpio.i2s_mclk_changed = revo_i2s_mclk_changed; break; case VT1724_SUBDEVICE_REVOLUTION51: - ice->num_total_dacs = 6; + ice->num_total_dacs = 8; ice->num_total_adcs = 2; break; case VT1724_SUBDEVICE_AUDIOPHILE192:

3 4

[alsa-devel] [PATCH - asihpi 1/3] Fix format in portable way.
by linux＠audioscience.com 21 Aug '08

21 Aug '08

From: Eliot Blennerhassett <eliot(a)zaphod.(none)> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/hpicmn.c b/pci/asihpi/hpicmn.c index 12bc6ad..c56e60d 100644 --- a/pci/asihpi/hpicmn.c +++ b/pci/asihpi/hpicmn.c @@ -288,8 +288,8 @@ short HpiCheckBufferMapping( sizeof(((struct hpi_control_cache_pad *)(NULL))->m) } struct pad_ofs_size { - size_t offset; - size_t field_size; + unsigned int offset; + unsigned int field_size; }; static struct pad_ofs_size aPadDesc[] = { @@ -441,10 +441,10 @@ short HpiCheckControlCache( else { unsigned int index = HPI_CTL_ATTR_INDEX(phm->u.c.wAttribute) - 1; - size_t offset = phm->u.c.dwParam1; - size_t pad_string_len, field_size; + unsigned int offset = phm->u.c.dwParam1; + unsigned int pad_string_len, field_size; char *pad_string; - size_t tocopy; + unsigned int tocopy; HPI_DEBUG_LOG(VERBOSE, "PADS HPI_PADS_ %d\n", phm->u.c.wAttribute); @@ -472,7 +472,7 @@ short HpiCheckControlCache( tocopy = sizeof(phr->u.cu.chars8.szData); HPI_DEBUG_LOG(VERBOSE, - "PADS memcpy(%zd), offset %zd \n", tocopy, + "PADS memcpy(%d), offset %d \n", tocopy, offset); memcpy(phr->u.cu.chars8.szData, &pad_string[offset], tocopy); -- 1.5.4.3

2 3

[alsa-devel] sound-problems with ALC262
by Tobias Kaefer 21 Aug '08

21 Aug '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I posted a message at alsa-user list just yesterday about my problem with the on-board ALC262 on my workstation, see mailings lists archive: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg23236.html Quote from my initial post: | I just got a new workstation. It's an Fujitsu-Siemens Esprimo P5625 | with an on-board ALC262 sound chip. I'm running a Kubuntu 8.04.1 on | that system. | | The sound I get from that chip is really horrible. There are | cracks and some kind of a reverberation in there. | | I compiled the recent version of alsa (1.0.17) and I tried all the | model-options for /etc/modprobe.d/alsa-base. "stan" told me to use this script http://www.alsa-project.org/alsa-info.sh to create more information about my system. Here is the pastebin link of the output of that script: http://pastebin.ca/1169153 Maybe I can help evolve the snd-hda-intel by being a tester. Unfortunately my kernel-hacking skills are as rusty as my C skills - because of being a Java developer for some years now. Greets Tobias - -- Mit freundlichen Gruessen / with kind regards Tobias S. Käfer Tobias S. Kaefer * Portal-Development ABAS Software AG, Suedendstrasse 42, 76135 Karlsruhe, GERMANY Fon: +49 721 967 23 0, Fax: +49 721 967 23 100 Managing board / Vorstand: Werner Strub (CEO, Vorsitzender), Peter Walser Chairman supervisory board / Aufsichtsratsvorsitzender: Thomas Frank Company Register / Handelsregister: HRB 107644 Amtsgericht Mannheim please visit our homepage / Bitte besuchen Sie unsere Homepage: http://www.abas.de -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIonopdJJ1JNEzpd0RAhCIAJ4lvvOwzQAhV3vtA2u7ilwtlBQPlQCgupns 63R7vjvwbY6O3QUEeNwdEO0= =Fxwe -----END PGP SIGNATURE-----

2 2

[alsa-devel] Small problem with VIA8237/ALC655 onboard
by Vaclav Peroutka 21 Aug '08

21 Aug '08

Hello all, I started experimenting with Software Defined Radio and currently using onboard sound VIA8237/ALC655 rev.0 I found funny behaviour of card inputs. Left channel is OK. But right channel is not routed to the connector and is very noisy. If I switch the input mixer to MIC input, both channels show me the same data - right channel is always connected to microphone input.. So it looks that there is a bug in mixer driver. I must confess that I have a little bit older Alsa. But before upgrade, is there any change log or bugfix log for snd-via82xx module ? Or does anybody remember that such bug was solved ? I am glad that sound works now. And I don't want to "destroy" the sound subsystem by upgrading Alsa... Thank you in advance, Vaclav

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[alsa-devel] PATCH - vt1618 7.1 Audio
by John L. Utz III 21 Aug '08

21 Aug '08

Signed-off-by: John L. Utz III john.utz(a)dmx.com diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..a34f1ea 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, // clean sheet of crinkled paper { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..2042415 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * nooding the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3576,6 +3576,413 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } + +/* use these alot in the 1618 code but i cant find a better place to put them */ + +static const char* std_enable[] = {"Enabled", "Disabled"}; +static const char* std_disable[] = {"Disabled","Enabled"}; + +/* disable enable c/lfe exchange */ + +static int snd_ac97_vt1618_clex_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_disable, 2); +} + +static int snd_ac97_vt1618_clex_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5a) & + 0x0100) >> 8; + return 0; +} + +static int snd_ac97_vt1618_clex_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5a, 0x0100, + ucontrol->value.enumerated.item[0] << 8); +} + + +/* disable enable dc offset */ + +static int snd_ac97_vt1618_dcof_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_disable, 2); +} + +static int snd_ac97_vt1618_dcof_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5a) & + 0x0400) >> 10; + return 0; +} + +static int snd_ac97_vt1618_dcof_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5a, 0x0400, + ucontrol->value.enumerated.item[0] << 10); +} + + +/* enable disable headphone amp */ + +static int snd_ac97_vt1618_hamp_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_hamp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0020) >> 5; + return 0; +} + +static int snd_ac97_vt1618_hamp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0020, + ucontrol->value.enumerated.item[0] << 5); +} + + +/* enable disable surround back channel */ + +static int snd_ac97_vt1618_srbk_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_srbk_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0008) >> 3; + return 0; +} + +static int snd_ac97_vt1618_srbk_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* enable disable soft mute */ + +static int snd_ac97_vt1618_smute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_smute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0001); + return 0; +} + +static int snd_ac97_vt1618_smute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0001, + ucontrol->value.enumerated.item[0]); +} + + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_auxin_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = {"Aux In", "Surround Back Out"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 2); +} + +static int snd_ac97_vt1618_auxin_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x76) & + 0x0008) >> 3; + + return 0; +} + +static int snd_ac97_vt1618_auxin_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack, it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get surround output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be + * applicable to them. + */ + +/* copied from ac97_surround_jack_mode_info() ordering in this list + * reflects vt1618 docs for Vendor Defined Register 0x60 */ + +static int snd_ac97_vt1618_UAJ0_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +/* UAJ1 and UAJ2 are not supposed to have 00 written to them?? i + * dunno, because thats something that i have to do to get 5.1 out to + * work. */ + +static int snd_ac97_vt1618_UAJ1_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Surround Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +static int snd_ac97_vt1618_UAJ2_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Center/LowFre Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned short mask) +{ + struct snd_ac97 *pac97; + ushort usDatPag, usUAJ, usShift = 0; // 0 is a possible value + + if(0x000C == mask) usShift = 2; + if(0x0030 == mask) usShift = 4; + + pac97 = snd_kcontrol_chip(kcontrol); + + mutex_lock(&pac97->page_mutex); + + usDatPag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + usUAJ = snd_ac97_read(pac97, 0x60) & mask; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, usDatPag); + mutex_unlock(&pac97->page_mutex); // unlock paging + + ucontrol->value.enumerated.item[0] = usUAJ >> usShift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + unsigned short mask) +{ + struct snd_ac97 *pac97; + ushort usUAJ, usShift = 0; /* 0 is a possible value */ + + if(0x000C == mask) usShift = 2; + if(0x0030 == mask) usShift = 4; + + pac97 = snd_kcontrol_chip(kcontrol); + + usUAJ = ucontrol->value.enumerated.item[0] << usShift; + + return ac97_update_bits_page(pac97, 0x60, mask, usUAJ, 0); +} + +static int snd_ac97_vt1618_UAJ0_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x0003); +} + +static int snd_ac97_vt1618_UAJ1_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x000C); +} + +static int snd_ac97_vt1618_UAJ2_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x0030); +} + +static int snd_ac97_vt1618_UAJ0_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x0003); +} + +static int snd_ac97_vt1618_UAJ1_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x000C); +} + +static int snd_ac97_vt1618_UAJ2_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x0030); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Center LFE Exchange", + .info = snd_ac97_vt1618_clex_info, + .get = snd_ac97_vt1618_clex_get, + .put = snd_ac97_vt1618_clex_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Offset", + .info = snd_ac97_vt1618_dcof_info, + .get = snd_ac97_vt1618_dcof_get, + .put = snd_ac97_vt1618_dcof_put + },diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..a34f1ea 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, // clean sheet of crinkled paper { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..2042415 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * nooding the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3576,6 +3576,413 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } + +/* use these alot in the 1618 code but i cant find a better place to put them */ + +static const char* std_enable[] = {"Enabled", "Disabled"}; +static const char* std_disable[] = {"Disabled","Enabled"}; + +/* disable enable c/lfe exchange */ + +static int snd_ac97_vt1618_clex_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_disable, 2); +} + +static int snd_ac97_vt1618_clex_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5a) & + 0x0100) >> 8; + return 0; +} + +static int snd_ac97_vt1618_clex_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5a, 0x0100, + ucontrol->value.enumerated.item[0] << 8); +} + + +/* disable enable dc offset */ + +static int snd_ac97_vt1618_dcof_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_disable, 2); +} + +static int snd_ac97_vt1618_dcof_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5a) & + 0x0400) >> 10; + return 0; +} + +static int snd_ac97_vt1618_dcof_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5a, 0x0400, + ucontrol->value.enumerated.item[0] << 10); +} + + +/* enable disable headphone amp */ + +static int snd_ac97_vt1618_hamp_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_hamp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0020) >> 5; + return 0; +} + +static int snd_ac97_vt1618_hamp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0020, + ucontrol->value.enumerated.item[0] << 5); +} + + +/* enable disable surround back channel */ + +static int snd_ac97_vt1618_srbk_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_srbk_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0008) >> 3; + return 0; +} + +static int snd_ac97_vt1618_srbk_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* enable disable soft mute */ + +static int snd_ac97_vt1618_smute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_smute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0001); + return 0; +} + +static int snd_ac97_vt1618_smute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0001, + ucontrol->value.enumerated.item[0]); +} + + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_auxin_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = {"Aux In", "Surround Back Out"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 2); +} + +static int snd_ac97_vt1618_auxin_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x76) & + 0x0008) >> 3; + + return 0; +} + +static int snd_ac97_vt1618_auxin_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack, it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get surround output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be + * applicable to them. + */ + +/* copied from ac97_surround_jack_mode_info() ordering in this list + * reflects vt1618 docs for Vendor Defined Register 0x60 */ + +static int snd_ac97_vt1618_UAJ0_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +/* UAJ1 and UAJ2 are not supposed to have 00 written to them?? i + * dunno, because thats something that i have to do to get 5.1 out to + * work. */ + +static int snd_ac97_vt1618_UAJ1_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Surround Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +static int snd_ac97_vt1618_UAJ2_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Center/LowFre Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned short mask) +{ + struct snd_ac97 *pac97; + ushort usDatPag, usUAJ, usShift = 0; // 0 is a possible value + + if(0x000C == mask) usShift = 2; + if(0x0030 == mask) usShift = 4; + + pac97 = snd_kcontrol_chip(kcontrol); + + mutex_lock(&pac97->page_mutex); + + usDatPag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + usUAJ = snd_ac97_read(pac97, 0x60) & mask; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, usDatPag); + mutex_unlock(&pac97->page_mutex); // unlock paging + + ucontrol->value.enumerated.item[0] = usUAJ >> usShift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + unsigned short mask) +{ + struct snd_ac97 *pac97; + ushort usUAJ, usShift = 0; /* 0 is a possible value */ + + if(0x000C == mask) usShift = 2; + if(0x0030 == mask) usShift = 4; + + pac97 = snd_kcontrol_chip(kcontrol); + + usUAJ = ucontrol->value.enumerated.item[0] << usShift; + + return ac97_update_bits_page(pac97, 0x60, mask, usUAJ, 0); +} + +static int snd_ac97_vt1618_UAJ0_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x0003); +} + +static int snd_ac97_vt1618_UAJ1_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x000C); +} + +static int snd_ac97_vt1618_UAJ2_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x0030); +} + +static int snd_ac97_vt1618_UAJ0_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x0003); +} + +static int snd_ac97_vt1618_UAJ1_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x000C); +} + +static int snd_ac97_vt1618_UAJ2_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x0030); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Center LFE Exchange", + .info = snd_ac97_vt1618_clex_info, + .get = snd_ac97_vt1618_clex_get, + .put = snd_ac97_vt1618_clex_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Offset", + .info = snd_ac97_vt1618_dcof_info, + .get = snd_ac97_vt1618_dcof_get, + .put = snd_ac97_vt1618_dcof_diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..a34f1ea 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, // clean sheet of crinkled paper { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..2042415 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * nooding the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3576,6 +3576,413 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } + +/* use these alot in the 1618 code but i cant find a better place to put them */ + +static const char* std_enable[] = {"Enabled", "Disabled"}; +static const char* std_disable[] = {"Disabled","Enabled"}; + +/* disable enable c/lfe exchange */ + +static int snd_ac97_vt1618_clex_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_disable, 2); +} + +static int snd_ac97_vt1618_clex_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5a) & + 0x0100) >> 8; + return 0; +} + +static int snd_ac97_vt1618_clex_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5a, 0x0100, + ucontrol->value.enumerated.item[0] << 8); +} + + +/* disable enable dc offset */ + +static int snd_ac97_vt1618_dcof_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_disable, 2); +} + +static int snd_ac97_vt1618_dcof_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5a) & + 0x0400) >> 10; + return 0; +} + +static int snd_ac97_vt1618_dcof_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5a, 0x0400, + ucontrol->value.enumerated.item[0] << 10); +} + + +/* enable disable headphone amp */ + +static int snd_ac97_vt1618_hamp_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_hamp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0020) >> 5; + return 0; +} + +static int snd_ac97_vt1618_hamp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0020, + ucontrol->value.enumerated.item[0] << 5); +} + + +/* enable disable surround back channel */ + +static int snd_ac97_vt1618_srbk_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_srbk_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0008) >> 3; + return 0; +} + +static int snd_ac97_vt1618_srbk_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* enable disable soft mute */ + +static int snd_ac97_vt1618_smute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, std_enable, 2); +} + +static int snd_ac97_vt1618_smute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & + 0x0001); + return 0; +} + +static int snd_ac97_vt1618_smute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x5c, 0x0001, + ucontrol->value.enumerated.item[0]); +} + + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_auxin_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = {"Aux In", "Surround Back Out"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 2); +} + +static int snd_ac97_vt1618_auxin_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x76) & + 0x0008) >> 3; + + return 0; +} + +static int snd_ac97_vt1618_auxin_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + /* notice relationship between mask and shift! */ + + return snd_ac97_update_bits(pac97, 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack, it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get surround output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be + * applicable to them. + */ + +/* copied from ac97_surround_jack_mode_info() ordering in this list + * reflects vt1618 docs for Vendor Defined Register 0x60 */ + +static int snd_ac97_vt1618_UAJ0_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +/* UAJ1 and UAJ2 are not supposed to have 00 written to them?? i + * dunno, because thats something that i have to do to get 5.1 out to + * work. */ + +static int snd_ac97_vt1618_UAJ1_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Surround Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +static int snd_ac97_vt1618_UAJ2_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static const char* texts[] = { "Center/LowFre Out", "DAC Unmixed Out", "Line In", "Mic In"}; + return ac97_enum_text_info(kcontrol, uinfo, texts, 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned short mask) +{ + struct snd_ac97 *pac97; + ushort usDatPag, usUAJ, usShift = 0; // 0 is a possible value + + if(0x000C == mask) usShift = 2; + if(0x0030 == mask) usShift = 4; + + pac97 = snd_kcontrol_chip(kcontrol); + + mutex_lock(&pac97->page_mutex); + + usDatPag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + usUAJ = snd_ac97_read(pac97, 0x60) & mask; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, usDatPag); + mutex_unlock(&pac97->page_mutex); // unlock paging + + ucontrol->value.enumerated.item[0] = usUAJ >> usShift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + unsigned short mask) +{ + struct snd_ac97 *pac97; + ushort usUAJ, usShift = 0; /* 0 is a possible value */ + + if(0x000C == mask) usShift = 2; + if(0x0030 == mask) usShift = 4; + + pac97 = snd_kcontrol_chip(kcontrol); + + usUAJ = ucontrol->value.enumerated.item[0] << usShift; + + return ac97_update_bits_page(pac97, 0x60, mask, usUAJ, 0); +} + +static int snd_ac97_vt1618_UAJ0_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x0003); +} + +static int snd_ac97_vt1618_UAJ1_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x000C); +} + +static int snd_ac97_vt1618_UAJ2_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_get(kcontrol, ucontrol, 0x0030); +} + +static int snd_ac97_vt1618_UAJ0_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x0003); +} + +static int snd_ac97_vt1618_UAJ1_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x000C); +} + +static int snd_ac97_vt1618_UAJ2_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return snd_ac97_vt1618_UAJ_put(kcontrol, ucontrol, 0x0030); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Center LFE Exchange", + .info = snd_ac97_vt1618_clex_info, + .get = snd_ac97_vt1618_clex_get, + .put = snd_ac97_vt1618_clex_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Offset", + .info = snd_ac97_vt1618_dcof_info, + .get = snd_ac97_vt1618_dcof_get, + .put = snd_ac97_vt1618_dcof_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Amp", + .info = snd_ac97_vt1618_hamp_info, + .get = snd_ac97_vt1618_hamp_get, + .put = snd_ac97_vt1618_hamp_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Surround Back", + .info = snd_ac97_vt1618_srbk_info, + .get = snd_ac97_vt1618_srbk_get, + .put = snd_ac97_vt1618_srbk_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Soft Mute", + .info = snd_ac97_vt1618_smute_info, + .get = snd_ac97_vt1618_smute_get, + .put = snd_ac97_vt1618_smute_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux In Jack", + .info = snd_ac97_vt1618_auxin_info, + .get = snd_ac97_vt1618_auxin_get, + .put = snd_ac97_vt1618_auxin_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack", + .info = snd_ac97_vt1618_UAJ0_info, + .get = snd_ac97_vt1618_UAJ0_get, + .put = snd_ac97_vt1618_UAJ0_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line In Jack", + .info = snd_ac97_vt1618_UAJ1_info, + .get = snd_ac97_vt1618_UAJ1_get, + .put = snd_ac97_vt1618_UAJ1_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic In Jack", + .info = snd_ac97_vt1618_UAJ2_info, + .get = snd_ac97_vt1618_UAJ2_get, + .put = snd_ac97_vt1618_UAJ2_put + } +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97) put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Amp", + .info = snd_ac97_vt1618_hamp_info, + .get = snd_ac97_vt1618_hamp_get, + .put = snd_ac97_vt1618_hamp_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Surround Back", + .info = snd_ac97_vt1618_srbk_info, + .get = snd_ac97_vt1618_srbk_get, + .put = snd_ac97_vt1618_srbk_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Soft Mute", + .info = snd_ac97_vt1618_smute_info, + .get = snd_ac97_vt1618_smute_get, + .put = snd_ac97_vt1618_smute_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux In Jack", + .info = snd_ac97_vt1618_auxin_info, + .get = snd_ac97_vt1618_auxin_get, + .put = snd_ac97_vt1618_auxin_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack", + .info = snd_ac97_vt1618_UAJ0_info, + .get = snd_ac97_vt1618_UAJ0_get, + .put = snd_ac97_vt1618_UAJ0_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line In Jack", + .info = snd_ac97_vt1618_UAJ1_info, + .get = snd_ac97_vt1618_UAJ1_get, + .put = snd_ac97_vt1618_UAJ1_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic In Jack", + .info = snd_ac97_vt1618_UAJ2_info, + .get = snd_ac97_vt1618_UAJ2_get, + .put = snd_ac97_vt1618_UAJ2_put + } +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97) + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Amp", + .info = snd_ac97_vt1618_hamp_info, + .get = snd_ac97_vt1618_hamp_get, + .put = snd_ac97_vt1618_hamp_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Surround Back", + .info = snd_ac97_vt1618_srbk_info, + .get = snd_ac97_vt1618_srbk_get, + .put = snd_ac97_vt1618_srbk_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Soft Mute", + .info = snd_ac97_vt1618_smute_info, + .get = snd_ac97_vt1618_smute_get, + .put = snd_ac97_vt1618_smute_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux In Jack", + .info = snd_ac97_vt1618_auxin_info, + .get = snd_ac97_vt1618_auxin_get, + .put = snd_ac97_vt1618_auxin_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack", + .info = snd_ac97_vt1618_UAJ0_info, + .get = snd_ac97_vt1618_UAJ0_get, + .put = snd_ac97_vt1618_UAJ0_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line In Jack", + .info = snd_ac97_vt1618_UAJ1_info, + .get = snd_ac97_vt1618_UAJ1_get, + .put = snd_ac97_vt1618_UAJ1_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic In Jack", + .info = snd_ac97_vt1618_UAJ2_info, + .get = snd_ac97_vt1618_UAJ2_get, + .put = snd_ac97_vt1618_UAJ2_put + } +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97)

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Re: [alsa-devel] maestro3 & es1988: mic range too low
by Takashi Iwai 20 Aug '08

20 Aug '08

At Mon, 18 Aug 2008 15:28:00 +0300, Ilkka Tengvall wrote: > > >> the reasoning for the original patch sounds reasonable. The > >> mic volume is tunable, only at the very low range. I recall the laptop > >> came with Win98, and there was no problem then with the sound -> not a > >> HW problem. I have never got this laptop mic working with linux, so > >> this is not anything that broke recently. I see there is > >> check_volume_resolution function which tries to guess some of the > >> register ranges, but maybe it get's it wrong with this HW. > > > > Maybe, maybe not. You should check rather ac97 registers in > > /proc/asound/card0/codec97#0/ac97#0-0+regs. If build with > > CONFIG_SND_DEBUG=y, you can write to the register directly, too. > > See Documentaiton[/sound/alsa]/Procfile.txt. > > > > I checked this file. It seems it gets the regs right for the mic > (range 0x1f). The capture range is 0xf. I don't have the specs to > compare it against, so I assume there is nothing to do more here. Which registers, exactly? > It seems mic value decreases while going up to full, where as capture > increases. I guess the former one is the mic loopback volume (attenuation), so no wonder it behaves so. > Another difference is maybe stereo / mono related, the > capture changes both bytes whereas mic only the other one. Also one > thing, should the mic level be around 0x40 - 0x5f, not 0x0 - 0x1f. The bit 0x40 indicates the mic boost +20dB. Takashi

2 3

Re: [alsa-devel] [PATCH v2] duplicate SNDRV_PCM_FMTBIT_S{16,24}_BE
by Takashi Iwai 20 Aug '08

20 Aug '08

At Tue, 19 Aug 2008 11:53:54 -0400, roel kluin wrote: > > Takashi Iwai wrote: > > At Tue, 19 Aug 2008 08:15:05 +0200 (CEST), > > Johannes Berg wrote: > >> roel kluin wrote: > >>> untested, is it correct? > >> not a clue, do you know how long ago that was? :) > >> does the driver check endianness anywhere? > > > > AFAIK snd-aoa supports only bit-endian formats (at least in > > sound/aoa/soundbus/i2sbus-pcm.c), so this addition makes little > > sense. > > > > Better to drop the duplicated words there. > > Thanks Johannes and Takashi, > > FWIW this removes the duplicates. > --- > Remove duplicate assignment of SNDRV_PCM_FMTBIT_S{16,24}_BE bits > > Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> Thanks, applied this one now. Takashi > --- > diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c > index 7a16a33..6c515b2 100644 > --- a/sound/aoa/codecs/snd-aoa-codec-tas.c > +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c > @@ -654,15 +654,13 @@ static struct snd_kcontrol_new bass_control = { > static struct transfer_info tas_transfers[] = { > { > /* input */ > - .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | > - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, > + .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE, > .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, > .transfer_in = 1, > }, > { > /* output */ > - .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | > - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, > + .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE, > .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, > .transfer_in = 0, > }, >

1 0

[alsa-devel] [PATCH] opl3sa2: replace nd_opl3sa2_info_xxx() with snd_wss_info_xxx()
by Krzysztof Helt 20 Aug '08

20 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Replace snd_opl3sa2_info_xxx() with snd_wss_info_xxx(). Drop check of card->private_data which cannot be NULL if card is not NULL (spotted by Rene Herman). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-ref/sound/isa/opl3sa2.c linux-2.6/sound/isa/opl3sa2.c --- linux-ref/sound/isa/opl3sa2.c 2008-08-20 21:19:04.000000000 +0200 +++ linux-2.6/sound/isa/opl3sa2.c 2008-08-20 21:19:29.000000000 +0200 @@ -300,7 +300,7 @@ static irqreturn_t snd_opl3sa2_interrupt struct snd_opl3sa2 *chip; int handled = 0; - if (card == NULL || card->private_data == NULL) + if (card == NULL) return IRQ_NONE; chip = card->private_data; @@ -338,29 +338,18 @@ static irqreturn_t snd_opl3sa2_interrupt #define OPL3SA2_SINGLE(xname, xindex, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opl3sa2_info_single, \ + .info = snd_wss_info_single, \ .get = snd_opl3sa2_get_single, .put = snd_opl3sa2_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } #define OPL3SA2_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .name = xname, .index = xindex, \ - .info = snd_opl3sa2_info_single, \ + .info = snd_wss_info_single, \ .get = snd_opl3sa2_get_single, .put = snd_opl3sa2_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ .tlv = { .p = (xtlv) } } -static int snd_opl3sa2_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - static int snd_opl3sa2_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol); @@ -404,29 +393,18 @@ static int snd_opl3sa2_put_single(struct #define OPL3SA2_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opl3sa2_info_double, \ + .info = snd_wss_info_double, \ .get = snd_opl3sa2_get_double, .put = snd_opl3sa2_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } #define OPL3SA2_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .name = xname, .index = xindex, \ - .info = snd_opl3sa2_info_double, \ + .info = snd_wss_info_double, \ .get = snd_opl3sa2_get_double, .put = snd_opl3sa2_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22), \ .tlv = { .p = (xtlv) } } -static int snd_opl3sa2_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - static int snd_opl3sa2_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol); ---------------------------------------------------------------------- Igrzyska na Dzikim Zachodzie! Sprawdz >> http://link.interia.pl/f1edc

3 2

[alsa-devel] [PATCH] Try to make our poor little ALS4000 driver worthy of its copious public specs (first step)
by Andreas Mohr 20 Aug '08

20 Aug '08

[with not-quite-so-unintentional CCs added] Hi, there you have it: Linux developers are hypocrites ;) Never omitting any opportunity to complain about missing vendor cooperation, but what do they do once people actually do publish ample specs? - yeah, you got that right: NOTHING!! Let's try to change that, at least one bit at a time. ;) ChangeLog: - use specs-derived register name enums instead of open-coded numeric values, for better understanding of things - fix naming confusion ("gcr" was _NOT_ the GCR register stuff, but instead the io _base_ which has multiplexed _access_ to GCR config space, at _sub_ registers 0x08 and 0x0c) - add FIXME comments about a race condition and MPU401 features Oh, and I chose an abbreviated "ALS4K" prefix since I would have gotten finger cramps otherwise. Patch created on 2.6.27-rc1-git4 (no als4000.c changes up to -rc3, verified), pushed through checkpatch.pl, verified on my ALS4000 card. More work to follow... e.g. would be nice to get audio wakeup things to work, since this would probably be the first (if admittedly crappy) sound card/driver combo on Linux to support such features, thus a nice broadening of Linux capabilities. Thanks! Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- sound/pci/als4000.c.org 2008-08-18 21:15:37.000000000 +0200 +++ sound/pci/als4000.c 2008-08-18 22:12:27.000000000 +0200 @@ -60,6 +60,7 @@ * * ToDo: * - Proper shared IRQ handling? + * - by default, don't enable legacy game and use PCI game I/O * - power management? (card can do voice wakeup according to datasheet!!) */ @@ -107,7 +108,7 @@ struct snd_card_als4000 { /* most frequent access first */ - unsigned long gcr; + unsigned long iobase; struct pci_dev *pci; struct snd_sb *chip; #ifdef SUPPORT_JOYSTICK @@ -122,24 +123,89 @@ MODULE_DEVICE_TABLE(pci, snd_als4000_ids); -static inline void snd_als4000_gcr_write_addr(unsigned long port, u32 reg, u32 val) +enum als4k_iobase_t { + /* IOx: B == Byte, W = Word, D = DWord */ + ALS4K_IOD_00_AC97_ACCESS = 0x00, + ALS4K_IOW_04_AC97_READ = 0x04, + ALS4K_IOB_06_AC97_STATUS = 0x06, + ALS4K_IOB_07_IRQSTATUS = 0x07, + ALS4K_IOD_08_GCR_DATA = 0x08, + ALS4K_IOB_0C_GCR_INDEX = 0x0c, + ALS4K_IOB_0E_SB_MPU_IRQ = 0x0e, + ALS4K_IOB_10_ADLIB_ADDR0 = 0x10, + ALS4K_IOB_11_ADLIB_ADDR1 = 0x11, + ALS4K_IOB_12_ADLIB_ADDR2 = 0x12, + ALS4K_IOB_13_ADLIB_ADDR3 = 0x13, + ALS4K_IOB_14_MIXER_INDEX = 0x14, + ALS4K_IOB_15_MIXER_DATA = 0x15, + ALS4K_IOB_16_ESP_RST_PORT = 0x16, + ALS4K_IOB_16_CR1E_ACK_PORT = 0x16, /* 2nd function */ + ALS4K_IOB_18_OPL_ADDR0 = 0x18, + ALS4K_IOB_19_OPL_ADDR1 = 0x19, + ALS4K_IOB_1A_ESP_RD_DATA = 0x1a, + ALS4K_IOB_1C_ESP_CMD_DATA = 0x1c, + ALS4K_IOB_1C_ESP_WR_STATUS = 0x1c, /* 2nd function */ + ALS4K_IOB_1E_ESP_RD_STATUS8 = 0x1e, + ALS4K_IOB_1F_ESP_RD_STATUS16 = 0x1f, + ALS4K_IOB_20_ESP_GAMEPORT_200 = 0x20, + ALS4K_IOB_21_ESP_GAMEPORT_201 = 0x21, + ALS4K_IOB_30_MIDI_DATA = 0x30, + ALS4K_IOB_31_MIDI_STATUS = 0x31, + ALS4K_IOB_31_MIDI_COMMAND = 0x31, /* 2nd function */ +}; + +enum als4k_gcr_t { + /* all registers 32bit wide */ + ALS4K_GCR_8C_MISC_CTRL = 0x8c, + ALS4K_GCR_90_TEST_MODE_REG = 0x90, + ALS4K_GCR_91_DMA0_ADDR = 0x91, + ALS4K_GCR_92_DMA0_MODE_COUNT = 0x92, + ALS4K_GCR_93_DMA1_ADDR = 0x93, + ALS4K_GCR_94_DMA1_MODE_COUNT = 0x94, + ALS4K_GCR_95_DMA3_ADDR = 0x95, + ALS4K_GCR_96_DMA3_MODE_COUNT = 0x96, + ALS4K_GCR_99_DMA_EMULATION_CTRL = 0x99, + ALS4K_GCR_A0_FIFO1_CURRENT_ADDR = 0xa0, + ALS4K_GCR_A1_FIFO1_STATUS_BYTECOUNT = 0xa1, + ALS4K_GCR_A2_FIFO2_PCIADDR = 0xa2, + ALS4K_GCR_A3_FIFO2_COUNT = 0xa3, + ALS4K_GCR_A4_FIFO2_CURRENT_ADDR = 0xa4, + ALS4K_GCR_A5_FIFO1_STATUS_BYTECOUNT = 0xa5, + ALS4K_GCR_A6_PM_CTRL = 0xa6, + ALS4K_GCR_A7_PCI_ACCESS_STORAGE = 0xa7, + ALS4K_GCR_A8_LEGACY_CFG1 = 0xa8, + ALS4K_GCR_A9_LEGACY_CFG2 = 0xa9, + ALS4K_GCR_FF_DUMMY_SCRATCH = 0xff, +}; + +enum als4k_gcr_8c_t { + ALS4K_GCR_8C_IRQ_MASK_CTRL_ENABLE = 0x8000, + ALS4K_GCR_8C_CHIP_REV_MASK = 0xf0000 +}; + +static inline void snd_als4000_gcr_write_addr(unsigned long iobase, + enum als4k_gcr_t reg, + u32 val) { - outb(reg, port+0x0c); - outl(val, port+0x08); + outb(reg, iobase + ALS4K_IOB_0C_GCR_INDEX); + outl(val, iobase + ALS4K_IOD_08_GCR_DATA); } -static inline void snd_als4000_gcr_write(struct snd_sb *sb, u32 reg, u32 val) +static inline void snd_als4000_gcr_write(struct snd_sb *sb, + enum als4k_gcr_t reg, + u32 val) { snd_als4000_gcr_write_addr(sb->alt_port, reg, val); } -static inline u32 snd_als4000_gcr_read_addr(unsigned long port, u32 reg) +static inline u32 snd_als4000_gcr_read_addr(unsigned long iobase, + enum als4k_gcr_t reg) { - outb(reg, port+0x0c); - return inl(port+0x08); + outb(reg, iobase + ALS4K_IOB_0C_GCR_INDEX); + return inl(iobase + ALS4K_IOD_08_GCR_DATA); } -static inline u32 snd_als4000_gcr_read(struct snd_sb *sb, u32 reg) +static inline u32 snd_als4000_gcr_read(struct snd_sb *sb, enum als4k_gcr_t reg) { return snd_als4000_gcr_read_addr(sb->alt_port, reg); } @@ -156,15 +222,17 @@ static inline void snd_als4000_set_capture_dma(struct snd_sb *chip, dma_addr_t addr, unsigned size) { - snd_als4000_gcr_write(chip, 0xa2, addr); - snd_als4000_gcr_write(chip, 0xa3, (size-1)); + snd_als4000_gcr_write(chip, ALS4K_GCR_A2_FIFO2_PCIADDR, addr); + snd_als4000_gcr_write(chip, ALS4K_GCR_A3_FIFO2_COUNT, (size-1)); } static inline void snd_als4000_set_playback_dma(struct snd_sb *chip, - dma_addr_t addr, unsigned size) + dma_addr_t addr, + unsigned size) { - snd_als4000_gcr_write(chip, 0x91, addr); - snd_als4000_gcr_write(chip, 0x92, (size-1)|0x180000); + snd_als4000_gcr_write(chip, ALS4K_GCR_91_DMA0_ADDR, addr); + snd_als4000_gcr_write(chip, ALS4K_GCR_92_DMA0_MODE_COUNT, + (size-1)|0x180000); } #define ALS4000_FORMAT_SIGNED (1<<0) @@ -305,6 +373,12 @@ struct snd_sb *chip = snd_pcm_substream_chip(substream); int result = 0; + /* FIXME race condition in here!!! + chip->mode non-atomic update gets consistently protected + by reg_lock always, _except_ for this place!! + Probably need to take reg_lock as outer (or inner??) lock, too. + (or serialize both lock operations? probably not, though... - racy?) + */ spin_lock(&chip->mixer_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -356,7 +430,8 @@ unsigned int result; spin_lock(&chip->reg_lock); - result = snd_als4000_gcr_read(chip, 0xa4) & 0xffff; + result = snd_als4000_gcr_read(chip, ALS4K_GCR_A4_FIFO2_CURRENT_ADDR); + result &= 0xffff; spin_unlock(&chip->reg_lock); return bytes_to_frames( substream->runtime, result ); } @@ -367,7 +442,8 @@ unsigned result; spin_lock(&chip->reg_lock); - result = snd_als4000_gcr_read(chip, 0xa0) & 0xffff; + result = snd_als4000_gcr_read(chip, ALS4K_GCR_A0_FIFO1_CURRENT_ADDR); + result &= 0xffff; spin_unlock(&chip->reg_lock); return bytes_to_frames( substream->runtime, result ); } @@ -376,12 +452,13 @@ * return IRQ_HANDLED no matter whether we actually had an IRQ flag or not). * ALS4000a.PDF writes that while ACKing IRQ in PCI block will *not* ACK * the IRQ in the SB core, ACKing IRQ in SB block *will* ACK the PCI IRQ - * register (alt_port + 0x0e). Probably something could be optimized here to - * query/write one register only... + * register (alt_port + ALS4K_IOB_0E_SB_MPU_IRQ). Probably something + * could be optimized here to query/write one register only... * And even if both registers need to be queried, then there's still the * question of whether it's actually correct to ACK PCI IRQ before reading * SB IRQ like we do now, since ALS4000a.PDF mentions that PCI IRQ will *clear* * SB IRQ status. + * (hmm, page 38 mentions it the other way around!) * And do we *really* need the lock here for *reading* SB_DSP4_IRQSTATUS?? * */ static irqreturn_t snd_als4000_interrupt(int irq, void *dev_id) @@ -391,7 +468,7 @@ unsigned sb_status; /* find out which bit of the ALS4000 produced the interrupt */ - gcr_status = inb(chip->alt_port + 0xe); + gcr_status = inb(chip->alt_port + ALS4K_IOB_0E_SB_MPU_IRQ); if ((gcr_status & 0x80) && (chip->playback_substream)) /* playback */ snd_pcm_period_elapsed(chip->playback_substream); @@ -400,7 +477,7 @@ if ((gcr_status & 0x10) && (chip->rmidi)) /* MPU401 interrupt */ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); /* release the gcr */ - outb(gcr_status, chip->alt_port + 0xe); + outb(gcr_status, chip->alt_port + ALS4K_IOB_0E_SB_MPU_IRQ); spin_lock(&chip->mixer_lock); sb_status = snd_sbmixer_read(chip, SB_DSP4_IRQSTATUS); @@ -543,25 +620,25 @@ /******************************************************************/ -static void snd_als4000_set_addr(unsigned long gcr, - unsigned int sb, - unsigned int mpu, - unsigned int opl, - unsigned int game) -{ - u32 confA = 0; - u32 confB = 0; - - if (mpu > 0) - confB |= (mpu | 1) << 16; - if (sb > 0) - confB |= (sb | 1); - if (game > 0) - confA |= (game | 1) << 16; - if (opl > 0) - confA |= (opl | 1); - snd_als4000_gcr_write_addr(gcr, 0xa8, confA); - snd_als4000_gcr_write_addr(gcr, 0xa9, confB); +static void snd_als4000_set_addr(unsigned long iobase, + unsigned int sb_io, + unsigned int mpu_io, + unsigned int opl_io, + unsigned int game_io) +{ + u32 cfg1 = 0; + u32 cfg2 = 0; + + if (mpu_io > 0) + cfg2 |= (mpu_io | 1) << 16; + if (sb_io > 0) + cfg2 |= (sb_io | 1); + if (game_io > 0) + cfg1 |= (game_io | 1) << 16; + if (opl_io > 0) + cfg1 |= (opl_io | 1); + snd_als4000_gcr_write_addr(iobase, ALS4K_GCR_A8_LEGACY_CFG1, cfg1); + snd_als4000_gcr_write_addr(iobase, ALS4K_GCR_A9_LEGACY_CFG2, cfg2); } static void snd_als4000_configure(struct snd_sb *chip) @@ -579,12 +656,15 @@ spin_unlock_irq(&chip->mixer_lock); spin_lock_irq(&chip->reg_lock); - /* magic number. Enables interrupts(?) */ - snd_als4000_gcr_write(chip, 0x8c, 0x28000); - for(i = 0x91; i <= 0x96; ++i) + /* enable interrupts */ + snd_als4000_gcr_write(chip, ALS4K_GCR_8C_MISC_CTRL, + ALS4K_GCR_8C_IRQ_MASK_CTRL_ENABLE); + + for (i = ALS4K_GCR_91_DMA0_ADDR; i <= ALS4K_GCR_96_DMA3_MODE_COUNT; ++i) snd_als4000_gcr_write(chip, i, 0); - snd_als4000_gcr_write(chip, 0x99, snd_als4000_gcr_read(chip, 0x99)); + snd_als4000_gcr_write(chip, ALS4K_GCR_99_DMA_EMULATION_CTRL, + snd_als4000_gcr_read(chip, ALS4K_GCR_99_DMA_EMULATION_CTRL)); spin_unlock_irq(&chip->reg_lock); } @@ -628,7 +708,7 @@ gameport_set_port_data(gp, r); /* Enable legacy joystick port */ - snd_als4000_set_addr(acard->gcr, 0, 0, 0, 1); + snd_als4000_set_addr(acard->iobase, 0, 0, 0, 1); gameport_register_port(acard->gameport); @@ -643,7 +723,9 @@ gameport_unregister_port(acard->gameport); acard->gameport = NULL; - snd_als4000_set_addr(acard->gcr, 0, 0, 0, 0); /* disable joystick */ + /* disable joystick */ + snd_als4000_set_addr(acard->iobase, 0, 0, 0, 0); + release_and_free_resource(r); } } @@ -654,10 +736,10 @@ static void snd_card_als4000_free( struct snd_card *card ) { - struct snd_card_als4000 * acard = (struct snd_card_als4000 *)card->private_data; + struct snd_card_als4000 *acard = card->private_data; /* make sure that interrupts are disabled */ - snd_als4000_gcr_write_addr( acard->gcr, 0x8c, 0); + snd_als4000_gcr_write_addr(acard->iobase, ALS4K_GCR_8C_MISC_CTRL, 0); /* free resources */ snd_als4000_free_gameport(acard); pci_release_regions(acard->pci); @@ -670,7 +752,7 @@ static int dev; struct snd_card *card; struct snd_card_als4000 *acard; - unsigned long gcr; + unsigned long iobase; struct snd_sb *chip; struct snd_opl3 *opl3; unsigned short word; @@ -699,7 +781,7 @@ pci_disable_device(pci); return err; } - gcr = pci_resource_start(pci, 0); + iobase = pci_resource_start(pci, 0); pci_read_config_word(pci, PCI_COMMAND, &word); pci_write_config_word(pci, PCI_COMMAND, word | PCI_COMMAND_IO); @@ -713,16 +795,16 @@ return -ENOMEM; } - acard = (struct snd_card_als4000 *)card->private_data; + acard = card->private_data; acard->pci = pci; - acard->gcr = gcr; + acard->iobase = iobase; card->private_free = snd_card_als4000_free; /* disable all legacy ISA stuff */ - snd_als4000_set_addr(acard->gcr, 0, 0, 0, 0); + snd_als4000_set_addr(acard->iobase, 0, 0, 0, 0); if ((err = snd_sbdsp_create(card, - gcr + 0x10, + iobase + ALS4K_IOB_10_ADLIB_ADDR0, pci->irq, snd_als4000_interrupt, -1, @@ -734,7 +816,7 @@ acard->chip = chip; chip->pci = pci; - chip->alt_port = gcr; + chip->alt_port = iobase; snd_card_set_dev(card, &pci->dev); snd_als4000_configure(chip); @@ -745,11 +827,16 @@ card->shortname, chip->alt_port, chip->irq); if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, - gcr+0x30, MPU401_INFO_INTEGRATED, + iobase + ALS4K_IOB_30_MIDI_DATA, + MPU401_INFO_INTEGRATED, pci->irq, 0, &chip->rmidi)) < 0) { - printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", gcr+0x30); + printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", + iobase + ALS4K_IOB_30_MIDI_DATA); goto out_err; } + /* FIXME: ALS4000 has interesting MPU401 configuration features + * at CR 0x1A (pass-thru / UART switching, fast MIDI clock, etc.), + * however there doesn't seem to be an ALSA API for this... */ if ((err = snd_als4000_pcm(chip, 0)) < 0) { goto out_err; @@ -758,10 +845,13 @@ goto out_err; } - if (snd_opl3_create(card, gcr+0x10, gcr+0x12, + if (snd_opl3_create(card, + iobase + ALS4K_IOB_10_ADLIB_ADDR0, + iobase + ALS4K_IOB_12_ADLIB_ADDR2, OPL3_HW_AUTO, 1, &opl3) < 0) { printk(KERN_ERR "als4000: no OPL device at 0x%lx-0x%lx?\n", - gcr+0x10, gcr+0x12 ); + iobase + ALS4K_IOB_10_ADLIB_ADDR0, + iobase + ALS4K_IOB_12_ADLIB_ADDR2); } else { if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { goto out_err; @@ -831,13 +921,13 @@ #ifdef SUPPORT_JOYSTICK if (acard->gameport) - snd_als4000_set_addr(acard->gcr, 0, 0, 0, 1); + snd_als4000_set_addr(acard->iobase, 0, 0, 0, 1); #endif snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif +#endif /* CONFIG_PM */ static struct pci_driver driver = {

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[alsa-devel] pcm write from posix threads
by Brian Rhodes 20 Aug '08

20 Aug '08

When calling snd_pcm_write* from a thread how do you safely cancel the thread? I've tried masking the signals which are used in alsa lib, yet I am getting a segfault on pthread_exit(). My thread is looping doing the following. testcancel snd_pcm_wait snd_pcm_avail snd_pcm_writei I am checking return values to handle those functions being woken up by pthread cancel and attempting to exit. In testing the process it performs fine until I start actually writing data. Once I write some data, a pthread_exit will segfault. My cancellation routine is calling snd_pcm_drop, then doing a cancel followed by join.

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[alsa-devel] ALSA Driver DB Says to Contact You if I have This Hardware... So Here I am.
by Bugs Bane 20 Aug '08

20 Aug '08

Hi there, I have an Edirol 100FX. It has two modes, standard and advanced. The driver DB says it's currently supported for standard mode only, but to get in touch if I have this device so advanced mode can be worked on. I'm wondering what info I can supply to be of help? lsusb reads: Bus 004 Device 002: ID 0582:0046 Roland Corp. Bugsbane

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Re: [alsa-devel] [snd-bt87x]: seems cannot powerup after powerdown...
by Trent Piepho 19 Aug '08

19 Aug '08

On Tue, 19 Aug 2008, Dmitry Butskoy wrote: > Takashi Iwai wrote: > > Added Clemens and alsa-devel to Cc. > > > > At Mon, 18 Aug 2008 19:22:20 +0400, > > Dmitry Butskoy wrote: > > > >> Hi, > >> > >> Recently snd-bt87x driver has started to use CTL_A_PWRDN bit, see > >> http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.26.y.git;a=commi… > >> > >> Unfortunately, it produces a regression, at least in my case. > >> > >> I have "AVerMedia TVPhone 98" card, lspci is: Which PCM device are you using?

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Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45
by S. Aguinaga 19 Aug '08

19 Aug '08

============================================== Trying to figure out where to add this "model=macpro" and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? below are the only files I have. /etc/modprobe.d/ anaconda blacklist-visor i915modeset blacklist floppy-pnp modprobe.conf.dist ============================================== Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), S. Aguinaga wrote: > > S. Aguinaga wrote: > > > > Hello Stan, > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > 2. aplay does not produce sound, period! > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > sudo cat /var/log/messages | grep ALSA > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped (snip) These messages are harmless. It's just information from the driver, enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > I think your problem is a misidentification of your sound chip. The > model being used for your 889A appears to be incompatible, thus all > the errors above. It appears that the macpro ALC885 is being selected > before it gets to the choice for the ALC889A. Perhaps a developer can > look into this. Well, ALC882, ALC885 and ALC889A are almost compatible. In the driver, they are all handled in the same routine. There are a few models for intel macs with ALC88x. Try to specify either model=macpro, model=mbp3 or model=imac24 as the module option of snd-hda-intel driver. Takashi

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[alsa-devel] [PATCH] Remove OSS driver for SGI HAL2 audio device
by Thomas Bogendoerfer 19 Aug '08

19 Aug '08

With the restructering of the indy button handling the old OSS HAL2 driver got broken. Since there is a new ALSA driver for HAL2, the experimental OSS driver is obsolete and will be removed by this patch. Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- sound/oss/Kconfig | 7 - sound/oss/Makefile | 1 - sound/oss/hal2.c | 1558 ---------------------------------------------------- sound/oss/hal2.h | 248 --------- 4 files changed, 0 insertions(+), 1814 deletions(-) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index d4fafb6..cbee9dc 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -24,13 +24,6 @@ config SOUND_VWSND <file:Documentation/sound/oss/vwsnd> for more info on this driver's capabilities. -config SOUND_HAL2 - tristate "SGI HAL2 sound (EXPERIMENTAL)" - depends on SGI_IP22 && EXPERIMENTAL - help - Say Y or M if you have an SGI Indy or Indigo2 system and want to be able to - use its on-board A2 audio system. - config SOUND_AU1550_AC97 tristate "Au1550/Au1200 AC97 Sound" depends on SOC_AU1550 || SOC_AU1200 diff --git a/sound/oss/Makefile b/sound/oss/Makefile index c611514..e0ae4d4 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -10,7 +10,6 @@ obj-$(CONFIG_SOUND_OSS) += sound.o # Please leave it as is, cause the link order is significant ! obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o -obj-$(CONFIG_SOUND_HAL2) += hal2.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o diff --git a/sound/oss/hal2.c b/sound/oss/hal2.c deleted file mode 100644 index a94b9df..0000000 --- a/sound/oss/hal2.c +++ /dev/null @@ -1,1558 +0,0 @@ -/* - * Driver for A2 audio system used in SGI machines - * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis(a)linux-mips.org> - * - * Based on Ulf Carlsson's code. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - * Supported devices: - * /dev/dsp standard dsp device, (mostly) OSS compatible - * /dev/mixer standard mixer device, (mostly) OSS compatible - * - */ -#include <linux/kernel.h> -#include <linux/module.h> -#include <linux/sched.h> -#include <linux/init.h> -#include <linux/slab.h> -#include <linux/poll.h> -#include <linux/interrupt.h> -#include <linux/dma-mapping.h> -#include <linux/sound.h> -#include <linux/soundcard.h> -#include <linux/mutex.h> - - -#include <asm/io.h> -#include <asm/sgi/hpc3.h> -#include <asm/sgi/ip22.h> - -#include "hal2.h" - -#if 0 -#define DEBUG(args...) printk(args) -#else -#define DEBUG(args...) -#endif - -#if 0 -#define DEBUG_MIX(args...) printk(args) -#else -#define DEBUG_MIX(args...) -#endif - -/* - * Before touching these look how it works. It is a bit unusual I know, - * but it helps to keep things simple. This driver is considered complete - * and I won't add any new features although hardware has many cool - * capabilities. - * (Historical note: HAL2 driver was first written by Ulf Carlsson - ALSA - * 0.3 running with 2.2.x kernel. Then ALSA changed completely and it - * seemed easier to me to write OSS driver from scratch - this one. Now - * when ALSA is official part of 2.6 kernel it's time to write ALSA driver - * using (hopefully) final version of ALSA interface) - */ -#define H2_BLOCK_SIZE 1024 -#define H2_ADC_BUFSIZE 8192 -#define H2_DAC_BUFSIZE 16834 - -struct hal2_pbus { - struct hpc3_pbus_dmacregs *pbus; - int pbusnr; - unsigned int ctrl; /* Current state of pbus->pbdma_ctrl */ -}; - -struct hal2_desc { - struct hpc_dma_desc desc; - u32 cnt; /* don't touch, it is also padding */ -}; - -struct hal2_codec { - unsigned char *buffer; - struct hal2_desc *desc; - int desc_count; - int tail, head; /* tail index, head index */ - struct hal2_pbus pbus; - unsigned int format; /* Audio data format */ - int voices; /* mono/stereo */ - unsigned int sample_rate; - unsigned int master; /* Master frequency */ - unsigned short mod; /* MOD value */ - unsigned short inc; /* INC value */ - - wait_queue_head_t dma_wait; - spinlock_t lock; - struct mutex sem; - - int usecount; /* recording and playback are - * independent */ -}; - -#define H2_MIX_OUTPUT_ATT 0 -#define H2_MIX_INPUT_GAIN 1 -#define H2_MIXERS 2 -struct hal2_mixer { - int modcnt; - unsigned int master; - unsigned int volume[H2_MIXERS]; -}; - -struct hal2_card { - int dev_dsp; /* audio device */ - int dev_mixer; /* mixer device */ - int dev_midi; /* midi device */ - - struct hal2_ctl_regs *ctl_regs; /* HAL2 ctl registers */ - struct hal2_aes_regs *aes_regs; /* HAL2 aes registers */ - struct hal2_vol_regs *vol_regs; /* HAL2 vol registers */ - struct hal2_syn_regs *syn_regs; /* HAL2 syn registers */ - - struct hal2_codec dac; - struct hal2_codec adc; - struct hal2_mixer mixer; -}; - -#define H2_INDIRECT_WAIT(regs) while (regs->isr & H2_ISR_TSTATUS); - -#define H2_READ_ADDR(addr) (addr | (1<<7)) -#define H2_WRITE_ADDR(addr) (addr) - -static char *hal2str = "HAL2"; - -/* - * I doubt anyone has a machine with two HAL2 cards. It's possible to - * have two HPC's, so it is probably possible to have two HAL2 cards. - * Try to deal with it, but note that it is not tested. - */ -#define MAXCARDS 2 -static struct hal2_card* hal2_card[MAXCARDS]; - -static const struct { - unsigned char idx:4, avail:1; -} mixtable[SOUND_MIXER_NRDEVICES] = { - [SOUND_MIXER_PCM] = { H2_MIX_OUTPUT_ATT, 1 }, /* voice */ - [SOUND_MIXER_MIC] = { H2_MIX_INPUT_GAIN, 1 }, /* mic */ -}; - -#define H2_SUPPORTED_FORMATS (AFMT_S16_LE | AFMT_S16_BE) - -static inline void hal2_isr_write(struct hal2_card *hal2, u16 val) -{ - hal2->ctl_regs->isr = val; -} - -static inline u16 hal2_isr_look(struct hal2_card *hal2) -{ - return hal2->ctl_regs->isr; -} - -static inline u16 hal2_rev_look(struct hal2_card *hal2) -{ - return hal2->ctl_regs->rev; -} - -#ifdef HAL2_DUMP_REGS -static u16 hal2_i_look16(struct hal2_card *hal2, u16 addr) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - return regs->idr0; -} -#endif - -static u32 hal2_i_look32(struct hal2_card *hal2, u16 addr) -{ - u32 ret; - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - ret = regs->idr0 & 0xffff; - regs->iar = H2_READ_ADDR(addr | 0x1); - H2_INDIRECT_WAIT(regs); - ret |= (regs->idr0 & 0xffff) << 16; - return ret; -} - -static void hal2_i_write16(struct hal2_card *hal2, u16 addr, u16 val) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->idr0 = val; - regs->idr1 = 0; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_write32(struct hal2_card *hal2, u16 addr, u32 val) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->idr0 = val & 0xffff; - regs->idr1 = val >> 16; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_setbit16(struct hal2_card *hal2, u16 addr, u16 bit) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - regs->idr0 = (regs->idr0 & 0xffff) | bit; - regs->idr1 = 0; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_setbit32(struct hal2_card *hal2, u16 addr, u32 bit) -{ - u32 tmp; - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - tmp = (regs->idr0 & 0xffff) | (regs->idr1 << 16) | bit; - regs->idr0 = tmp & 0xffff; - regs->idr1 = tmp >> 16; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_clearbit16(struct hal2_card *hal2, u16 addr, u16 bit) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - regs->idr0 = (regs->idr0 & 0xffff) & ~bit; - regs->idr1 = 0; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -#if 0 -static void hal2_i_clearbit32(struct hal2_card *hal2, u16 addr, u32 bit) -{ - u32 tmp; - hal2_ctl_regs_t *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - tmp = ((regs->idr0 & 0xffff) | (regs->idr1 << 16)) & ~bit; - regs->idr0 = tmp & 0xffff; - regs->idr1 = tmp >> 16; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} -#endif - -#ifdef HAL2_DUMP_REGS -static void hal2_dump_regs(struct hal2_card *hal2) -{ - DEBUG("isr: %08hx ", hal2_isr_look(hal2)); - DEBUG("rev: %08hx\n", hal2_rev_look(hal2)); - DEBUG("relay: %04hx\n", hal2_i_look16(hal2, H2I_RELAY_C)); - DEBUG("port en: %04hx ", hal2_i_look16(hal2, H2I_DMA_PORT_EN)); - DEBUG("dma end: %04hx ", hal2_i_look16(hal2, H2I_DMA_END)); - DEBUG("dma drv: %04hx\n", hal2_i_look16(hal2, H2I_DMA_DRV)); - DEBUG("syn ctl: %04hx ", hal2_i_look16(hal2, H2I_SYNTH_C)); - DEBUG("aesrx ctl: %04hx ", hal2_i_look16(hal2, H2I_AESRX_C)); - DEBUG("aestx ctl: %04hx ", hal2_i_look16(hal2, H2I_AESTX_C)); - DEBUG("dac ctl1: %04hx ", hal2_i_look16(hal2, H2I_ADC_C1)); - DEBUG("dac ctl2: %08x ", hal2_i_look32(hal2, H2I_ADC_C2)); - DEBUG("adc ctl1: %04hx ", hal2_i_look16(hal2, H2I_DAC_C1)); - DEBUG("adc ctl2: %08x ", hal2_i_look32(hal2, H2I_DAC_C2)); - DEBUG("syn map: %04hx\n", hal2_i_look16(hal2, H2I_SYNTH_MAP_C)); - DEBUG("bres1 ctl1: %04hx ", hal2_i_look16(hal2, H2I_BRES1_C1)); - DEBUG("bres1 ctl2: %04x ", hal2_i_look32(hal2, H2I_BRES1_C2)); - DEBUG("bres2 ctl1: %04hx ", hal2_i_look16(hal2, H2I_BRES2_C1)); - DEBUG("bres2 ctl2: %04x ", hal2_i_look32(hal2, H2I_BRES2_C2)); - DEBUG("bres3 ctl1: %04hx ", hal2_i_look16(hal2, H2I_BRES3_C1)); - DEBUG("bres3 ctl2: %04x\n", hal2_i_look32(hal2, H2I_BRES3_C2)); -} -#endif - -static struct hal2_card* hal2_dsp_find_card(int minor) -{ - int i; - - for (i = 0; i < MAXCARDS; i++) - if (hal2_card[i] != NULL && hal2_card[i]->dev_dsp == minor) - return hal2_card[i]; - return NULL; -} - -static struct hal2_card* hal2_mixer_find_card(int minor) -{ - int i; - - for (i = 0; i < MAXCARDS; i++) - if (hal2_card[i] != NULL && hal2_card[i]->dev_mixer == minor) - return hal2_card[i]; - return NULL; -} - -static void hal2_inc_head(struct hal2_codec *codec) -{ - codec->head++; - if (codec->head == codec->desc_count) - codec->head = 0; -} - -static void hal2_inc_tail(struct hal2_codec *codec) -{ - codec->tail++; - if (codec->tail == codec->desc_count) - codec->tail = 0; -} - -static void hal2_dac_interrupt(struct hal2_codec *dac) -{ - int running; - - spin_lock(&dac->lock); - /* if tail buffer contains zero samples DMA stream was already - * stopped */ - running = dac->desc[dac->tail].cnt; - dac->desc[dac->tail].cnt = 0; - dac->desc[dac->tail].desc.cntinfo = HPCDMA_XIE | HPCDMA_EOX; - /* we just proccessed empty buffer, don't update tail pointer */ - if (running) - hal2_inc_tail(dac); - spin_unlock(&dac->lock); - - wake_up(&dac->dma_wait); -} - -static void hal2_adc_interrupt(struct hal2_codec *adc) -{ - int running; - - spin_lock(&adc->lock); - /* if head buffer contains nonzero samples DMA stream was already - * stopped */ - running = !adc->desc[adc->head].cnt; - adc->desc[adc->head].cnt = H2_BLOCK_SIZE; - adc->desc[adc->head].desc.cntinfo = HPCDMA_XIE | HPCDMA_EOR; - /* we just proccessed empty buffer, don't update head pointer */ - if (running) - hal2_inc_head(adc); - spin_unlock(&adc->lock); - - wake_up(&adc->dma_wait); -} - -static irqreturn_t hal2_interrupt(int irq, void *dev_id) -{ - struct hal2_card *hal2 = dev_id; - irqreturn_t ret = IRQ_NONE; - - /* decide what caused this interrupt */ - if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { - hal2_dac_interrupt(&hal2->dac); - ret = IRQ_HANDLED; - } - if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { - hal2_adc_interrupt(&hal2->adc); - ret = IRQ_HANDLED; - } - return ret; -} - -static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate) -{ - unsigned short mod; - - DEBUG("rate: %d\n", rate); - - if (rate < 4000) rate = 4000; - else if (rate > 48000) rate = 48000; - - if (44100 % rate < 48000 % rate) { - mod = 4 * 44100 / rate; - codec->master = 44100; - } else { - mod = 4 * 48000 / rate; - codec->master = 48000; - } - - codec->inc = 4; - codec->mod = mod; - rate = 4 * codec->master / mod; - - DEBUG("real_rate: %d\n", rate); - - return rate; -} - -static void hal2_set_dac_rate(struct hal2_card *hal2) -{ - unsigned int master = hal2->dac.master; - int inc = hal2->dac.inc; - int mod = hal2->dac.mod; - - DEBUG("master: %d inc: %d mod: %d\n", master, inc, mod); - - hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0); - hal2_i_write32(hal2, H2I_BRES1_C2, ((0xffff & (inc - mod - 1)) << 16) | inc); -} - -static void hal2_set_adc_rate(struct hal2_card *hal2) -{ - unsigned int master = hal2->adc.master; - int inc = hal2->adc.inc; - int mod = hal2->adc.mod; - - DEBUG("master: %d inc: %d mod: %d\n", master, inc, mod); - - hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0); - hal2_i_write32(hal2, H2I_BRES2_C2, ((0xffff & (inc - mod - 1)) << 16) | inc); -} - -static void hal2_setup_dac(struct hal2_card *hal2) -{ - unsigned int fifobeg, fifoend, highwater, sample_size; - struct hal2_pbus *pbus = &hal2->dac.pbus; - - DEBUG("hal2_setup_dac\n"); - - /* Now we set up some PBUS information. The PBUS needs information about - * what portion of the fifo it will use. If it's receiving or - * transmitting, and finally whether the stream is little endian or big - * endian. The information is written later, on the start call. - */ - sample_size = 2 * hal2->dac.voices; - /* Fifo should be set to hold exactly four samples. Highwater mark - * should be set to two samples. */ - highwater = (sample_size * 2) >> 1; /* halfwords */ - fifobeg = 0; /* playback is first */ - fifoend = (sample_size * 4) >> 3; /* doublewords */ - pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD | - (highwater << 8) | (fifobeg << 16) | (fifoend << 24) | - (hal2->dac.format & AFMT_S16_LE ? HPC3_PDMACTRL_SEL : 0); - /* We disable everything before we do anything at all */ - pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; - hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); - /* Setup the HAL2 for playback */ - hal2_set_dac_rate(hal2); - /* Set endianess */ - if (hal2->dac.format & AFMT_S16_LE) - hal2_i_setbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); - else - hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); - /* Set DMA bus */ - hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); - /* We are using 1st Bresenham clock generator for playback */ - hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) - | (1 << H2I_C1_CLKID_SHIFT) - | (hal2->dac.voices << H2I_C1_DATAT_SHIFT)); -} - -static void hal2_setup_adc(struct hal2_card *hal2) -{ - unsigned int fifobeg, fifoend, highwater, sample_size; - struct hal2_pbus *pbus = &hal2->adc.pbus; - - DEBUG("hal2_setup_adc\n"); - - sample_size = 2 * hal2->adc.voices; - highwater = (sample_size * 2) >> 1; /* halfwords */ - fifobeg = (4 * 4) >> 3; /* record is second */ - fifoend = (4 * 4 + sample_size * 4) >> 3; /* doublewords */ - pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD | - (highwater << 8) | (fifobeg << 16) | (fifoend << 24) | - (hal2->adc.format & AFMT_S16_LE ? HPC3_PDMACTRL_SEL : 0); - pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; - hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); - /* Setup the HAL2 for record */ - hal2_set_adc_rate(hal2); - /* Set endianess */ - if (hal2->adc.format & AFMT_S16_LE) - hal2_i_setbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); - else - hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); - /* Set DMA bus */ - hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); - /* We are using 2nd Bresenham clock generator for record */ - hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) - | (2 << H2I_C1_CLKID_SHIFT) - | (hal2->adc.voices << H2I_C1_DATAT_SHIFT)); -} - -static dma_addr_t hal2_desc_addr(struct hal2_codec *codec, int i) -{ - if (--i < 0) - i = codec->desc_count - 1; - return codec->desc[i].desc.pnext; -} - -static void hal2_start_dac(struct hal2_card *hal2) -{ - struct hal2_codec *dac = &hal2->dac; - struct hal2_pbus *pbus = &dac->pbus; - - pbus->pbus->pbdma_dptr = hal2_desc_addr(dac, dac->tail); - pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; - /* enable DAC */ - hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); -} - -static void hal2_start_adc(struct hal2_card *hal2) -{ - struct hal2_codec *adc = &hal2->adc; - struct hal2_pbus *pbus = &adc->pbus; - - pbus->pbus->pbdma_dptr = hal2_desc_addr(adc, adc->head); - pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; - /* enable ADC */ - hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); -} - -static inline void hal2_stop_dac(struct hal2_card *hal2) -{ - hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; - /* The HAL2 itself may remain enabled safely */ -} - -static inline void hal2_stop_adc(struct hal2_card *hal2) -{ - hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; -} - -static int hal2_alloc_dmabuf(struct hal2_codec *codec, int size, - int count, int cntinfo, int dir) -{ - struct hal2_desc *desc, *dma_addr; - int i; - - DEBUG("allocating %dk DMA buffer.\n", size / 1024); - - codec->buffer = (unsigned char *)__get_free_pages(GFP_KERNEL | GFP_DMA, - get_order(size)); - if (!codec->buffer) - return -ENOMEM; - desc = dma_alloc_coherent(NULL, count * sizeof(struct hal2_desc), - (dma_addr_t *)&dma_addr, GFP_KERNEL); - if (!desc) { - free_pages((unsigned long)codec->buffer, get_order(size)); - return -ENOMEM; - } - codec->desc = desc; - for (i = 0; i < count; i++) { - desc->desc.pbuf = dma_map_single(NULL, - (void *)(codec->buffer + i * H2_BLOCK_SIZE), - H2_BLOCK_SIZE, dir); - desc->desc.cntinfo = cntinfo; - desc->desc.pnext = (i == count - 1) ? - (u32)dma_addr : (u32)(dma_addr + i + 1); - desc->cnt = 0; - desc++; - } - codec->desc_count = count; - codec->head = codec->tail = 0; - return 0; -} - -static int hal2_alloc_dac_dmabuf(struct hal2_codec *codec) -{ - return hal2_alloc_dmabuf(codec, H2_DAC_BUFSIZE, - H2_DAC_BUFSIZE / H2_BLOCK_SIZE, - HPCDMA_XIE | HPCDMA_EOX, - DMA_TO_DEVICE); -} - -static int hal2_alloc_adc_dmabuf(struct hal2_codec *codec) -{ - return hal2_alloc_dmabuf(codec, H2_ADC_BUFSIZE, - H2_ADC_BUFSIZE / H2_BLOCK_SIZE, - HPCDMA_XIE | H2_BLOCK_SIZE, - DMA_TO_DEVICE); -} - -static void hal2_free_dmabuf(struct hal2_codec *codec, int size, int dir) -{ - dma_addr_t dma_addr; - int i; - - dma_addr = codec->desc[codec->desc_count - 1].desc.pnext; - for (i = 0; i < codec->desc_count; i++) - dma_unmap_single(NULL, codec->desc[i].desc.pbuf, - H2_BLOCK_SIZE, dir); - dma_free_coherent(NULL, codec->desc_count * sizeof(struct hal2_desc), - (void *)codec->desc, dma_addr); - free_pages((unsigned long)codec->buffer, get_order(size)); -} - -static void hal2_free_dac_dmabuf(struct hal2_codec *codec) -{ - return hal2_free_dmabuf(codec, H2_DAC_BUFSIZE, DMA_TO_DEVICE); -} - -static void hal2_free_adc_dmabuf(struct hal2_codec *codec) -{ - return hal2_free_dmabuf(codec, H2_ADC_BUFSIZE, DMA_FROM_DEVICE); -} - -/* - * Add 'count' bytes to 'buffer' from DMA ring buffers. Return number of - * bytes added or -EFAULT if copy_from_user failed. - */ -static int hal2_get_buffer(struct hal2_card *hal2, char *buffer, int count) -{ - unsigned long flags; - int size, ret = 0; - unsigned char *buf; - struct hal2_desc *tail; - struct hal2_codec *adc = &hal2->adc; - - DEBUG("getting %d bytes ", count); - - spin_lock_irqsave(&adc->lock, flags); - tail = &adc->desc[adc->tail]; - /* enable DMA stream if there are no data */ - if (!tail->cnt && !(adc->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT)) - hal2_start_adc(hal2); - while (tail->cnt > 0 && count > 0) { - size = min((int)tail->cnt, count); - buf = &adc->buffer[(adc->tail + 1) * H2_BLOCK_SIZE - tail->cnt]; - spin_unlock_irqrestore(&adc->lock, flags); - dma_sync_single(NULL, tail->desc.pbuf, size, DMA_FROM_DEVICE); - if (copy_to_user(buffer, buf, size)) { - ret = -EFAULT; - goto out; - } - spin_lock_irqsave(&adc->lock, flags); - tail->cnt -= size; - /* buffer is empty, update tail pointer */ - if (tail->cnt == 0) { - tail->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE; - hal2_inc_tail(adc); - tail = &adc->desc[adc->tail]; - /* enable DMA stream again if needed */ - if (!(adc->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT)) - hal2_start_adc(hal2); - } - buffer += size; - ret += size; - count -= size; - - DEBUG("(%d) ", size); - } - spin_unlock_irqrestore(&adc->lock, flags); -out: - DEBUG("\n"); - - return ret; -} - -/* - * Add 'count' bytes from 'buffer' to DMA ring buffers. Return number of - * bytes added or -EFAULT if copy_from_user failed. - */ -static int hal2_add_buffer(struct hal2_card *hal2, char *buffer, int count) -{ - unsigned long flags; - unsigned char *buf; - int size, ret = 0; - struct hal2_desc *head; - struct hal2_codec *dac = &hal2->dac; - - DEBUG("adding %d bytes ", count); - - spin_lock_irqsave(&dac->lock, flags); - head = &dac->desc[dac->head]; - while (head->cnt == 0 && count > 0) { - size = min((int)H2_BLOCK_SIZE, count); - buf = &dac->buffer[dac->head * H2_BLOCK_SIZE]; - spin_unlock_irqrestore(&dac->lock, flags); - if (copy_from_user(buf, buffer, size)) { - ret = -EFAULT; - goto out; - } - dma_sync_single(NULL, head->desc.pbuf, size, DMA_TO_DEVICE); - spin_lock_irqsave(&dac->lock, flags); - head->desc.cntinfo = size | HPCDMA_XIE; - head->cnt = size; - buffer += size; - ret += size; - count -= size; - hal2_inc_head(dac); - head = &dac->desc[dac->head]; - - DEBUG("(%d) ", size); - } - if (!(dac->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT) && ret > 0) - hal2_start_dac(hal2); - spin_unlock_irqrestore(&dac->lock, flags); -out: - DEBUG("\n"); - - return ret; -} - -#define hal2_reset_dac_pointer(hal2) hal2_reset_pointer(hal2, 1) -#define hal2_reset_adc_pointer(hal2) hal2_reset_pointer(hal2, 0) -static void hal2_reset_pointer(struct hal2_card *hal2, int is_dac) -{ - int i; - struct hal2_codec *codec = (is_dac) ? &hal2->dac : &hal2->adc; - - DEBUG("hal2_reset_pointer\n"); - - for (i = 0; i < codec->desc_count; i++) { - codec->desc[i].cnt = 0; - codec->desc[i].desc.cntinfo = HPCDMA_XIE | (is_dac) ? - HPCDMA_EOX : H2_BLOCK_SIZE; - } - codec->head = codec->tail = 0; -} - -static int hal2_sync_dac(struct hal2_card *hal2) -{ - DECLARE_WAITQUEUE(wait, current); - struct hal2_codec *dac = &hal2->dac; - int ret = 0; - unsigned long flags; - signed long timeout = 1000 * H2_BLOCK_SIZE * 2 * dac->voices * - HZ / dac->sample_rate / 900; - - while (dac->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT) { - add_wait_queue(&dac->dma_wait, &wait); - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(timeout); - spin_lock_irqsave(&dac->lock, flags); - if (dac->desc[dac->tail].cnt) - ret = -ETIME; - spin_unlock_irqrestore(&dac->lock, flags); - if (signal_pending(current)) - ret = -ERESTARTSYS; - if (ret) { - hal2_stop_dac(hal2); - hal2_reset_dac_pointer(hal2); - } - remove_wait_queue(&dac->dma_wait, &wait); - } - - return ret; -} - -static int hal2_write_mixer(struct hal2_card *hal2, int index, int vol) -{ - unsigned int l, r, tmp; - - DEBUG_MIX("mixer %d write\n", index); - - if (index >= SOUND_MIXER_NRDEVICES || !mixtable[index].avail) - return -EINVAL; - - r = (vol >> 8) & 0xff; - if (r > 100) - r = 100; - l = vol & 0xff; - if (l > 100) - l = 100; - - hal2->mixer.volume[mixtable[index].idx] = l | (r << 8); - - switch (mixtable[index].idx) { - case H2_MIX_OUTPUT_ATT: - - DEBUG_MIX("output attenuator %d,%d\n", l, r); - - if (r | l) { - tmp = hal2_i_look32(hal2, H2I_DAC_C2); - tmp &= ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); - - /* Attenuator has five bits */ - l = 31 * (100 - l) / 99; - r = 31 * (100 - r) / 99; - - DEBUG_MIX("left: %d, right %d\n", l, r); - - tmp |= (l << H2I_C2_L_ATT_SHIFT) & H2I_C2_L_ATT_M; - tmp |= (r << H2I_C2_R_ATT_SHIFT) & H2I_C2_R_ATT_M; - hal2_i_write32(hal2, H2I_DAC_C2, tmp); - } else - hal2_i_setbit32(hal2, H2I_DAC_C2, H2I_C2_MUTE); - break; - case H2_MIX_INPUT_GAIN: - - DEBUG_MIX("input gain %d,%d\n", l, r); - - tmp = hal2_i_look32(hal2, H2I_ADC_C2); - tmp &= ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M); - - /* Gain control has four bits */ - l = 16 * l / 100; - r = 16 * r / 100; - - DEBUG_MIX("left: %d, right %d\n", l, r); - - tmp |= (l << H2I_C2_L_GAIN_SHIFT) & H2I_C2_L_GAIN_M; - tmp |= (r << H2I_C2_R_GAIN_SHIFT) & H2I_C2_R_GAIN_M; - hal2_i_write32(hal2, H2I_ADC_C2, tmp); - - break; - } - - return 0; -} - -static void hal2_init_mixer(struct hal2_card *hal2) -{ - int i; - - for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) - if (mixtable[i].avail) - hal2->mixer.volume[mixtable[i].idx] = 100 | (100 << 8); - - /* disable attenuator */ - hal2_i_write32(hal2, H2I_DAC_C2, 0); - /* set max input gain */ - hal2_i_write32(hal2, H2I_ADC_C2, H2I_C2_MUTE | - (H2I_C2_L_GAIN_M << H2I_C2_L_GAIN_SHIFT) | - (H2I_C2_R_GAIN_M << H2I_C2_R_GAIN_SHIFT)); - /* set max volume */ - hal2->mixer.master = 0xff; - hal2->vol_regs->left = 0xff; - hal2->vol_regs->right = 0xff; -} - -/* - * XXX: later i'll implement mixer for main volume which will be disabled - * by default. enabling it users will be allowed to have master volume level - * control on panel in their favourite X desktop - */ -static void hal2_volume_control(int direction) -{ - unsigned int master = hal2_card[0]->mixer.master; - struct hal2_vol_regs *vol = hal2_card[0]->vol_regs; - - /* volume up */ - if (direction > 0 && master < 0xff) - master++; - /* volume down */ - else if (direction < 0 && master > 0) - master--; - /* TODO: mute/unmute */ - vol->left = master; - vol->right = master; - hal2_card[0]->mixer.master = master; -} - -static int hal2_mixer_ioctl(struct hal2_card *hal2, unsigned int cmd, - unsigned long arg) -{ - int val; - - if (cmd == SOUND_MIXER_INFO) { - mixer_info info; - - memset(&info, 0, sizeof(info)); - strlcpy(info.id, hal2str, sizeof(info.id)); - strlcpy(info.name, hal2str, sizeof(info.name)); - info.modify_counter = hal2->mixer.modcnt; - if (copy_to_user((void *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - if (cmd == SOUND_OLD_MIXER_INFO) { - _old_mixer_info info; - - memset(&info, 0, sizeof(info)); - strlcpy(info.id, hal2str, sizeof(info.id)); - strlcpy(info.name, hal2str, sizeof(info.name)); - if (copy_to_user((void *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - if (cmd == OSS_GETVERSION) - return put_user(SOUND_VERSION, (int *)arg); - - if (_IOC_TYPE(cmd) != 'M' || _IOC_SIZE(cmd) != sizeof(int)) - return -EINVAL; - - if (_IOC_DIR(cmd) == _IOC_READ) { - switch (_IOC_NR(cmd)) { - /* Give the current record source */ - case SOUND_MIXER_RECSRC: - val = 0; /* FIXME */ - break; - /* Give the supported mixers, all of them support stereo */ - case SOUND_MIXER_DEVMASK: - case SOUND_MIXER_STEREODEVS: { - int i; - - for (val = i = 0; i < SOUND_MIXER_NRDEVICES; i++) - if (mixtable[i].avail) - val |= 1 << i; - break; - } - /* Arg contains a bit for each supported recording source */ - case SOUND_MIXER_RECMASK: - val = 0; - break; - case SOUND_MIXER_CAPS: - val = 0; - break; - /* Read a specific mixer */ - default: { - int i = _IOC_NR(cmd); - - if (i >= SOUND_MIXER_NRDEVICES || !mixtable[i].avail) - return -EINVAL; - val = hal2->mixer.volume[mixtable[i].idx]; - break; - } - } - return put_user(val, (int *)arg); - } - - if (_IOC_DIR(cmd) != (_IOC_WRITE|_IOC_READ)) - return -EINVAL; - - hal2->mixer.modcnt++; - - if (get_user(val, (int *)arg)) - return -EFAULT; - - switch (_IOC_NR(cmd)) { - /* Arg contains a bit for each recording source */ - case SOUND_MIXER_RECSRC: - return 0; /* FIXME */ - default: - return hal2_write_mixer(hal2, _IOC_NR(cmd), val); - } - - return 0; -} - -static int hal2_open_mixdev(struct inode *inode, struct file *file) -{ - struct hal2_card *hal2 = hal2_mixer_find_card(iminor(inode)); - - if (hal2) { - file->private_data = hal2; - return nonseekable_open(inode, file); - } - return -ENODEV; -} - -static int hal2_release_mixdev(struct inode *inode, struct file *file) -{ - return 0; -} - -static int hal2_ioctl_mixdev(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - return hal2_mixer_ioctl((struct hal2_card *)file->private_data, cmd, arg); -} - -static int hal2_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - int val; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *)arg); - - case SNDCTL_DSP_SYNC: - if (file->f_mode & FMODE_WRITE) - return hal2_sync_dac(hal2); - return 0; - - case SNDCTL_DSP_SETDUPLEX: - return 0; - - case SNDCTL_DSP_GETCAPS: - return put_user(DSP_CAP_DUPLEX | DSP_CAP_MULTI, (int *)arg); - - case SNDCTL_DSP_RESET: - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2_reset_adc_pointer(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2_reset_dac_pointer(hal2); - } - return 0; - - case SNDCTL_DSP_SPEED: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - val = hal2_compute_rate(&hal2->adc, val); - hal2->adc.sample_rate = val; - hal2_set_adc_rate(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - val = hal2_compute_rate(&hal2->dac, val); - hal2->dac.sample_rate = val; - hal2_set_dac_rate(hal2); - } - return put_user(val, (int *)arg); - - case SNDCTL_DSP_STEREO: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2->adc.voices = (val) ? 2 : 1; - hal2_setup_adc(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2->dac.voices = (val) ? 2 : 1; - hal2_setup_dac(hal2); - } - return 0; - - case SNDCTL_DSP_CHANNELS: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val != 0) { - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2->adc.voices = (val == 1) ? 1 : 2; - hal2_setup_adc(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2->dac.voices = (val == 1) ? 1 : 2; - hal2_setup_dac(hal2); - } - } - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.voices; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.voices; - return put_user(val, (int *)arg); - - case SNDCTL_DSP_GETFMTS: /* Returns a mask */ - return put_user(H2_SUPPORTED_FORMATS, (int *)arg); - - case SNDCTL_DSP_SETFMT: /* Selects ONE fmt*/ - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val != AFMT_QUERY) { - if (!(val & H2_SUPPORTED_FORMATS)) - return -EINVAL; - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2->adc.format = val; - hal2_setup_adc(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2->dac.format = val; - hal2_setup_dac(hal2); - } - } else { - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.format; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.format; - } - return put_user(val, (int *)arg); - - case SNDCTL_DSP_POST: - return 0; - - case SNDCTL_DSP_GETOSPACE: { - audio_buf_info info; - int i; - unsigned long flags; - struct hal2_codec *dac = &hal2->dac; - - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - info.fragments = 0; - spin_lock_irqsave(&dac->lock, flags); - for (i = 0; i < dac->desc_count; i++) - if (dac->desc[i].cnt == 0) - info.fragments++; - spin_unlock_irqrestore(&dac->lock, flags); - info.fragstotal = dac->desc_count; - info.fragsize = H2_BLOCK_SIZE; - info.bytes = info.fragsize * info.fragments; - - return copy_to_user((void *)arg, &info, sizeof(info)) ? -EFAULT : 0; - } - - case SNDCTL_DSP_GETISPACE: { - audio_buf_info info; - int i; - unsigned long flags; - struct hal2_codec *adc = &hal2->adc; - - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - info.fragments = 0; - info.bytes = 0; - spin_lock_irqsave(&adc->lock, flags); - for (i = 0; i < adc->desc_count; i++) - if (adc->desc[i].cnt > 0) { - info.fragments++; - info.bytes += adc->desc[i].cnt; - } - spin_unlock_irqrestore(&adc->lock, flags); - info.fragstotal = adc->desc_count; - info.fragsize = H2_BLOCK_SIZE; - - return copy_to_user((void *)arg, &info, sizeof(info)) ? -EFAULT : 0; - } - - case SNDCTL_DSP_NONBLOCK: - file->f_flags |= O_NONBLOCK; - return 0; - - case SNDCTL_DSP_GETBLKSIZE: - return put_user(H2_BLOCK_SIZE, (int *)arg); - - case SNDCTL_DSP_SETFRAGMENT: - return 0; - - case SOUND_PCM_READ_RATE: - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.sample_rate; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.sample_rate; - return put_user(val, (int *)arg); - - case SOUND_PCM_READ_CHANNELS: - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.voices; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.voices; - return put_user(val, (int *)arg); - - case SOUND_PCM_READ_BITS: - return put_user(16, (int *)arg); - } - - return hal2_mixer_ioctl(hal2, cmd, arg); -} - -static ssize_t hal2_read(struct file *file, char *buffer, - size_t count, loff_t *ppos) -{ - ssize_t err; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - struct hal2_codec *adc = &hal2->adc; - - if (!count) - return 0; - if (mutex_lock_interruptible(&adc->sem)) - return -EINTR; - if (file->f_flags & O_NONBLOCK) { - err = hal2_get_buffer(hal2, buffer, count); - err = err == 0 ? -EAGAIN : err; - } else { - do { - /* ~10% longer */ - signed long timeout = 1000 * H2_BLOCK_SIZE * - 2 * adc->voices * HZ / adc->sample_rate / 900; - unsigned long flags; - DECLARE_WAITQUEUE(wait, current); - ssize_t cnt = 0; - - err = hal2_get_buffer(hal2, buffer, count); - if (err > 0) { - count -= err; - cnt += err; - buffer += err; - err = cnt; - } - if (count > 0 && err >= 0) { - add_wait_queue(&adc->dma_wait, &wait); - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(timeout); - spin_lock_irqsave(&adc->lock, flags); - if (!adc->desc[adc->tail].cnt) - err = -EAGAIN; - spin_unlock_irqrestore(&adc->lock, flags); - if (signal_pending(current)) - err = -ERESTARTSYS; - remove_wait_queue(&adc->dma_wait, &wait); - if (err < 0) { - hal2_stop_adc(hal2); - hal2_reset_adc_pointer(hal2); - } - } - } while (count > 0 && err >= 0); - } - mutex_unlock(&adc->sem); - - return err; -} - -static ssize_t hal2_write(struct file *file, const char *buffer, - size_t count, loff_t *ppos) -{ - ssize_t err; - char *buf = (char*) buffer; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - struct hal2_codec *dac = &hal2->dac; - - if (!count) - return 0; - if (mutex_lock_interruptible(&dac->sem)) - return -EINTR; - if (file->f_flags & O_NONBLOCK) { - err = hal2_add_buffer(hal2, buf, count); - err = err == 0 ? -EAGAIN : err; - } else { - do { - /* ~10% longer */ - signed long timeout = 1000 * H2_BLOCK_SIZE * - 2 * dac->voices * HZ / dac->sample_rate / 900; - unsigned long flags; - DECLARE_WAITQUEUE(wait, current); - ssize_t cnt = 0; - - err = hal2_add_buffer(hal2, buf, count); - if (err > 0) { - count -= err; - cnt += err; - buf += err; - err = cnt; - } - if (count > 0 && err >= 0) { - add_wait_queue(&dac->dma_wait, &wait); - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(timeout); - spin_lock_irqsave(&dac->lock, flags); - if (dac->desc[dac->head].cnt) - err = -EAGAIN; - spin_unlock_irqrestore(&dac->lock, flags); - if (signal_pending(current)) - err = -ERESTARTSYS; - remove_wait_queue(&dac->dma_wait, &wait); - if (err < 0) { - hal2_stop_dac(hal2); - hal2_reset_dac_pointer(hal2); - } - } - } while (count > 0 && err >= 0); - } - mutex_unlock(&dac->sem); - - return err; -} - -static unsigned int hal2_poll(struct file *file, struct poll_table_struct *wait) -{ - unsigned long flags; - unsigned int mask = 0; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - - if (file->f_mode & FMODE_READ) { - struct hal2_codec *adc = &hal2->adc; - - poll_wait(file, &adc->dma_wait, wait); - spin_lock_irqsave(&adc->lock, flags); - if (adc->desc[adc->tail].cnt > 0) - mask |= POLLIN; - spin_unlock_irqrestore(&adc->lock, flags); - } - - if (file->f_mode & FMODE_WRITE) { - struct hal2_codec *dac = &hal2->dac; - - poll_wait(file, &dac->dma_wait, wait); - spin_lock_irqsave(&dac->lock, flags); - if (dac->desc[dac->head].cnt == 0) - mask |= POLLOUT; - spin_unlock_irqrestore(&dac->lock, flags); - } - - return mask; -} - -static int hal2_open(struct inode *inode, struct file *file) -{ - int err; - struct hal2_card *hal2 = hal2_dsp_find_card(iminor(inode)); - - if (!hal2) - return -ENODEV; - file->private_data = hal2; - if (file->f_mode & FMODE_READ) { - struct hal2_codec *adc = &hal2->adc; - - if (adc->usecount) - return -EBUSY; - /* OSS spec wanted us to use 8 bit, 8 kHz mono by default, - * but HAL2 can't do 8bit audio */ - adc->format = AFMT_S16_BE; - adc->voices = 1; - adc->sample_rate = hal2_compute_rate(adc, 8000); - hal2_set_adc_rate(hal2); - err = hal2_alloc_adc_dmabuf(adc); - if (err) - return err; - hal2_setup_adc(hal2); - adc->usecount++; - } - if (file->f_mode & FMODE_WRITE) { - struct hal2_codec *dac = &hal2->dac; - - if (dac->usecount) - return -EBUSY; - dac->format = AFMT_S16_BE; - dac->voices = 1; - dac->sample_rate = hal2_compute_rate(dac, 8000); - hal2_set_dac_rate(hal2); - err = hal2_alloc_dac_dmabuf(dac); - if (err) - return err; - hal2_setup_dac(hal2); - dac->usecount++; - } - - return nonseekable_open(inode, file); -} - -static int hal2_release(struct inode *inode, struct file *file) -{ - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - - if (file->f_mode & FMODE_READ) { - struct hal2_codec *adc = &hal2->adc; - - mutex_lock(&adc->sem); - hal2_stop_adc(hal2); - hal2_free_adc_dmabuf(adc); - adc->usecount--; - mutex_unlock(&adc->sem); - } - if (file->f_mode & FMODE_WRITE) { - struct hal2_codec *dac = &hal2->dac; - - mutex_lock(&dac->sem); - hal2_sync_dac(hal2); - hal2_free_dac_dmabuf(dac); - dac->usecount--; - mutex_unlock(&dac->sem); - } - - return 0; -} - -static const struct file_operations hal2_audio_fops = { - .owner = THIS_MODULE, - .llseek = no_llseek, - .read = hal2_read, - .write = hal2_write, - .poll = hal2_poll, - .ioctl = hal2_ioctl, - .open = hal2_open, - .release = hal2_release, -}; - -static const struct file_operations hal2_mixer_fops = { - .owner = THIS_MODULE, - .llseek = no_llseek, - .ioctl = hal2_ioctl_mixdev, - .open = hal2_open_mixdev, - .release = hal2_release_mixdev, -}; - -static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3, - int index) -{ - codec->pbus.pbusnr = index; - codec->pbus.pbus = &hpc3->pbdma[index]; - init_waitqueue_head(&codec->dma_wait); - mutex_init(&codec->sem); - spin_lock_init(&codec->lock); -} - -static int hal2_detect(struct hal2_card *hal2) -{ - unsigned short board, major, minor; - unsigned short rev; - - /* reset HAL2 */ - hal2_isr_write(hal2, 0); - /* release reset */ - hal2_isr_write(hal2, H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N); - - hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE); - if ((rev = hal2_rev_look(hal2)) & H2_REV_AUDIO_PRESENT) - return -ENODEV; - - board = (rev & H2_REV_BOARD_M) >> 12; - major = (rev & H2_REV_MAJOR_CHIP_M) >> 4; - minor = (rev & H2_REV_MINOR_CHIP_M); - - printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n", - board, major, minor); - - return 0; -} - -static int hal2_init_card(struct hal2_card **phal2, struct hpc3_regs *hpc3) -{ - int ret = 0; - struct hal2_card *hal2; - - hal2 = kzalloc(sizeof(struct hal2_card), GFP_KERNEL); - if (!hal2) - return -ENOMEM; - - hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0]; - hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1]; - hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2]; - hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3]; - - if (hal2_detect(hal2) < 0) { - ret = -ENODEV; - goto free_card; - } - - hal2_init_codec(&hal2->dac, hpc3, 0); - hal2_init_codec(&hal2->adc, hpc3, 1); - - /* - * All DMA channel interfaces in HAL2 are designed to operate with - * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles - * in D5. HAL2 is a 16-bit device which can accept both big and little - * endian format. It assumes that even address bytes are on high - * portion of PBUS (15:8) and assumes that HPC3 is programmed to - * accept a live (unsynchronized) version of P_DREQ_N from HAL2. - */ -#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \ - (2 << HPC3_DMACFG_D4R_SHIFT) | \ - (2 << HPC3_DMACFG_D5R_SHIFT) | \ - (0 << HPC3_DMACFG_D3W_SHIFT) | \ - (2 << HPC3_DMACFG_D4W_SHIFT) | \ - (2 << HPC3_DMACFG_D5W_SHIFT) | \ - HPC3_DMACFG_DS16 | \ - HPC3_DMACFG_EVENHI | \ - HPC3_DMACFG_RTIME | \ - (8 << HPC3_DMACFG_BURST_SHIFT) | \ - HPC3_DMACFG_DRQLIVE) - /* - * Ignore what's mentioned in the specification and write value which - * works in The Real World (TM) - */ - hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844; - hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844; - - if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED, - hal2str, hal2)) { - printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ); - ret = -EAGAIN; - goto free_card; - } - - hal2->dev_dsp = register_sound_dsp(&hal2_audio_fops, -1); - if (hal2->dev_dsp < 0) { - ret = hal2->dev_dsp; - goto free_irq; - } - - hal2->dev_mixer = register_sound_mixer(&hal2_mixer_fops, -1); - if (hal2->dev_mixer < 0) { - ret = hal2->dev_mixer; - goto unregister_dsp; - } - - hal2_init_mixer(hal2); - - *phal2 = hal2; - return 0; -unregister_dsp: - unregister_sound_dsp(hal2->dev_dsp); -free_irq: - free_irq(SGI_HPCDMA_IRQ, hal2); -free_card: - kfree(hal2); - - return ret; -} - -extern void (*indy_volume_button)(int); - -/* - * Assuming only one HAL2 card. Mail me if you ever meet machine with - * more than one. - */ -static int __init init_hal2(void) -{ - int i, error; - - for (i = 0; i < MAXCARDS; i++) - hal2_card[i] = NULL; - - error = hal2_init_card(&hal2_card[0], hpc3c0); - - /* let Indy's volume buttons work */ - if (!error && !ip22_is_fullhouse()) - indy_volume_button = hal2_volume_control; - - return error; - -} - -static void __exit exit_hal2(void) -{ - int i; - - /* unregister volume butons callback function */ - indy_volume_button = NULL; - - for (i = 0; i < MAXCARDS; i++) - if (hal2_card[i]) { - free_irq(SGI_HPCDMA_IRQ, hal2_card[i]); - unregister_sound_dsp(hal2_card[i]->dev_dsp); - unregister_sound_mixer(hal2_card[i]->dev_mixer); - kfree(hal2_card[i]); - } -} - -module_init(init_hal2); -module_exit(exit_hal2); - -MODULE_DESCRIPTION("OSS compatible driver for SGI HAL2 audio"); -MODULE_AUTHOR("Ladislav Michl"); -MODULE_LICENSE("GPL"); diff --git a/sound/oss/hal2.h b/sound/oss/hal2.h deleted file mode 100644 index 2bd3b52..0000000 --- a/sound/oss/hal2.h +++ /dev/null @@ -1,248 +0,0 @@ -#ifndef __HAL2_H -#define __HAL2_H - -/* - * Driver for HAL2 sound processors - * Copyright (c) 1999 Ulf Carlsson <ulfc(a)bun.falkenberg.se> - * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis(a)linux-mips.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#include <asm/addrspace.h> -#include <asm/sgi/hpc3.h> -#include <linux/spinlock.h> -#include <linux/types.h> - -/* Indirect status register */ - -#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */ -#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */ -#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */ -#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */ -#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */ - -/* Revision register */ - -#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */ -#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */ -#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */ -#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */ - -/* Indirect address register */ - -/* - * Address of indirect internal register to be accessed. A write to this - * register initiates read or write access to the indirect registers in the - * HAL2. Note that there af four indirect data registers for write access to - * registers larger than 16 byte. - */ - -#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */ - /* block the register resides in */ - /* 1=DMA Port */ - /* 9=Global DMA Control */ - /* 2=Bresenham */ - /* 3=Unix Timer */ -#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */ - /* blockin which the indirect */ - /* register resides */ - /* If IAR_TYPE_M=DMA Port: */ - /* 1=Synth In */ - /* 2=AES In */ - /* 3=AES Out */ - /* 4=DAC Out */ - /* 5=ADC Out */ - /* 6=Synth Control */ - /* If IAR_TYPE_M=Global DMA Control: */ - /* 1=Control */ - /* If IAR_TYPE_M=Bresenham: */ - /* 1=Bresenham Clock Gen 1 */ - /* 2=Bresenham Clock Gen 2 */ - /* 3=Bresenham Clock Gen 3 */ - /* If IAR_TYPE_M=Unix Timer: */ - /* 1=Unix Timer */ -#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */ -#define H2_IAR_PARAM 0x000C /* Parameter Select */ -#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */ - /* 00:word0 */ - /* 01:word1 */ - /* 10:word2 */ - /* 11:word3 */ -/* - * HAL2 internal addressing - * - * The HAL2 has "indirect registers" (idr) which are accessed by writing to the - * Indirect Data registers. Write the address to the Indirect Address register - * to transfer the data. - * - * We define the H2IR_* to the read address and H2IW_* to the write address and - * H2I_* to be fields in whatever register is referred to. - * - * When we write to indirect registers which are larger than one word (16 bit) - * we have to fill more than one indirect register before writing. When we read - * back however we have to read several times, each time with different Read - * Back Indexes (there are defs for doing this easily). - */ - -/* - * Relay Control - */ -#define H2I_RELAY_C 0x9100 -#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */ - -/* DMA port enable */ - -#define H2I_DMA_PORT_EN 0x9104 -#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */ -#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */ -#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */ -#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */ -#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */ - -#define H2I_DMA_END 0x9108 /* global dma endian select */ -#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */ -#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */ -#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */ -#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */ -#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */ - /* 0=b_end 1=l_end */ - -#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */ - -#define H2I_SYNTH_C 0x1104 /* Synth DMA control */ - -#define H2I_AESRX_C 0x1204 /* AES RX dma control */ - -#define H2I_C_TS_EN 0x20 /* Timestamp enable */ -#define H2I_C_TS_FRMT 0x40 /* Timestamp format */ -#define H2I_C_NAUDIO 0x80 /* Sign extend */ - -/* AESRX CTL, 16 bit */ - -#define H2I_AESTX_C 0x1304 /* AES TX DMA control */ -#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ -#define H2I_AESTX_C_CLKID_M 0x18 -#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ -#define H2I_AESTX_C_DATAT_M 0x300 - -/* CODEC registers */ - -#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */ -#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */ -#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */ -#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */ - -/* Bits in CTL1 register */ - -#define H2I_C1_DMA_SHIFT 0 /* DMA channel */ -#define H2I_C1_DMA_M 0x7 -#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ -#define H2I_C1_CLKID_M 0x18 -#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ -#define H2I_C1_DATAT_M 0x300 - -/* Bits in CTL2 register */ - -#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */ -#define H2I_C2_R_GAIN_M 0xf -#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */ -#define H2I_C2_L_GAIN_M 0xf0 -#define H2I_C2_R_SEL 0x100 /* right input select */ -#define H2I_C2_L_SEL 0x200 /* left input select */ -#define H2I_C2_MUTE 0x400 /* mute */ -#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */ -#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */ -#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */ -#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */ -#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */ -#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */ - -#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */ - -/* Clock generator CTL 1, 16 bit */ - -#define H2I_BRES1_C1 0x2104 -#define H2I_BRES2_C1 0x2204 -#define H2I_BRES3_C1 0x2304 - -#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */ -#define H2I_BRES_C1_M 0x03 - -/* Clock generator CTL 2, 32 bit */ - -#define H2I_BRES1_C2 0x2108 -#define H2I_BRES2_C2 0x2208 -#define H2I_BRES3_C2 0x2308 - -#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */ -#define H2I_BRES_C2_INC_M 0xffff -#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */ -#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */ - -/* Unix timer, 64 bit */ - -#define H2I_UTIME 0x3104 -#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */ -#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */ -#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */ -#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */ -#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */ - -struct hal2_ctl_regs { - u32 _unused0[4]; - volatile u32 isr; /* 0x10 Status Register */ - u32 _unused1[3]; - volatile u32 rev; /* 0x20 Revision Register */ - u32 _unused2[3]; - volatile u32 iar; /* 0x30 Indirect Address Register */ - u32 _unused3[3]; - volatile u32 idr0; /* 0x40 Indirect Data Register 0 */ - u32 _unused4[3]; - volatile u32 idr1; /* 0x50 Indirect Data Register 1 */ - u32 _unused5[3]; - volatile u32 idr2; /* 0x60 Indirect Data Register 2 */ - u32 _unused6[3]; - volatile u32 idr3; /* 0x70 Indirect Data Register 3 */ -}; - -struct hal2_aes_regs { - volatile u32 rx_stat[2]; /* Status registers */ - volatile u32 rx_cr[2]; /* Control registers */ - volatile u32 rx_ud[4]; /* User data window */ - volatile u32 rx_st[24]; /* Channel status data */ - - volatile u32 tx_stat[1]; /* Status register */ - volatile u32 tx_cr[3]; /* Control registers */ - volatile u32 tx_ud[4]; /* User data window */ - volatile u32 tx_st[24]; /* Channel status data */ -}; - -struct hal2_vol_regs { - volatile u32 right; /* Right volume */ - volatile u32 left; /* Left volume */ -}; - -struct hal2_syn_regs { - u32 _unused0[2]; - volatile u32 page; /* DOC Page register */ - volatile u32 regsel; /* DOC Register selection */ - volatile u32 dlow; /* DOC Data low */ - volatile u32 dhigh; /* DOC Data high */ - volatile u32 irq; /* IRQ Status */ - volatile u32 dram; /* DRAM Access */ -}; - -#endif /* __HAL2_H */

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Re: [alsa-devel] [PATCH] duplicate SNDRV_PCM_FMTBIT_S{16,24}_BE
by Johannes Berg 19 Aug '08

19 Aug '08

roel kluin wrote: > untested, is it correct? not a clue, do you know how long ago that was? :) does the driver check endianness anywhere? > duplicate SNDRV_PCM_FMTBIT_S{16,24}_BE > > Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> > --- > diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c > b/sound/aoa/codecs/snd-aoa-codec-tas.c > index 7a16a33..c922505 100644 > --- a/sound/aoa/codecs/snd-aoa-codec-tas.c > +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c > @@ -654,15 +654,15 @@ static struct snd_kcontrol_new bass_control = { > static struct transfer_info tas_transfers[] = { > { > /* input */ > - .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | > - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | > + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE, > .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | > SNDRV_PCM_RATE_48000, > .transfer_in = 1, > }, > { > /* output */ > - .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | > - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | > + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE, > .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | > SNDRV_PCM_RATE_48000, > .transfer_in = 0, > }, > >

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[alsa-devel] 1.0.18rc1 release. alsa-lib 1.0.17a
by Jaroslav Kysela 19 Aug '08

19 Aug '08

Hi all, I released alsa-driver, alsa-utils, and alsa-plugins packages version 1.0.18rc1. The alsa-lib tree had a few short bugfix commits - so stable alsa-lib 1.0.17a is available for download now. For changes see: http://www.alsa-project.org/main/index.php/Changes_v1.0.17_v1.0.17a http://www.alsa-project.org/main/index.php/Changes_v1.0.17a_v1.0.18rc1 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] alsa git: which version of ac97_codec.c am i supposed to hxr?
by John L. Utz III 18 Aug '08

18 Aug '08

Hello All; I feel confused by the presence of 2 different copies of ac97_codec.c: jutz-gnto ~ # find . -name ac97_codec.c ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c worse yet,they are not in sync: jutz-gnto ~ # diff -w ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c 1d0 < #include "adriver.h" 38c37 < #include "../../alsa-kernel/pci/ac97/ac97_id.h" --- > #include "ac97_id.h" 40c39 < #include "../../alsa-kernel/pci/ac97/ac97_patch.c" --- > #include "ac97_patch.c" 1917d1915 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 0) 1929d1926 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 14) 1931d1927 < #endif 1953,1956d1948 < #else /* <= 2.6.0 */ < #define snd_ac97_dev_register NULL < #define snd_ac97_dev_disconnect NULL < #endif Which of these am i supposed to use? am i supposed to change both of them for some backward compatibility purpose? tnx! johnu

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[alsa-devel] alsactl init - next round
by Jaroslav Kysela 18 Aug '08

18 Aug '08

Hello, The 'alsactl init' command becomes more powerful now. - support for dB - better support for enumerated controls - better "ready to use" udev support, expected use: RUN="/usr/sbin/alsactl -r /var/run/alsactl.init restore $name" - initialization using a guess method - the init/default configuration mostly copied from the alsaconf script, see http://git.alsa-project.org/?p=alsa-utils.git;a=blob_plain;f=alsactl/init/d… The cardinfo database is slowly filling. We have over 120 files there now. Any volunteer for watching of these files and writting a suitable configuration files for 'alsactl init' here? I'm not sure about the initialization procedure. I selected the method - look to database and if not found - use the default "guess" method (without any warranty). But the opposite way is also possible - do a universal initialization and modify only special parameters for specific hardware. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

On Mon, Aug 18, 2008 at 1:20 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 17 Aug 2008 22:07:19 +0300, > > Note that "Mic Playback Volume" controls only the volume of analog > playback. It doesn't change the recording level at all. thanks for reminding, sure. >> the reasoning for the original patch sounds reasonable. The >> mic volume is tunable, only at the very low range. I recall the laptop >> came with Win98, and there was no problem then with the sound -> not a >> HW problem. I have never got this laptop mic working with linux, so >> this is not anything that broke recently. I see there is >> check_volume_resolution function which tries to guess some of the >> register ranges, but maybe it get's it wrong with this HW. > > Maybe, maybe not. You should check rather ac97 registers in > /proc/asound/card0/codec97#0/ac97#0-0+regs. If build with > CONFIG_SND_DEBUG=y, you can write to the register directly, too. > See Documentaiton[/sound/alsa]/Procfile.txt. > I checked this file. It seems it gets the regs right for the mic (range 0x1f). The capture range is 0xf. I don't have the specs to compare it against, so I assume there is nothing to do more here. It seems mic value decreases while going up to full, where as capture increases. Another difference is maybe stereo / mono related, the capture changes both bytes whereas mic only the other one. Also one thing, should the mic level be around 0x40 - 0x5f, not 0x0 - 0x1f. Here's the diff of the registers between mixing the capture and mic from min to max: $ diff /tmp/ac97_regs_capt_{min,full} 15c15 < 0:1c = 0000 --- > 0:1c = 0f0f $ diff /tmp/ac97_regs_mic_{min,full} 8c8 < 0:0e = 005f --- > 0:0e = 0040 I'll recompile it with sound debug just to test it... Thanks, ikke

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Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45
by S. Aguinaga 18 Aug '08

18 Aug '08

Stan, Thank you for spending time to look at this issue. I owe you a drink, when you are in the chicago area let me know and I'll treat you to one. Ok, so I will check with the developers an see what can be done to correct this. Thank you. //Salvador 1. Re: latest Mac Pro (quad core) audio issues (stan) Message: 1 Date: Fri, 15 Aug 2008 15:14:12 -0700 From: stan <ghjeold_i_mwee(a)cox.net> Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues To: "S. Aguinaga" <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Message-ID: <48A5FFB4.10700(a)cox.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed S. Aguinaga wrote: > > Hello Stan, > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > 2. aplay does not produce sound, period! > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > sudo cat /var/log/messages | grep ALSA > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped > > Let me know if I should send something else. > > // Salvador > // Northwestern University > I think your problem is a misidentification of your sound chip. The model being used for your 889A appears to be incompatible, thus all the errors above. It appears that the macpro ALC885 is being selected before it gets to the choice for the ALC889A. Perhaps a developer can look into this. !!HDA-Intel Codec information !!--------------------------- Codec: Realtek ALC889A !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 One shows ALC889A and the other shows ALC885. In the driver code there is a comment to the effect that the 889A is best handled as an ALC888, yet the macpro is listed under the ALC885. I am not a developer so can't go beyond this, but it seems that there is a problem. There should be a macpro under the ALC888 as well as the 885. Here are the model options for the two candidates ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 targa Targa T8, MSI-1049 T8 asus-a7j ASUS A7J asus-a7m ASUS A7M macpro MacPro support mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) ALC883/888 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O 3stack-6ch 3-jack 6-channel 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) acer-aspire Acer Aspire 9810 medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targs/MSI with 2-channel laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) _From the code, here is the comment indicating it should be treated as an ALC888: if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { /* Pick up systems that don't supply PCI SSID */ switch (codec->subsystem_id) { case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: /* ALC889A is handled better as ALC888-compatible */ if (codec->revision_id == 0x100103) { alc_free(codec); return patch_alc883(codec); } printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); board_config = ALC882_AUTO; } } ------------------------------ Message: 2 Date: Fri, 15 Aug 2008 16:15:45 -0700 From: "John L. Utz III" <john.utz(a)dmx.com> Subject: [alsa-devel] alsa git: which version of ac97_codec.c am i supposed to hxr? To: "ALSA Developers" <alsa-devel(a)alsa-project.org> Message-ID: <op.ufyawjke647oa9@jutz-gnto64> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes Hello All; I feel confused by the presence of 2 different copies of ac97_codec.c: jutz-gnto ~ # find . -name ac97_codec.c ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c worse yet,they are not in sync: jutz-gnto ~ # diff -w ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c 1d0 < #include "adriver.h" 38c37 < #include "../../alsa-kernel/pci/ac97/ac97_id.h" --- > #include "ac97_id.h" 40c39 < #include "../../alsa-kernel/pci/ac97/ac97_patch.c" --- > #include "ac97_patch.c" 1917d1915 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 0) 1929d1926 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 14) 1931d1927 < #endif 1953,1956d1948 < #else /* <= 2.6.0 */ < #define snd_ac97_dev_register NULL < #define snd_ac97_dev_disconnect NULL < #endif Which of these am i supposed to use? am i supposed to change both of them for some backward compatibility purpose? tnx! johnu ------------------------------ Message: 3 Date: Sat, 16 Aug 2008 06:27:02 +0000 (UTC) From: Simon Tremblay <bidou(a)videotron.qc.ca> Subject: [alsa-devel] Configure and use dmix with a new ASoC driver To: alsa-devel(a)alsa-project.org Message-ID: <loom.20080816T061550-625(a)post.gmane.org> Content-Type: text/plain; charset=us-ascii Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done? ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 18, Issue 45 ******************************************

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Re: [alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by krzysztof.h1＠poczta.fm 18 Aug '08

18 Aug '08

> At Mon, 18 Aug 2008 10:50:15 +0200, > Rene Herman wrote: > > > > On 17-08-08 22:38, Krzysztof Helt wrote: > > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > > > Mute sound by setting mute bit without > > > setting volume to 0. It makes both source code > > > and binary shorter. > > > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Grmmbbled-over-by: Rene Herman <rene.herman(a)gmail.com> > > > > This should be safe and the current function indeed generates crap code > > > but in the sense of always expecting the worst broken hardware variants > > > to be out there I must say I'd rather have seen that GCC were beter. But > > > yes, sure. > > Yeah, the slightest concern is the intention of using 0x80 there. > If this patch was already tested on some real hardwares (and I guess > it should work), I'm willing to apply it. Krzystof, was this tested > on your machines? > I have tested it only on cs4236 (inside laptop) and opti931 I have in the old PC at the moment. If you want I can retest on more hardware. The second check I noticed that for some registers the volume is defined as gain (so 0 is the least audible) and for some (CS4231_MONO_CTRL, CS4231_LEFT/RIGHT_OUTPUT) is defined as attentuation (the 0 is the most audible). In two cases, the old code set volume to the loudest and mute it with mute bit, probably the opposite what was intended. It is not a crucial patch. You may drop it. Regards, Krzysztof ---------------------------------------------------------------------- Wygraj laptopa lub iPoda! Wez udzial w igrzyskach! Kliknij >>> http://link.interia.pl/f1eda

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[alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by Krzysztof Helt 18 Aug '08

18 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Mute sound by setting mute bit without setting volume to 0. It makes both source code and binary shorter. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp alsa-ref/sound/isa/wss/wss_lib.c alsa-driver/sound/isa/wss/wss_lib.c --- alsa-ref/sound/isa/wss/wss_lib.c 2008-08-14 00:05:30.000000000 +0200 +++ alsa-driver/sound/isa/wss/wss_lib.c 2008-08-17 22:33:24.620085406 +0200 @@ -574,7 +574,7 @@ static void snd_wss_calibrate_mute(struc { unsigned long flags; - mute = mute ? 1 : 0; + mute = mute ? 0x80 : 0; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->calibrate_mute == mute) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -589,34 +589,34 @@ static void snd_wss_calibrate_mute(struc chip->image[CS4231_LOOPBACK]); } snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); + mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX1_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); + mute | chip->image[CS4231_AUX2_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX2_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); + mute | chip->image[CS4231_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + mute | chip->image[CS4231_RIGHT_OUTPUT]); if (!(chip->hardware & WSS_HW_AD1848_MASK)) { snd_wss_dout(chip, CS4231_LEFT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + mute | chip->image[CS4231_LEFT_LINE_IN]); snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + mute | chip->image[CS4231_RIGHT_LINE_IN]); snd_wss_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); } if (chip->hardware == WSS_HW_INTERWAVE) { snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); + mute | chip->image[CS4231_LEFT_MIC_INPUT]); snd_wss_dout(chip, CS4231_RIGHT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); + mute | chip->image[CS4231_RIGHT_MIC_INPUT]); snd_wss_dout(chip, CS4231_LINE_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); + mute | chip->image[CS4231_LINE_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_LINE_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); + mute | chip->image[CS4231_LINE_RIGHT_OUTPUT]); } chip->calibrate_mute = mute; spin_unlock_irqrestore(&chip->reg_lock, flags); ---------------------------------------------------------------------- Igrzyska na Dzikim Zachodzie! Sprawdz >> http://link.interia.pl/f1edc

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Re: [alsa-devel] developing audio with ALSA
by Henry Ho 18 Aug '08

18 Aug '08

Hi After reading the cross-compilation instruction in INSTALL file, I already successfully configured the alsa-lib-2.0.17. I also successfully made the library. Adding --prefix=xxx during configuration can set the output directory to xxx. I also modify my Makefile for application, including the search path of libasound.so.2.***. Now I can successfully make it. However, when I run my application on the target, I got the following error message: ****************************************************************************************** ./pcm: error while loading shared libraries: libasound.so.2: cannot open shared object file: No such file or directory ****************************************************************************************** I have put libasound.so.2.0.0 in /tmp of the target. I also modified /etc/ld.so.conf, adding /tmp in ld.so.conf file. However, I cannot find ldconfig on the target. Is anyone can tell me how to solve it? Should I rebuild a busybox with this command? Henry ===================================================================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of RichWave is strictly prohibited; and any information in this email irrelevant to the official business of RichWave shall be deemed as neither given nor endorsed by RichWave.

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[alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

Hi, I'm having mic problems with snd_maestro3 and es1988 allegro hw. The laptop is HP OmniBook Xe3. The mic is way too silent (unusable) even with 20db boost option on. I found this seems to have been really common problem, but only one came with a fix [1] to the problem, enabling one more bit to volume control. Could someone please help me apply that fix to current alsa? I tried compiling 2.6.26.2 kernel and made the first change of the patch from mail [1] attachement [2]. No change. The second part of patch I did not find anymore from the code. There are similar parts in ac97_patch.c. Any hint what should be added there? The original patched code is somewhat outdated compared to modern one... But the reasoning for the original patch sounds reasonable. The mic volume is tunable, only at the very low range. I recall the laptop came with Win98, and there was no problem then with the sound -> not a HW problem. I have never got this laptop mic working with linux, so this is not anything that broke recently. I see there is check_volume_resolution function which tries to guess some of the register ranges, but maybe it get's it wrong with this HW. I tried with external headset, the behavior is just the same. The headset is working fine with IBM T41, so not due the mic itself. Any help would be preciated, thanks, ikke [1] http://thread.gmane.org/gmane.linux.alsa.devel/2025 [2] http://cache.gmane.org//gmane/linux/alsa/devel/2025-001.bin Other posts related, with no solutions: http://article.gmane.org/gmane.linux.alsa.user/9802 http://article.gmane.org/gmane.linux.alsa.user/15260 http://article.gmane.org/gmane.linux.alsa.user/9663 http://article.gmane.org/gmane.linux.alsa.devel/27195 HW is HP Omnibook Xe3, 00:08.0 Multimedia audio controller: ESS Technology ES1988 Allegro-1 (rev 12) Ubuntu up to date hardy, with manually compiled linux-2.6.26.2. the patch: ----------------------------------- --- ac97_codec.c Fri Jun 21 16:09:15 2002 +++ /usr/src/linux/sound/pci/ac97/ac97_codec.c Sun Jun 16 21:56:00 2002 @@ -579,7 +579,7 @@ static const snd_kcontrol_new_t snd_ac97_controls_mic[3] = { AC97_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), -AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 15, 1), +AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0) }; @@ -1137,7 +1137,7 @@ for (idx = 0; idx < 3; idx++) if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic[idx], ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_MIC, 0x801f); + snd_ac97_write_cache(ac97, AC97_MIC, 0x805f); /* build Line controls */ for (idx = 0; idx < 2; idx++) ----------------------------------- amixer settings: ----------------------------------- $ amixer Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 21 [68%] [-15.00dB] [on] Front Right: Playback 21 [68%] [-15.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 15 [100%] [0.00dB] [on] Simple mixer control 'Aux',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 15 [100%] [22.50dB] [on] Front Right: Capture 15 [100%] [22.50dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] ----------------------------------- ---> looking at the mic range it seems to be 0-31 already, capture is only 0-15, I wonder if that's right...

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[alsa-devel] Configure and use dmix with a new ASoC driver
by Simon Tremblay 18 Aug '08

18 Aug '08

Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done?

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[alsa-devel] [PATCH] opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
by Krzysztof Helt 17 Aug '08

17 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove card pointer from the snd_opl3sa2 structure and break circular reference snd_card->snd_opl3sa2->snd_card. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A very small opl3sa2 clean up. diff -urp linux-ref/sound/isa/opl3sa2.c linux-2.6/sound/isa/opl3sa2.c --- linux-ref/sound/isa/opl3sa2.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/opl3sa2.c 2008-08-15 12:30:58.000000000 +0200 @@ -124,7 +124,6 @@ static int pnpc_registered; #define OPL3SA2_PM_D3 (OPL3SA2_PM_ADOWN|OPL3SA2_PM_PSV|OPL3SA2_PM_PDN|OPL3SA2_PM_PDX) struct snd_opl3sa2 { - struct snd_card *card; int version; /* 2 or 3 */ unsigned long port; /* control port */ struct resource *res_port; /* control port resource */ @@ -222,14 +221,13 @@ static void snd_opl3sa2_write(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_detect(struct snd_card *card) { - struct snd_card *card; + struct snd_opl3sa2 *chip = card->private_data; unsigned long port; unsigned char tmp, tmp1; char str[2]; - card = chip->card; port = chip->port; if ((chip->res_port = request_region(port, 2, "OPL3-SA control")) == NULL) { snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); @@ -298,12 +296,14 @@ static int __devinit snd_opl3sa2_detect( static irqreturn_t snd_opl3sa2_interrupt(int irq, void *dev_id) { unsigned short status; - struct snd_opl3sa2 *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_opl3sa2 *chip; int handled = 0; - if (chip == NULL || chip->card == NULL) + if (card == NULL || card->private_data == NULL) return IRQ_NONE; + chip = card->private_data; status = snd_opl3sa2_read(chip, OPL3SA2_IRQ_STATUS); if (status & 0x20) { @@ -327,8 +327,10 @@ static irqreturn_t snd_opl3sa2_interrupt snd_opl3sa2_read(chip, OPL3SA2_MASTER_RIGHT); snd_opl3sa2_read(chip, OPL3SA2_MASTER_LEFT); if (chip->master_switch && chip->master_volume) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } } return IRQ_RETVAL(handled); @@ -512,9 +514,9 @@ static void snd_opl3sa2_master_free(stru chip->master_volume = NULL; } -static int __devinit snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; + struct snd_opl3sa2 *chip = card->private_data; struct snd_ctl_elem_id id1, id2; struct snd_kcontrol *kctl; unsigned int idx; @@ -650,7 +652,6 @@ static struct snd_card *snd_opl3sa2_card chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; - chip->card = card; card->private_free = snd_opl3sa2_free; return card; } @@ -672,9 +673,12 @@ static int __devinit snd_opl3sa2_probe(s xdma2 = dma2[dev]; if (xdma2 < 0) chip->single_dma = 1; - if ((err = snd_opl3sa2_detect(chip)) < 0) + err = snd_opl3sa2_detect(card); + if (err < 0) return err; - if (request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, "OPL3-SA2", chip)) { + err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + "OPL3-SA2", card); + if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); return -ENODEV; } @@ -694,7 +698,8 @@ static int __devinit snd_opl3sa2_probe(s err = snd_wss_mixer(wss); if (err < 0) return err; - if ((err = snd_opl3sa2_mixer(chip)) < 0) + err = snd_opl3sa2_mixer(card); + if (err < 0) return err; err = snd_wss_timer(wss, 0, NULL); if (err < 0) ---------------------------------------------------------------------- Prosty kredyt na wszystkie potrzeby. Od 1000 zl do 120 000 zl. Bez poreczycieli i zabezpieczen >>> http://link.interia.pl/f1edf

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[alsa-devel] Problem with latest alsa-info.sh patch
by Travis Place 17 Aug '08

17 Aug '08

The latest patch to allow the uploading to alsa-project.org seems to work fine, for the most part, unless you want to use the --pastebin option for the legacy upload to pastebin.ca . It seems that the uploaded data gets truncated. Not sure if this is the script, or the pastebin having problems. Ill look into it a bit more when i have time (its 5:30am here, and i need sleep). Just thought id point this out. I realise the 'default' behaviour is to use alsa-project.org for uploading, but i feel the other pastebin.ca mode should still function(I personally get people to upload there first, and then 'submit' card details to alsa-project.org/db/ afterwards. Thanks, Travis Place (wishie)

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[alsa-devel] [SOUND/SOC] removed unused #include <version.h>
by Huang Weiyi 17 Aug '08

17 Aug '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/wm8753.c sound/soc/codecs/wm9712.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809..dc7b18f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -34,7 +34,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fb7f9a..2f1c91b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h>

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Re: [alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 16 Aug '08

16 Aug '08

Hello Stan, This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" 2. aplay does not produce sound, period! 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: sudo cat /var/log/messages | grep ALSA Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped Let me know if I should send something else. // Salvador // Northwestern University ----- Original Message ---- From: stan <ghjeold_i_mwee(a)cox.net> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Friday, August 15, 2008 11:54:12 AM Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues S. Aguinaga wrote: > Hello, > > I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. > > // Salvador > Fedora uses pulseaudio by default. How are you determining that there is no output? i.e. what program are you using, and how Try aplay -vv some.wav Try aplay -vv -Dhw:0,0 some.wav Try aplay -vv -Dplughw:0,0 some.wav Are there error messages or does it look like it is playing without sound? Are there messages in /var/log/messages regarding sound at boot or while trying to play. I don't use pulseaudio and don't know it very well, but there is more information here. http://www.pulseaudio.org Could you run a script which provides more information about your sound setup and post the link back here? The script is found at http://www.alsa-project.org/alsa-info.sh You can run the script so the results occur locally, by specifying the ---no-upload option. alsa-info.sh --no-upload You would then paste the output into your email instead of providing a link. This information will enable me to determine if your setup is OK, and look for other problems.

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[alsa-devel] [PATCH 1/2] wm8990: Fix routing of left DAC to speaker mixer
by Mark Brown 15 Aug '08

15 Aug '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) This ought to go into 2.6.27 - it's a fairly straightforward bug fix. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 3ecce51..9505a18 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, - {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"}, /* LONMIX */ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, -- 1.5.6.3

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[alsa-devel] WM8750 utilization
by Harald Radke 15 Aug '08

15 Aug '08

Hi there! Sorry, if this is the wrong list for this, just tell me if so (and maybe point out where to go instead *g*). I am trying to improve usage of the wm8750 codec on my Loox720 PDA. Its a PXA272 device. So far WM8750 driver works quite ok (via I2S), however there are some things that are kinda strange. The Loox720 has a headphone jack, a earpiece and a speaker. When driving LOUT1/ROUT1, sound goes via headphones, LOUT2/ROUT2 drives the earpiece. When setting OUT3 on ROUT1, the speaker plays fine. Strange: - when disabling either LOUT2 or ROUT2 (while the other is enabled), the speaker plays the sound (not sure if the earpiece still is active as speaker output is much louder, but I think so) - when disabling ROUT1/LOUT1 with LOUT2/ROUT2 active, the earpiece plays fine, however there is still some output on the headphone (really really silently) - Trying to utilize headphone switch via the WM8750 registers doesn't really work ): depending on the polarity bit either one of the OUTs is either not muted at all or activated at all. Any ideas or hints? Thanks a lot! Harald Radke

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[alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 15 Aug '08

15 Aug '08

Hello, I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. // Salvador

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[alsa-devel] SPDIF audio / non-audio bit
by Ben Stanley 15 Aug '08

15 Aug '08

Dear alsa-devel, I recently advised this list of some problems I experienced with the SPDIF audio / non-audio bit being set inappropriately while using the ca0106 driver [1]. Seeing as I got no replies, I am trying again. When playing AC3 encoded streams (Dolby Digital, DTS), the SPDIF non-audio bit is set. Otherwise, for PCM streams, the SPDIF is set to audio. Could someone explain to me what part of the software is responsible for correctly setting the SPDIF audio / non-audio bit? Is it * alsa-driver * alsa-lib * application ? It seems to me that currently some applications (MythTV, xine) set it correctly, but most other applications ignore it. So if it is set to non-audio (because for example MythTV crashed before re-setting it to audio), then all applications that are not aware of the setting have their audio broken. Could someone set me straight here so that I can try to produce a permanent fix for this? Thanks, Ben Stanley. [1] http://thread.gmane.org/gmane.linux.alsa.devel/54398/focus=55188

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.27-rc3-74-g027ee62
by Takashi Iwai 15 Aug '08

15 Aug '08

At Fri, 15 Aug 2008 12:39:19 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > @@ -362,10 +363,8 @@ static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream) > } > if (substream->pcm->device & 2) > runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID); > - if ((err = add_playback_constraints(runtime)) < 0) { > - kfree(dpcm); > + if ((err = add_playback_constraints(runtime)) < 0) > return err; > - } Run checkpatch.pl please... Takashi

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[alsa-devel] [Fwd: [Alsa-user] found bug, fixed bug, cannot tell anyone]
by stan 15 Aug '08

15 Aug '08

Forwarded as per request -------- Original Message -------- From: - Thu Aug 14 15:28:40 2008 Message-ID: <001d01c8fe5e$3e4eed40$0402a8c0@sj4> From: Wesley Johnson <johnson2412(a)usgo.net> To: <alsa-user(a)lists.sourceforge.net> Date: Thu, 14 Aug 2008 15:36:55 -0500 (CDT) MIME-Version: 1.0 How do you notify ALSA dev team about a bug that has been in 1.0.16 and 1.0.17. ??? Bugtracker will not let me login, nor email an account confirmation. Getting an ALSA home page account seems to have no purpose at all, it does not let you do anything. BUG: When you try to compile driver for ens1371 by itself the compile will fail. The driver for ens1371 has an include of the file for ens1370, but the makefule does not have this dependency so the ens1370 file is missing. The dev team is probably always compiling for all. Diff file that fixes this problem follows. *** pci/Makefile.orig Mon Jul 14 02:53:57 2008 --- pci/Makefile Thu Jul 17 21:25:42 2008 *************** *** 27,34 **** --- 27,35 ---- atiixp_modem.c: atiixp_modem.patch $(SND_TOPDIR)/alsa-kernel/pci/atiixp_modem.c bt87x.c: bt87x.patch $(SND_TOPDIR)/alsa-kernel/pci/bt87x.c cmipci.c: cmipci.patch $(SND_TOPDIR)/alsa-kernel/pci/cmipci.c ens1370.c: ens1370.patch $(SND_TOPDIR)/alsa-kernel/pci/ens1370.c + ens1371.c: ens1370.c fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c I have been trying for since July 17 to try to send this to ALSA dev team, and have been blocked at every attempt It is now Aug 13. If someone knows how to do this, please post a reply. Wesley D. Johnson --- Get FREE High Speed Internet from USFamily.Net! -- http://www.usfamily.net/mkt-freepromo.html --- ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

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Re: [alsa-devel] [Alsa-user] found bug, fixed bug, cannot tell anyone
by Rene Herman 15 Aug '08

15 Aug '08

On 14-08-08 22:36, Wesley Johnson wrote: > How do you notify ALSA dev team about a bug that has been in 1.0.16 > and 1.0.17. ??? Here will do, although the alsa-devel mailing list would've been better still. Added to CC. > Bugtracker will not let me login, nor email an account confirmation. > Getting an ALSA home page account seems to have no purpose at all, it > does not let you do anything. The homepage is a wiki so it surely does. > BUG: > When you try to compile driver for ens1371 by itself the compile will fail. > The driver for ens1371 has an include of the file for ens1370, but the > makefule does not have this dependency > so the ens1370 file is missing. The dev team is probably always compiling > for all. > > Diff file that fixes this problem follows. > > *** pci/Makefile.orig Mon Jul 14 02:53:57 2008 > --- pci/Makefile Thu Jul 17 21:25:42 2008 > *************** > *** 27,34 **** > --- 27,35 ---- > atiixp_modem.c: atiixp_modem.patch > $(SND_TOPDIR)/alsa-kernel/pci/atiixp_modem.c > bt87x.c: bt87x.patch $(SND_TOPDIR)/alsa-kernel/pci/bt87x.c > cmipci.c: cmipci.patch $(SND_TOPDIR)/alsa-kernel/pci/cmipci.c > ens1370.c: ens1370.patch $(SND_TOPDIR)/alsa-kernel/pci/ens1370.c > + ens1371.c: ens1370.c I expect this should be a straight copy of the ens1370.c line but never use the standalone alsa-driver package. Now that we're on alsa-devel, people who do will reply though. > fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c > intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c > maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c > via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c > > > I have been trying for since July 17 to try to send this to ALSA dev team, > and have been blocked at every attempt > It is now Aug 13. If someone knows how to do this, please post a reply. Well, not to unduly annoy you, and thank you very much for the report and the persistence, but basically all Linux development happens on mailinglists and the mailinglist information is right there on the homepage so, well... The alsa-devel mailinglist is moderated for non-subscribers; if you did email alsa-devel before and just weren't let through the moderation queue then please say so because in that case we definitely have a problem to fix. Rene.

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Re: [alsa-devel] sound-problems with ALC262
by Takashi Iwai 15 Aug '08

15 Aug '08

At Thu, 14 Aug 2008 07:09:41 +0200, Tobias Kaefer wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > > Takashi Iwai schrieb: > | At Wed, 13 Aug 2008 08:07:37 +0200, > | Tobias Kaefer wrote: > |> ... > |> "stan" told me to use this script > |> http://www.alsa-project.org/alsa-info.sh to create more information > |> about my system. Here is the pastebin link of the output of that script: > |> http://pastebin.ca/1169153 > | > | At a quick glance, I see no problem in the configuration. > | > | Does the problem exist even if you play via PCM "hw" device, such as > | > | % aplay -Dhw foo.wav > | > | ?? > | If your WAV file isn't with the supported format/rate, use plughw: > | > | % aplay -Dplughw foo.wav > > Yes. The problem still exists when using the PCM "hw" or "plughw" device. What happens when you load the driver with bdl_pos_adj=0 option? Takashi

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Re: [alsa-devel] [PATCH] hda: dynamically create capture mux controls
by Matthew Ranostay 15 Aug '08

15 Aug '08

Dynamically create capture mux volume controls when a output amp is detected. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..eae6df1 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -794,7 +794,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), { } /* end */ }; @@ -908,12 +907,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - /* analog pc-beep replaced with digital beep support */ /* HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), @@ -931,11 +927,9 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -943,7 +937,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; @@ -953,12 +946,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -967,11 +957,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -982,15 +970,12 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2351,7 +2336,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) +static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, + int idx, const char *name, unsigned long val) { struct snd_kcontrol_new *knew; @@ -2371,6 +2357,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = stac92xx_control_templates[type]; + knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; @@ -2379,6 +2366,14 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char return 0; } + +/* add dynamic controls */ +static int stac92xx_add_control(struct sigmatel_spec *spec, int type, + const char *name, unsigned long val) +{ + return stac92xx_add_control_idx(spec, type, 0, name, val); +} + /* flag inputs as additional dynamic lineouts */ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) { @@ -2780,6 +2775,26 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, return 0; } +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + err = stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3078,6 +3093,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2)

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[alsa-devel] [PATCH] hda: 92HD75xx fixes
by Matthew Ranostay 15 Aug '08

15 Aug '08

Fixed several noise issues with DACs and ADCs on some 92HD75xxx based codecs with certain revision id's. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..9ec0610 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -145,6 +145,9 @@ struct sigmatel_spec { unsigned int gpio_data; unsigned int gpio_mute; + /* stream */ + unsigned int stream_delay; + /* analog loopback */ unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -190,6 +193,7 @@ struct sigmatel_spec { unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; + unsigned int powerdown_adcs; /* i/o switches */ unsigned int io_switch[2]; @@ -2010,6 +2014,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + if (spec->stream_delay) + msleep(spec->stream_delay); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -2073,9 +2079,14 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); + if (spec->powerdown_adcs) { + msleep(40); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } @@ -2084,8 +2095,12 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + snd_hda_codec_cleanup_stream(codec, nid); + if (spec->powerdown_adcs) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); return 0; } @@ -3275,6 +3290,12 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init); + /* power down adcs initially */ + if (spec->powerdown_adcs) + for (i = 0; i < spec->num_adcs; i++) + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ @@ -3909,6 +3930,47 @@ again: return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_POWER_STATE, pwr); + + msleep(1); + for (i = 0; i < spec->num_adcs; i++) { + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, pwr); + } +}; + +static int stac92hd71xx_resume(struct hda_codec *codec) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D0); + return stac92xx_resume(codec); +} + +static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + return 0; +}; + +#endif + +static struct hda_codec_ops stac92hd71bxx_patch_ops = { + .build_controls = stac92xx_build_controls, + .build_pcms = stac92xx_build_pcms, + .init = stac92xx_init, + .free = stac92xx_free, + .unsol_event = stac92xx_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .resume = stac92hd71xx_resume, + .suspend = stac92hd71xx_suspend, +#endif +}; static int patch_stac92hd71bxx(struct hda_codec *codec) { @@ -3920,6 +3982,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = stac92xx_patch_ops; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; @@ -3951,6 +4014,14 @@ again: spec->init = stac92hd71bxx_core_init; break; case 0x111d7608: /* 5 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 0 || + (codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; @@ -3960,6 +4031,13 @@ again: stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; /* fallthru */ @@ -3976,6 +4054,7 @@ again: spec->gpio_dir = 0x01; spec->gpio_data = 0x01; + spec->powerdown_adcs = 1; spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -4008,8 +4087,6 @@ again: return err; } - codec->patch_ops = stac92xx_patch_ops; - return 0; };

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