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[alsa-devel] sscape: general code improvements
by Krzysztof Helt 07 Oct '09

07 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> General code and coding style improvements. Add a missing argument for SET_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- What is a standard name for switch to select internal or external midi? It is very easy to add such a functionality to the driver. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be50..4380e85 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -190,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -331,7 +334,8 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) */ static inline void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -352,22 +356,25 @@ static void soundscape_free(struct snd_card *c) */ static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -429,11 +436,12 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) */ static int upload_dma_data(struct soundscape *s, const unsigned char *data, - size_t size) + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +451,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +483,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +500,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -591,7 +603,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +613,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +627,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +651,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +753,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +777,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +793,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +804,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +818,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +881,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +930,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1155,7 +1157,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1186,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1240,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1297,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c index 30c36d1..7e06424 100644 --- a/sound/oss/sscape.c +++ b/sound/oss/sscape.c @@ -160,19 +160,27 @@ static void sleep(unsigned howlong) schedule_timeout(howlong); } +static unsigned char __sscape_read(struct sscape_info *devc, int reg) +{ + unsigned char val; + + outb(reg, PORT(ODIE_ADDR)); + val = inb(PORT(ODIE_DATA)); + return val; +} + static unsigned char sscape_read(struct sscape_info *devc, int reg) { unsigned long flags; unsigned char val; spin_lock_irqsave(&devc->lock,flags); - outb(reg, PORT(ODIE_ADDR)); - val = inb(PORT(ODIE_DATA)); + val = __sscape_read(devc, reg); spin_unlock_irqrestore(&devc->lock,flags); return val; } -static void __sscape_write(int reg, int data) +static void __sscape_write(struct sscape_info *devc, int reg, int data) { outb(reg, PORT(ODIE_ADDR)); outb(data, PORT(ODIE_DATA)); @@ -183,7 +191,7 @@ static void sscape_write(struct sscape_info *devc, int reg, int data) unsigned long flags; spin_lock_irqsave(&devc->lock,flags); - __sscape_write(reg, data); + __sscape_write(devc, reg, data); spin_unlock_irqrestore(&devc->lock,flags); } @@ -342,9 +350,9 @@ static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, in temp |= 0x80; /* 8 bit DMA channel */ temp |= 1; /* Trigger DMA */ - sscape_write(devc, GA_DMAA_REG, temp); + __sscape_write(devc, GA_DMAA_REG, temp); temp &= 0xfe; /* Clear DMA trigger */ - sscape_write(devc, GA_DMAA_REG, temp); + __sscape_write(devc, GA_DMAA_REG, temp); } static int verify_mpu(struct sscape_info *devc) @@ -392,7 +400,7 @@ static void sscape_coproc_close(void *dev_info, int sub_device) spin_lock_irqsave(&devc->lock,flags); if (devc->dma_allocated) { - __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */ + __sscape_write(devc, GA_DMAA_REG, 0x20); /* DMA channel disabled */ devc->dma_allocated = 0; } spin_unlock_irqrestore(&devc->lock,flags); @@ -418,7 +426,7 @@ static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, */ spin_lock_irqsave(&devc->lock,flags); - codec_dma_bits = sscape_read(devc, GA_CDCFG_REG); + codec_dma_bits = __sscape_read(devc, GA_CDCFG_REG); if (devc->dma_allocated == 0) devc->dma_allocated = 1; @@ -500,11 +508,11 @@ static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, x = inb(PORT(HOST_DATA)); if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ { - DDB(printk("Soundscape: Acknowledge = %x\n", x)); + (printk("Soundscape: Acknowledge = %x\n", x)); done = 1; } } - sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); + __sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); spin_unlock_irqrestore(&devc->lock,flags); if (!done) @@ -604,6 +612,16 @@ static coproc_operations sscape_coproc_operations = static struct resource *sscape_ports; static int sscape_is_pnp; +static int sscape_read_host_ctrl(sscape_info* devc) +{ + return host_read(devc); +} + +static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) +{ + host_command2(devc, a, b); +} + static void __init attach_sscape(struct address_info *hw_config) { #ifndef SSCAPE_REGS @@ -669,22 +687,25 @@ static void __init attach_sscape(struct address_info *hw_config) spin_lock_irqsave(&devc->lock,flags); /* Host interrupt enable */ - sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ + __sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ /* DMA A status/trigger register */ - sscape_write(devc, 2, 0x20); /* DMA channel disabled */ + __sscape_write(devc, 2, 0x20); /* DMA channel disabled */ /* DMA B status/trigger register */ - sscape_write(devc, 3, 0x20); /* DMA channel disabled */ + __sscape_write(devc, 3, 0x20); /* DMA channel disabled */ /* Host interrupt config reg */ - sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); + __sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); /* Don't destroy CD-ROM DMA config bits (0xc0) */ - sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0)); + __sscape_write(devc, 5, (regs[5] & 0x3f) | (__sscape_read(devc, 5) & 0xc0)); /* CD-ROM config (WSS codec actually) */ - sscape_write(devc, 6, regs[6]); - sscape_write(devc, 7, regs[7]); - sscape_write(devc, 8, regs[8]); + __sscape_write(devc, 6, regs[6]); + __sscape_write(devc, 7, regs[7]); + __sscape_write(devc, 8, regs[8]); /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */ - sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08); + __sscape_write(devc, 9, (__sscape_read(devc, 9) & 0xf0) | 0x08); spin_unlock_irqrestore(&devc->lock,flags); + sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */ + sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */ + sscape_write_host_ctrl2( devc, 0x8A, 0); } #ifdef SSCAPE_DEBUG2 /* @@ -721,7 +742,7 @@ static int detect_ga(sscape_info * devc) { unsigned char save; - DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base)); + printk(KERN_DEBUG "Entered Soundscape detect_ga(%x)\n", devc->base); /* * First check that the address register of "ODIE" is @@ -771,16 +792,6 @@ static int detect_ga(sscape_info * devc) return 1; } -static int sscape_read_host_ctrl(sscape_info* devc) -{ - return host_read(devc); -} - -static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) -{ - host_command2(devc, a, b); -} - static int sscape_alloc_dma(sscape_info *devc) { char *start_addr, *end_addr; @@ -869,8 +880,8 @@ static void sscape_pnp_start_dma(sscape_info* devc, int arg ) if (arg == 0) reg = 2; else reg = 3; - sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); - sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); + __sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); + __sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); } static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) @@ -885,11 +896,11 @@ static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) sleep ( 1 ); i = 0; do { - d = sscape_read(devc, reg) & 1; + d = __sscape_read(devc, reg) & 1; if ( d == 1) break; i++; } while (i < 500000); - d = sscape_read(devc, reg) & 1; + d = __sscape_read(devc, reg) & 1; return d; } @@ -1242,8 +1253,11 @@ static int __init probe_sscape(struct address_info *hw_config) } if (!detect_ga(devc)) { - if (detect_sscape_pnp(devc)) + printk(KERN_DEBUG "GA detected!\n"); + if (detect_sscape_pnp(devc)) { + printk(KERN_DEBUG "PnP detected!\n"); return 1; + } release_region(devc->base, 2); release_region(devc->base + 2, 6); return 0; ---------------------------------------------------------------------- Zobacz najwiekszy samolot na swiecie! 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2 1

[alsa-devel] PCM device volume
by Milan Křápek 07 Oct '09

07 Oct '09

Hi, I am new with alsa. I am developing application that will have audio output on linux systems via alsa. I found many hints when I was trying to play some audio. I used snd_pcm_open(&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); snd_pcm_hw_params_any(pcm_handle, hwparams); to open and set my PCM device for writting audio. But I have got lost when I tried to change my audio volume. I suppose that I need to get some mixer int snd_mixer_open (snd_mixer_t **mixer, int mode) but I could not find how to code that the new mixer will control the PCM device. Pleas is there any tutorial how to do it? Or can somebody show me any code snippet? I have tried to find any similar topic, but without success. Thanks for any advice Milan

1 0

[alsa-devel] Application supporting forward/rewind APIs
by Anuj Aggarwal 07 Oct '09

07 Oct '09

Hi, I want to try the forward/rewind APIs supported by the ALSA library. Can someone point to an ALSA application which exercises these APIs? Best Regards, Anuj Aggarwal

4 8

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 06 Oct '09

06 Oct '09

The following changes since commit 88439ac793934a47f47ad285656b63d09f5937c8: Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The new commits that aren't part of v2.6.32-rc3 (which is being merged up via fix/asoc here) are: Jonathan Cameron (1): ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Mark Brown (4): Merge branch 'for-2.6.32' into for-2.6.33 ASoC: Simplify code for DAPM widget updates ASoC: Push DAPM enumeration register change test out ASoC: Add virtual enumeration support for DAPM muxes Peter Ujfalusi (2): ASoC: Fix SND_SOC_DAPM_LINE handling ASoC: Improve the debugfs hierarchy

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[alsa-devel] ALSA integration CS4281 PMC on PPC platform
by Harish Kumar 06 Oct '09

06 Oct '09

Hi, I am in the process of integrating CS4281 on PPC platform. I have included all the alsa driver and selected the CS4281 in my kernel configuration. The alsa driver is integrated as part of the kernel image. The kernel version is 2.6.15.7. I compiled the alsa-lib and utils and copied to the RFS and tried executing aplay /audio8k16S.wav which flagged the following error. LSA lib pcm.c:2099:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:547: audio open error: No such file or directory Pls find the attachement for the capture of debug messages from proc interface. Thanks and Regards Harish Kumar.B

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[alsa-devel] hda_intel: SPDIF not detected on Asus P7P55D PRO
by Robert Hancock 06 Oct '09

06 Oct '09

I just set up a new system running an Asus P7P55D PRO motherboard. It looks like this uses the VIA VT1828S codec. The analog output seems to work but the SPDIF output isn't recognized. Originally tried 2.6.30.5, but Linus' current git doesn't seem any different. The device only gets detected as HDA Generic, which, from my reading of the code, ignores SPDIF outputs. I tried hacking patch_via.c to add the codec ID to match existing types like VT1708S and VT1702, each of those got the SPDIF output to show up, but no output showed up on the receiver, and also the analog output then stopped being recognized (looks like some node IDs don't match). Is there some extra support that needs to be added for this codec, or anything else I can try to get this working? The SPDIF does work in Windows 7 with just the standard Microsoft HDA driver.. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 The codec data seems to have references to the SPDIF output, but no device is actually getting created. The data is below. Any suggestions? cat /proc/asound/card0/codec#0 Codec: VIA ID 4441 Address: 0 Function Id: 0x1 Vendor Id: 0x11064441 Subsystem Id: 0x104383cf Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x08 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x2a 0x2a] Converter: stream=5, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x09 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0a [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0b [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0c [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x2a 0x2a] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0f [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x10 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x14 0x14] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x8b 0x8b] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x08 0x21 Node 0x19 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x09 0x21 Node 0x1a [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x0b 0x21 Node 0x1b [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x34 0x21 Node 0x1c [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x35 0x21 Node 0x1d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1e [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x2c* 0x2b 0x2a 0x29 0x28 0x21 Node 0x1f [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x2c 0x2b* 0x2a 0x29 0x28 0x21 Node 0x20 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Power: setting=D0, actual=D0 Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x24 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x19 Node 0x26 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x0a Node 0x27 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1a Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1b Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19037: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1c Node 0x2a [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x09* 0x0c Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x0a* 0x0c Node 0x2c [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x90331138: [Fixed] CD at Int N/A Conn = ATAPI, Color = Black DefAssociation = 0x3, Sequence = 0x8 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power: setting=D0, actual=D0 Node 0x2d [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000090: OUT HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 1 0x0e Node 0x2e [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074411f0: [Jack] SPDIF Out at Ext Rear Panel Conn = RCA, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x2f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x30 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x31 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x32 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x33 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x34 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 3 0x08* 0x0b 0x0c Node 0x35 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 3 0x08* 0x0b 0x0c

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[alsa-devel] [PATCH 1/1] ICE1712/24 - Change the Multi Track Peak control (level meters) from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: pavel <pavel(a)insite.cz> * PLEASE NOTE - this change requires the corresponding update of envy24control for ice1712 - kind of an ABI change. * The "Multi Track Peak" control is read-only level meters indicator. * The control is VERY confusing to most users since it is currently displayed in regular mixers. E.g. alsamixer ignores its read-only status and allows changing the levels with keys which makes no sense. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 56d8d67..c7cff6f 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2269,7 +2269,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ice1712_pro_peak_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3896fb9..28ca8a4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2160,7 +2160,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, -- 1.5.4.3

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[alsa-devel] alsa-utils cross compilation error
by Harish Kumar 06 Oct '09

06 Oct '09

Hi, I am trying to cross compile the alsa utils for PPC and as well as arm architecture. the compilation of alsa-libs was successful with their respective toolchains. But configuring the alsa-util is failing. I am using alsa-utils-1.0.11rc2,alsa-lib-1.0.11rc2. I am getting the error checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. I feel i have to provide the path of the include files but not sure which path to be passed and how it needs to be passed to configure. Request your help to resolve the same. I invoked the configure using the following command and the output of the configure is as below. root@localhost alsa-utils-1.0.11rc2]# CC=arm-linux-gcc ./configure --target=arm-linux --host=i686-linux configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... gawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ANSI C... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld checking if the linker (/opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking whether NLS is requested... yes checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ANSI C... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. [2]+ Done gedit configure [root@localhost alsa-utils-1.0.11rc2]# Thanks and Regards Harish Kumar.B

1 0

[alsa-devel] [PATCH 1/1] Changing the Multi Track Peak control from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> * The "Multi Track Peak" control is now of PCM type, to avoid confusing users in other alsa mixers. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/envy24control/levelmeters.c b/envy24control/levelmeters.c index 3bf1a94..fa44979 100644 --- a/envy24control/levelmeters.c +++ b/envy24control/levelmeters.c @@ -261,9 +261,15 @@ void level_meters_reset_peaks(GtkButton *button, gpointer data) void level_meters_init(void) { + int err; + snd_ctl_elem_value_malloc(&peaks); - snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_PCM); snd_ctl_elem_value_set_name(peaks, "Multi Track Peak"); + if ((err = snd_ctl_elem_read(ctl, peaks)) < 0) + /* older ALSA driver, using MIXER type */ + snd_ctl_elem_value_set_interface(peaks, + SND_CTL_ELEM_IFACE_MIXER); } void level_meters_postinit(void) -- 1.5.4.3

1 0

[alsa-devel] [PATCH 1/1] Changing the Multi Track Peak control from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> * The "Multi Track Peak" control is now of PCM type, to avoid confusing users in other alsa mixers. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/envy24control/levelmeters.c b/envy24control/levelmeters.c index 3bf1a94..9bdaace 100644 --- a/envy24control/levelmeters.c +++ b/envy24control/levelmeters.c @@ -262,7 +262,7 @@ void level_meters_reset_peaks(GtkButton *button, gpointer data) void level_meters_init(void) { snd_ctl_elem_value_malloc(&peaks); - snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_PCM); snd_ctl_elem_value_set_name(peaks, "Multi Track Peak"); } -- 1.5.4.3

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