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[alsa-devel] [PATCH] ASoC: TWL4030: Replace comma with semicolon in probe function
by Peter Ujfalusi 13 Jan '10

13 Jan '10

The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a27f7b..74f0d65 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2192,7 +2192,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); -- 1.6.6

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[alsa-devel] [PATCH] ALSA: ASoC: ak4104: allow more sample rates
by Daniel Mack 13 Jan '10

13 Jan '10

The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4104.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 3a14c6f..b9ef7e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_32000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- 1.6.5.2

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[alsa-devel] Personal invitation from Kunal Narkhede
by Kunal Narkhede 13 Jan '10

13 Jan '10

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Improved MacBook (Pro) 5, 1 / 5, 2 support
by murray.alex＠gmail.com 13 Jan '10

13 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 files changed, 26 insertions(+), 2 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index f858dcd..bd65e47 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -7201,8 +7201,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7603,6 +7603,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7787,6 +7788,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9233,6 +9255,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- 1.6.3.3

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[alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 13 Jan '10

13 Jan '10

Hi, I am working on android and and in our device we use ALSA. I have enabled dmix support for andoid as by default android doesn't have this plugin enabled and soon will post it in android forum however i have run into some problems.Hoping takashi can help me out on this. Problem is when i am trying to use dmix plugin i am not getting any sound (logs shown below).Is it because i don't have mmap mode support on my device??? If yes then can't dmix work without mmap mode??? Logs shown below: Entered s3c24xx_pcm_hw_params, Entered s3c24xx_pcm_hw_params, line 330 Entered s3c24xx_pcm_mmap Entered s3c24xx_pcm_prepare Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer res_value[0] snd_pcm_update_hw_ptr_pos xun: [0]occured XRUN! Normal playback and recording is fine if i dont use dmix plugin.During my normal playback and recording i don't use mmap as i preallocate DMA memory and use them. Regards, anish

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[alsa-devel] [2.6.33-rc3] HDA works in polling mode
by Tvrtko Ursulin 13 Jan '10

13 Jan '10

Hi, I have a AMD 785G motherboard where there is a worrying situation logged by ALSA drivers. Sound nevertheless works. lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) dmesg: HDA Intel 0000:01:05.1: PCI INT B disabled HDA Intel 0000:00:14.2: PCI INT A disabled HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: irq 28 for MSI/MSI-X HDA Intel 0000:01:05.1: setting latency timer to 64 hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 I tried position_fix=0, 1 and 2 with no difference. I also tried enable_msi=0 to make it use IO-APIC interrupt. Then this error is not displayed, sound works, but again no interrupts are generated at least count is not growing in /proc/interrupts. Please tell me what else I can try and what debugging info you need. Thanks, Tvrtko

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[alsa-devel] No CD audio with kernel 2.6.32.3
by Chris Clayton 12 Jan '10

12 Jan '10

Hi, Please cc me on any reply - I'm not subscribed. Except for CD audio, sound is working fine on my new HP G61 laptop with kernel 2.6.32.3. I can play CD audio fine in Windows. IIRC, on my (very) old Dell laptop, alsamixer showed a CD audio device, but that's not the case on the G61. I have asla built as modules. Diagnostics: [chris:~]$ lsmod | grep snd snd_hda_codec_intelhdmi 10250 1 snd_hda_codec_idt 40146 1 snd_hda_intel 17040 1 snd_hda_codec 46119 3 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel snd_pcm 51424 2 snd_hda_intel,snd_hda_codec snd_timer 15133 1 snd_pcm snd 37459 8 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_pcm,snd_timer snd_page_alloc 5657 2 snd_hda_intel,snd_pcm [chris:~]$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: INTEL HDMI [INTEL HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 [chris:~]$ amixer controls numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=5,iface=MIXER,name='Headphone Playback Switch' numid=4,iface=MIXER,name='Headphone Playback Volume' numid=15,iface=MIXER,name='PCM Playback Volume' numid=3,iface=MIXER,name='Mic Jack Mode' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=11,iface=MIXER,name='IEC958 Playback Con Mask' numid=12,iface=MIXER,name='IEC958 Playback Pro Mask' numid=13,iface=MIXER,name='IEC958 Playback Default' numid=14,iface=MIXER,name='IEC958 Playback Switch' numid=8,iface=MIXER,name='Mux Capture Volume' numid=2,iface=MIXER,name='Speaker Playback Switch' numid=1,iface=MIXER,name='Speaker Playback Volume' [chris:~]$ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 Simple mixer control 'Speaker',0 Simple mixer control 'PCM',0 Simple mixer control 'Mic Jack Mode',0 Simple mixer control 'IEC958',0 Simple mixer control 'Capture',0 Simple mixer control 'Mux',0 Extract from dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 Let me know if there is any other information I can provide. Thanks Chris -- No, Sir; there is nothing which has yet been contrived by man, by which so much happiness is produced as by a good tavern or inn - Doctor Samuel Johnson

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[alsa-devel] X-Fi notebook support
by The Source 12 Jan '10

12 Jan '10

Hello. I have some info on that card, but not much. I have discovered that this card is detected connected via USB rather than PCI-E (the slot is ExpressCard). snd_ctxfi uses PCI interfaces right? Can X-Fi support be also added to snd_usb_audio? Or can snd_ctxfi support USB? lsusb -v output on that card is attached (sorry for duplicate mail, forgot to attach it first time). I now have this card and willing to help to test (or perhaps create) proper driver.

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 12 Jan '10

12 Jan '10

The following changes since commit 7f50548abb5454bd82c25aae15f0a3bf6a530f46: Mark Brown (1): Merge commit 'v2.6.33-rc2' into for-2.6.33 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 Manuel Lauss (1): ASoC: fixup oops in generic AC97 codec glue Mark Brown (1): ASoC: Fix WM8350 DSP mode B configuration sound/soc/codecs/ac97.c | 6 ++++++ sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 7 insertions(+), 1 deletions(-)

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[alsa-devel] hda-intel: no codecs found
by Ozan Çağlayan 12 Jan '10

12 Jan '10

Hi, Does that warning can be interpreted as a hardware failure? I'm reverting to older alsa-driver snapshots that I'm sure was working OK but still having this warning. I'm suspecting the power outage that the system suffered on Friday. Is there a way to understand whether it's driver related or HW related? Thanks, Ozan

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