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[alsa-devel] [PATCH 1/1] sound: usb-audio: support for Digidesign Mbox 2
by Damien Zammit 27 Aug '09

27 Aug '09

Provides magic boot sequence which activates the device, must loop until device is ready, (returns status bytes). Provides extra magic to set the device into 24 bit mode. -Playback works @ 24-bit -MIDI I/O works -Capture is untested, but PCM stream is active -S/PDIF (RCA) I/O is untested, but lights activate on the device S/PDIF mode works by setting the required flag in usbaudio.c. I think the device can't do both simultaneously, since different altsettings are required to set it. Remaining issues: 1) Double check endianess of 24 bit capture. 2) Check duplex stability 3) Tell ALSA to correctly detect S24_3BE on the device. Currently, speaker-test assumes S16_LE and won't allow S24_3BE mode, but jackd 0.116-2 detects and plays S24_3BE ? 4) Create a kernel module switch to select spdif/analogue I/O Playback sample format : S24_3BE MIDI format : MIDIMAN protocol Signed-off-by: Damien Zammit <damien.zammit(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 43d7d88..3060941 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -943,7 +943,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream { struct snd_usb_substream *subs; snd_pcm_uframes_t hwptr_done; - + subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; @@ -2441,6 +2441,12 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor fp->altsetting, sample_width, sample_bytes); break; } + + /* Digidesign Mbox 2 workaround: + * supports S24_3BE BIG ENDIAN */ + if (chip->usb_id == USB_ID(0x0dba, 0x3000)) + pcm_format = SNDRV_PCM_FORMAT_S24_3BE; + break; case USB_AUDIO_FORMAT_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; @@ -2550,13 +2556,20 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * switch (chip->usb_id) { case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - if (device_setup[chip->index] == 0x00 && + if (device_setup[chip->index] == 0x00 && fp->altsetting == 6) pcm_format = SNDRV_PCM_FORMAT_S16_BE; else pcm_format = SNDRV_PCM_FORMAT_S16_LE; break; - default: + + /* Digidesign Mbox 2 supports SPDIF 24 bit (havent tested endianess) */ + case USB_ID(0x0dba, 0x3000): + pcm_format = SNDRV_PCM_FORMAT_S24_3BE; + break; + + default: + snd_printdd("Format III detected, setting default S16_LE\n"); pcm_format = SNDRV_PCM_FORMAT_S16_LE; } } else { @@ -2655,6 +2668,8 @@ static unsigned char parse_datainterval(struct snd_usb_audio *chip, static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -2700,13 +2715,19 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x0763, 0x2003) && + if (chip->usb_id == USB_ID(0x0763, 0x2003) && audiophile_skip_setting_quirk(chip, iface_no, altno)) continue; + /* Digidesign Mbox 2: skip altsets incompatible with device_setup + */ + if (chip->usb_id == USB_ID(0x0dba, 0x3000) && + mbox2_skip_setting_quirk(chip, iface_no, altno)) + continue; + /* get audio formats */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); if (!fmt) { @@ -2995,7 +3016,7 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. */ static int create_uaxx_quirk(struct snd_usb_audio *chip, @@ -3264,6 +3285,97 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac return 0; } + +#define MBOX2_FIRMWARE_SIZE 646 +#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ +#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ + +static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + u8 enablemagic[3]; + u8 bootresponse; + int fwsize; + + if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { + snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); + + /* From USB Snoop, + * + * SetupPacket = RT RQ VHVL INDX SIZE + * RT = 0xRT Request Type + * RQ = 0xRQ Request + * VHVL = 0xVLVH Value in reverse byte order + * INDX = 0xDXIN Index in reverse byte order + * SIZE = 0xZESI Size in reverse byte order + * + * Magic boot code setup packet: c0 85 01 00 00 00 12 00 + * RQ RT VLVH DXIN ZESI + * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT + * for snd_usb_ctl_msg() + */ + +mbox2_reboot: + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); + + if (bootresponse == MBOX2_BOOT_LOADING) { + snd_printdd("device not ready, resending boot sequence...\n"); + goto mbox2_reboot; + } + + if (bootresponse == MBOX2_BOOT_READY) { + snd_printdd("device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); + err = usb_reset_configuration(dev); + if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); + snd_printdd("mbox2_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) + * but now we will try sending capture/playback enable magic + * + * 80 bb 00 = 24bit mode - S24_3BE + * 44 ac 00 = 16bit mode? + * + */ + //enablemagic[0]=0x44; + //enablemagic[1]=0xac; + //enablemagic[2]=0x00; + + /* We want 24 bit mode for now */ + enablemagic[0]=0x80; + enablemagic[1]=0xbb; + enablemagic[2]=0x00; + + /* Send the magic! */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); + + return 0; /* Succesful boot */ + } + + snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); + return -ENODEV; + } + snd_printdd("Invalid firmware size: %d\n",fwsize); + return -ENODEV; +} + static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) { u8 buf = 1; @@ -3366,10 +3478,79 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5) return 1; /* skip this altsetting */ - } + } return 0; /* keep this altsetting */ } +/* Digidesign Mbox 2 I/O selector + * Uncomment this line to enable SPDIF */ +//#define MBOX2_SPDIF_IO + +#define MBOX2_FIRMWARE_INTERFACE 0 +#define MBOX2_CONTROL_INTERFACE 1 +#define MBOX2_MIDI_INTERFACE 6 + +static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* Reset ifaces 2-5 to 0 altsetting. */ + usb_set_interface(chip->dev, iface, 0); + +#ifdef MBOX2_SPDIF_IO + /* Enable SPDIF altsettings */ + switch (iface) { + case MBOX2_FIRMWARE_INTERFACE: + case MBOX2_CONTROL_INTERFACE: + return 0; /*keep*/ + break; + case 2: + if (altno == 2) return 0; else return 1; + break; + case 3: + return 1; + break; + case 4: + if (altno == 2) return 0; else return 1; + break; + case 5: + return 1; + break; + case MBOX2_MIDI_INTERFACE: + return 0; /*keep*/ + break; + default: + snd_printdd("Invalid interface for Mbox 2: %d\n",iface); + return 1; /* skip altsetting */ + } +#else + /* Use analogue I/O */ + switch (iface) { + case MBOX2_FIRMWARE_INTERFACE: + case MBOX2_CONTROL_INTERFACE: + return 0; + break; + case 2: + if (altno == 3) return 0; else return 1; + break; + case 3: + if (altno == 6) return 0; else return 1; + break; + case 4: + if (altno == 3) return 0; else return 1; + break; + case 5: + if (altno == 3) return 0; else return 1; + break; + case MBOX2_MIDI_INTERFACE: + return 0; + break; + default: + snd_printdd("Invalid interface for Mbox 2: %d\n",iface); + return 1; /* skip altsetting */ + } +#endif +} + /* * audio-interface quirks * @@ -3392,6 +3573,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, + [QUIRK_MIDI_MBOX2] = snd_usb_create_midi_interface, [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, @@ -3425,7 +3607,7 @@ static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_ { struct snd_usb_audio *chip = entry->private_data; if (!chip->shutdown) - snd_iprintf(buffer, "%04x:%04x\n", + snd_iprintf(buffer, "%04x:%04x\n", USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); } @@ -3616,6 +3798,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* Digidesign Mbox 2 needs magic boot sequence */ + if (id == USB_ID(0x0dba, 0x3000)) { + if (snd_usb_mbox2_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f789..21aab48 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -157,6 +157,7 @@ enum quirk_type { QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, QUIRK_MIDI_US122L, + QUIRK_MIDI_MBOX2, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19c..bc11fe4 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -423,7 +423,7 @@ static void snd_usbmidi_maudio_broken_running_status_input( u8 cin = buffer[i] & 0x0f; struct usbmidi_in_port *port = &ep->ports[cable]; int length; - + length = snd_usbmidi_cin_length[cin]; if (cin == 0xf && buffer[i + 1] >= 0xf8) ; /* realtime msg: no running status change */ @@ -617,13 +617,13 @@ static struct usb_protocol_ops snd_usbmidi_standard_ops = { static struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; @@ -1577,7 +1577,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) - return 0; + return 0; intf = umidi->iface; if (!intf || intf->num_altsetting < 1) @@ -1615,7 +1615,7 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint_info* endpoints) { int err, i; - + err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { if (endpoints[i].out_ep) @@ -1876,6 +1876,18 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_MBOX2: + /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ + umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; + /* + * We have to make sure that the USB core looks + * again at interface 6 by calling usb_set_interface() on it. + */ + usb_set_interface(umidi->chip->dev, 6, 0); + memcpy(&endpoints[0], quirk->data, + sizeof(struct snd_usb_midi_endpoint_info)); + err = 0; + break; case QUIRK_MIDI_FASTLANE: umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; /* diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201e..e961f5e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2082,6 +2082,60 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* DIGIDESIGN MBOX 2 */ +{ + /* Damien Zammit <damien.zammit(a)gmail.com> */ + + USB_DEVICE(0x0dba, 0x3000), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 2", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 5, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 6, + .type = QUIRK_MIDI_MBOX2, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_ep = 0x02, + .out_cables = 0x0001, + .in_ep = 0x81, + .in_interval = 0x01, + .in_cables = 0x0001 + } + + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- 1.5.4.3

2 3

[alsa-devel] ALSA default buffer sizes (recent now being subscribed with correct address... hopefully)
by Manuel Jander 26 Aug '09

26 Aug '09

Hi there, I'm somewhat upset because audacious does not work with my AU8830. The problem is that audacious never calls any function that would give any clue about the desired buffer size. So my AU8830 ends up with 32 periods, 1 byte each. Great ! I'm not going to blame the author of that driver :P ... but: The question is, who is the entity deciding those default values ? Other soundcards seem to get reasonable defaults, but the AU8830 does not. So I would like to fix that, but I want to do that at the right place. Best Regards, and many thanks for any hint.

3 4

[alsa-devel] [PATCH] vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
by Roel Kluin 26 Aug '09

26 Aug '09

Since !LI_CCFG_* evaluates to 0, this did not change anything to cfgval and ctlval. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- // vi sound/oss/vwsnd.c +392 # define LI_CCTL_RESET 0x80000000 ... # define LI_CCTL_DMA_ENABLE 0x08000000 ... # define LI_CCFG_LOCK 0x00008000 I am not sure this is what was intended, does anyone have the specs? diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 187f727..6713110 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -628,7 +628,7 @@ static void li_setup_dma(dma_chan_t *chan, ASSERT(!(buffer_paddr & 0xFF)); chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8); - chan->cfgval = (!LI_CCFG_LOCK | + chan->cfgval = ((chan->cfgval & ~LI_CCFG_LOCK) | SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) | desc->direction | mode | @@ -638,9 +638,9 @@ static void li_setup_dma(dma_chan_t *chan, tmask = 13 - fragshift; /* See Lithium DMA Notes above. */ ASSERT(size >= 2 && size <= 7); ASSERT(tmask >= 1 && tmask <= 7); - chan->ctlval = (!LI_CCTL_RESET | + chan->ctlval = ((chan->ctlval & ~LI_CCTL_RESET) | SHIFT_FIELD(size, LI_CCTL_SIZE) | - !LI_CCTL_DMA_ENABLE | + (chan->ctlval & ~LI_CCTL_DMA_ENABLE) | SHIFT_FIELD(tmask, LI_CCTL_TMASK) | SHIFT_FIELD(0, LI_CCTL_TPTR));

2 1

[alsa-devel] [PATCH 1/4] SOUND: atiixp, use uninitialized_var(chip)
by Jiri Slaby 26 Aug '09

26 Aug '09

To avoid wrong compiler warnings, use unitialized_var(chip) in snd_atiixp_probe's from atiixp and atiixp_modem drivers. 'chip' is unused when unset due to retval being nonzero. Signed-off-by: Jiri Slaby <jirislaby(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752df..436ffea 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1648,7 +1648,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp *chip; + struct atiixp *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147b..b6191f2 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1285,7 +1285,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct snd_card *card; - struct atiixp_modem *chip; + struct atiixp_modem *uninitialized_var(chip); int err; err = snd_card_create(index, id, THIS_MODULE, 0, &card); -- 1.6.3.3

2 4

[alsa-devel] alsa random crash
by some guy 26 Aug '09

26 Aug '09

Problem: Alsa randomly crashes after ~10 hours of use; this varies from 2 to 20 hours. Symptoms: Sound stops while media is playing (in web browser, or mplayer), or convulses sporatically. The issue is not always reproducible (i.e I can reproduce the result when I want), but it is perisistant. Info: http://www.alsa-project.org/db/?f=fe434884ea7a16b1ae7c06cfd282832971f41fd6 lspci erhttp://pastebin.ca/1542518 kernel version is gentoo-2.6.28-r5 I really appreciate it. -anonymous alsa'er. _________________________________________________________________ More storage. Better anti-spam and antivirus protection. Hotmail makes it simple. http://go.microsoft.com/?linkid=9671357

1 0

[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..4a3f62d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

2 1

Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 25 Aug '09

25 Aug '09

2009/8/25 Takashi Iwai <tiwai(a)suse.de> > BTW, did you check, as I suggested, whether acer-aspire-6530g with > changing the channel mode from 2ch to 6ch works for 7730g? > I guess we can merge acer-aspire-* variants to one or two models. > Can you describe in details what I need to do, please?

1 0

Re: [alsa-devel] Acer Aspire 7730G support
by Takashi Iwai 25 Aug '09

25 Aug '09

At Mon, 24 Aug 2009 19:57:58 +0400, Denis Kup wrote: > > 2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > > At Mon, 24 Aug 2009 17:10:10 +0400, > > Denis Kup wrote: > > > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > > What about my patch? > > > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > > whether it still work and repost the patch. There have been lots of > > changes in HD-audio driver code, so an old patch might not work any > > more. > > > > > > thanks, > > > > Takashi > > > I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it > should. Try rather with the snapshot tarball in ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… The version you tried isn't the latest one. > It's diff between patched version( patch_realtek_new.c ) and unpatched: Make a patch with diff -u option (and better with -p, too). thanks, Takashi

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[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 25 Aug '09

25 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..7ad99d2 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x3); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x2); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

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[alsa-devel] [PATCH] ASoC: s3c24xx platform: Fix s3c2410_dma_started called at wrong time
by Shine Liu 25 Aug '09

25 Aug '09

s3c24xx dma has the auto reload feature, when the the trnasfer is done, CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So the transmission is repeated. IRQ is issued while auto reload occurs. We change the DISRC and DCON[19:0] in the ISR, but at this time, the auto reload has been performed already. The first block is being re-transmitted by the DMA. So we need rewrite the DISRC and DCON[19:0] for the next block immediatly after the this block has been started to be transported. The function s3c2410_dma_started() is for this perpose, which is called in the form of "s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger(). But it is not correct. DMA transmission won't start until DMA REQ signal arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in s3c24xx_i2s_trigger(). In the current framework, s3c24xx_pcm_trigger() is always called before s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in this function. However, s3c2410_dma_started() is dma related, to call this function we should provide the channel number, which is given by substream->runtime->private_data->params->channel. The private_data points to a struct s3c24xx_runtime_data object, which is define in s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c To move the struct s3c24xx_runtime_data defination to s3c24xx-pcm.h is one solution. If not moved, the patch shold be like: Signed-off-by: Shine Liu <shinel(a)foxmail.com> ---------------------------------------------------------------- --- a/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-20 16:06:14.000000000 +0800 @@ -28,4 +28,6 @@ extern struct snd_soc_platform s3c24xx_soc_platform; extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream); + #endif --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-20 15:56:46.000000000 +0800 @@ -66,6 +66,28 @@ struct s3c24xx_pcm_dma_params *params; }; +/* s3c24xx_pcm_dma_started + * + * To load the second dma buffer immediately after the + * first dma buffer has been started transferring, so that + * in the dma autoreload situation, the first data block + * won't be re-transferred after the first irq issued. + * + * This function is called in the s3c24xx-i2s-trigger() of + * s3c24xx-i2s.c, just after the dma transfer is really started. + * + * It's ugly here. Need a elegant way to fix this bug. + * FIXME + * + */ + +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); + +} + /* s3c24xx_pcm_enqueue * * place a dma buffer onto the queue for the dma system @@ -255,7 +277,6 @@ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-20 16:08:06.000000000 +0800 @@ -296,6 +296,9 @@ s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + /* DMA transfer is really started, equeue the next buffer */ + s3c24xx_pcm_dma_started(substream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

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