September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Exposure of the input devices of Conexant CX20561 amplifier
by Matthias Bläsing 24 Sep '09

24 Sep '09

Hello, it would be nice if I could control the input channels of my lenovo laptop (Lenovo R500). These are configured for microphone input, but the amplifier can be deactivated, so that it can work with the higher line-in level. The resulting quality is not good, but at least usable. The volume control is joint, but the amplifier control are seperated. With the HDAAnalyser I could find the Nodes (0x17 (rear) and 0x18 (front)) can be used to control the amplification. The setting can be changed between 0 and 4, where 0 makes it possible to use the input for line-in, while 4 is better suited for microphone input. I'm not sure whether it would be better to expose this as a stepped control or just a switch (amplifier on/off). These are the loaded modules (from kernel 2.6.31): snd_usb_audio 74152 3 snd_usb_lib 13904 1 snd_usb_audio snd_hda_codec_conexant 17684 1 snd_seq_dummy 2748 0 snd_seq_oss 24620 0 snd_hda_intel 24064 9 snd_hda_codec 65724 2 snd_hda_codec_conexant,snd_hda_intel snd_seq_midi 6232 0 snd_hwdep 6772 3 snd_usb_audio,snd_hda_codec snd_rawmidi 19600 2 snd_usb_lib,snd_seq_midi snd_seq_midi_event 6580 2 snd_seq_oss,snd_seq_midi snd_pcm_oss 33516 0 snd_mixer_oss 13416 1 snd_pcm_oss snd_seq 44744 7 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_pcm 64872 6 snd_usb_audio,snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 18448 2 snd_seq,snd_pcm snd_seq_device 6424 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd_page_alloc 8556 2 snd_hda_intel,snd_pcm snd 52452 35 snd_usb_audio,snd_hda_codec_conexant,snd_seq_oss,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_rawmidi,snd_pcm_oss,snd_mixer_oss,snd_seq,snd_pcm,snd_timer,snd_seq_device soundcore 6876 1 snd I attached the /proc/asound/card0/codec#0 info to this email. If you need further information - please ask. Matthias

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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Fix divide by zero error during 1st execution
by Troy Kisky 24 Sep '09

24 Sep '09

When both playback and capture stream were open davinci_i2s_hw_params was setting parameters for the wrong stream. The fix for davinci_i2s_hw_params is sufficient, but it looks like a race still happens in davici_pcm_open. This patch also makes the race smaller but the next patch provides a better fix. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 3 ++- sound/soc/davinci/davinci-pcm.c | 12 +++++------- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..d32e197 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,8 +353,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 091dacb..002808b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; struct edmacc_param p_ram; int ret; - if (!dma_data) - return -ENODEV; - - prtd->params = dma_data; - /* Request master DMA channel */ ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, @@ -244,6 +237,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *params = rtd->dai->cpu_dai->dma_data; + if (!params) + return -ENODEV; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ @@ -257,6 +254,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = params; runtime->private_data = prtd; -- 1.5.6.3

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[alsa-devel] AIC23 playback volume control not working
by Aggarwal, Anuj 24 Sep '09

24 Sep '09

Hi, I am trying to change the playback volume using amixer commands but am not able to see any effect. Register writes are happening correctly but no change in volume. Here is a short log: root@omap3evm:~# amixer cset numid=1 0 tlv320aic23_write: reg=0x2, value=0x0 data[0]=0x4, data[1]=0x0 tlv320aic23_write: reg=0x3, value=0x0 data[0]=0x6, data[1]=0x0 numid=1,iface=MIXER,name='Digital Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=0,0 | dBscale-min=-121.00dB,step=1.00dB,mute=0 root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# amixer cset numid=1 127 tlv320aic23_write: reg=0x2, value=0x7f data[0]=0x4, data[1]=0x7f tlv320aic23_write: reg=0x3, value=0x7f data[0]=0x6, data[1]=0x7f numid=1,iface=MIXER,name='Digital Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=127,127 | dBscale-min=-121.00dB,step=1.00dB,mute=0 root@omap3evm:~# root@omap3evm:~# Any suggestions are appreciated. Regards, Anuj Aggarwal

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[alsa-devel] Capture not working in MMAP mode with '-v'
by Aggarwal, Anuj 24 Sep '09

24 Sep '09

Hi, When I try to capture audio in MMAP mode using the following command, it works properly: arecord -f cd -d 10 -M rec01.wav However, if I specify '-v' along with the above command, it fails to capture anything and prints log and returns immediately: Recording WAVE 'rec03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'omap3evm' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 1 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 mmap_area[0] = 0x4030e000,0,32 (16) mmap_area[1] = 0x4030e000,16,32 (16) I have not observed this problem while playing audio, only while capture this issue crops up. Any pointers? Thanks & Regards, Anuj Aggarwal

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Re: [alsa-devel] [PATCH - hda-intel conexant benq r55 laptop 1/1] [ALSA] CD-audio sound for hda-intel conexant benq laptop
by Takashi Iwai 24 Sep '09

24 Sep '09

At Tue, 22 Sep 2009 21:42:40 +0200, nowymarluk(a)wp.pl wrote: > > From: edoiks <edoiks(a)edoiks-laptop.(none)> > > After puting a cd-audio inside my laptop there was no sound out here, so I decided to install alsa-driver with debug level and setup a model=test, it didn't help, but then I look at source code and added this few lines, now cd-audio is working both when playback/recording. > > Signed-off-by: Lukasz Marcinowski <nowymarluk(a)wp.pl> > Signed-off-by: edoiks <edoiks(a)edoiks-laptop.(none)> Judging from your mail address, I stripped this edoiks line. Also, I fixed some strings for mixer elements and capture source items to follow the standard rule. E.g. "CD Audio xxx" is "CD xxx", and remove trailing spaces from the string. Now applied to sound git tree, and will be eventually merged to the upstream, too. thanks, Takashi > diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c > index 9d899ed..9afacb6 100644 > --- a/pci/hda/patch_conexant.c > +++ b/pci/hda/patch_conexant.c > @@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = { > }; > > static struct hda_input_mux cxt5045_capture_source_benq = { > - .num_items = 3, > + .num_items = 5, > .items = { > { "IntMic", 0x1 }, > { "ExtMic", 0x2 }, > { "LineIn", 0x3 }, > + { "CD Pin", 0x4 }, > + { "Mixer ", 0x0 }, > } > }; > > @@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static struct snd_kcontrol_new cxt5045_benq_mixers[] = { > + HDA_CODEC_VOLUME("CD Audio Capture Volume", 0x1a, 0x04, HDA_INPUT), > + HDA_CODEC_MUTE("CD Audio Capture Switch", 0x1a, 0x04, HDA_INPUT), > + HDA_CODEC_VOLUME("CD Audio Playback Volume", 0x17, 0x4, HDA_INPUT), > + HDA_CODEC_MUTE("CD Audio Playback Switch", 0x17, 0x4, HDA_INPUT), > + > HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > + HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), > + > {} > }; > > -- > 1.6.0.4 >

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 24 Sep '09

24 Sep '09

The following changes since commit 877ae70763fa0b19ffbe25e7e119fc96f52fbbd3: Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Chaithrika U S (1): ASoC: DaVinci: Correct McASP FIFO initialization Cliff Cai (1): ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Troy Kisky (2): ASoC: DaVinci: Fix divide by zero error during 1st execution ASoC: Davinci: Fix race with cpu_dai->dma_data sound/soc/blackfin/bf5xx-i2s.c | 8 ++-- sound/soc/blackfin/bf5xx-tdm.c | 8 ++-- sound/soc/davinci/davinci-i2s.c | 37 ++++++----------- sound/soc/davinci/davinci-mcasp.c | 80 +++++++++++++------------------------ sound/soc/davinci/davinci-mcasp.h | 7 +++- sound/soc/davinci/davinci-pcm.c | 13 +++--- sound/soc/davinci/davinci-pcm.h | 1 - 7 files changed, 60 insertions(+), 94 deletions(-)

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[alsa-devel] Audio problems with AD1981A (Subsystem: 0e11:00ad)
by Simos Xenitellis 24 Sep '09

24 Sep '09

Hi All, I am helping a user with the following hardware, http://www.alsa-project.org/db/?f=7f926200ee31b5b22222336b9e25bb0e60671a90 (Desktop computer Compaq Evo D510 SFF, with onboard sound card) lspci -vvc shows: 00:1f.5 0401: 8086:24c5 (rev 01) Subsystem: 0e11:00ad Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 17 Region 0: I/O ports at 2000 [size=256] Region 1: I/O ports at 2400 [size=64] Region 2: Memory at fc480400 (32-bit, non-prefetchable) [size=512] Region 3: Memory at fc480600 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: Intel ICH Kernel modules: snd-intel8x0 What does not work: 1. Jack sense. Audio plays on onboard speakers (the computer has in-built speakers) but does not work with headset. 2. Recording. Only a 'Capture' control is shown in the recording tab. No other controls for recording are available. When trying to record with gnome-sound-recorder, the application hangs, which then kills pulseaudio. Cheers, Simos

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[alsa-devel] Is there a way to initialize mixer elements?
by Ozan Çağlayan 24 Sep '09

24 Sep '09

Hi, Sometimes specific hardwares needs special tweaks (e.g. a mixer element is muted by default which avoids getting sound from HP, or the master volume's gain is very low, etc.). Do we have some kind of quirks (kernel-space/user-space) for that situations? As a real-world example, I have a user with a VIA VT1708B codec which is not able to get sound from HP unless he mutes the 'Independent HP' control: http://www.alsa-project.org/db/?f=4e7672aee15a64f67286a951e75e4168c6cd43d4 Thanks, Ozan Caglayan

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 23 Sep '09

23 Sep '09

The following changes since commit 9f072b7b2260a588bfb5fb43418c3358104e7c13: Mark Brown (1): Merge branch 'for-2.6.32' 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 Barry Song (2): ASoC: some minor changes for AD1836 and AD1938 codec drivers ASoC: Blackfin I2S: use dai state rather than local counter Chaithrika U S (1): ASoC: DaVinci: Correct McASP FIFO initialization Cliff Cai (1): ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Mark Brown (9): ASoC: Update WM8711 to driver model registration method Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into upstream/wm8711 ASoC: Factor out WM8711 cache I/O ASoC: Add SPI support to WM8711 ASoC: WM8711 minor cleanups ASoC: Add TLV information to WM8711 ASoC: Convert soc-cache to use C99 style initialisers for the table Merge branch 'upstream/wm8711' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Mike Arthur (1): ASoC: Add WM8711 CODEC driver Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo Troy Kisky (2): ASoC: DaVinci: Fix divide by zero error during 1st execution ASoC: Davinci: Fix race with cpu_dai->dma_data jassi brar (1): ASoC: Support WM8580 based audio subsystem on SMDK64xx machines sound/soc/blackfin/bf5xx-i2s.c | 23 +- sound/soc/blackfin/bf5xx-tdm.c | 8 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 3 +- sound/soc/codecs/ad1938.c | 2 +- sound/soc/codecs/wm8711.c | 658 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8711.h | 42 +++ sound/soc/codecs/wm8753.c | 1 + sound/soc/davinci/davinci-i2s.c | 37 +-- sound/soc/davinci/davinci-mcasp.c | 80 ++--- sound/soc/davinci/davinci-mcasp.h | 7 +- sound/soc/davinci/davinci-pcm.c | 13 +- sound/soc/davinci/davinci-pcm.h | 1 - sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 273 +++++++++++++++ sound/soc/soc-cache.c | 18 +- 18 files changed, 1068 insertions(+), 115 deletions(-) create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c

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[alsa-devel] [PATCH] ASoC: DaVinci: Correct McASP FIFO initialization
by Chaithrika U S 23 Sep '09

23 Sep '09

McASP write FIFO registers should be modified for playback and read FIFO registers for capture. Check the PCM mode before manipulating the FIFO registers. Currently, irrespective of playback/capture both the FIFOs are enabled or disbaled. This resulted in errors in audio loopback mode. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 36 ++++++++++++++++++------------------ 1 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7a06c0a..613fd98 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -386,17 +386,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_start_tx(dev); - else + } else { + if (dev->rxnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_start_rx(dev); - - /* enable FIFO */ - if (dev->txnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -413,17 +413,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_stop_tx(dev); - else + } else { + if (dev->rxnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_stop_rx(dev); - - /* disable FIFO */ - if (dev->txnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, -- 1.5.6

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