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

[alsa-devel] [PATCH] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. Signed-off-by: Graeme Gregory <dp(a)xora.org.uk> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi Peter, Thanks very much for the help. > I think the problem is here. > The MCBSP_CLKS pin is connected to CLK256FS pin of TWL4030 (or TPA). > Since you are not using the TWL codec, there will be no clock on the MCBSP_CLKS > pin, thus no functional clock for McBSP, thus no bits are shifted in or out. > You can try to use the internal functional clock: > > ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 0, > SND_SOC_CLOCK_IN); I have checked the CLKOUT (WM8731) -> McBSP_CLKS (OMAP3) pin and it has a 12288000 clock on it. I've tried using the internal clock as you show above, just to see if I get anything out on DX but I get the same thing: DX just goes low for the "aplay" duration. Any idea why I'm not getting DMA interrupts? This would indeed cause no data at DX. omap2_dma_irq_handler() never gets called but omap_init_dma() and omap_start_dma() get called. Do you think there may be any issues with my DMA code (specifically arch/arm/plat-omap/dma.c). As I mentioned, I'm at "2.6.29-rc3-omap1" Thanks, Rick

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[alsa-devel] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

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[alsa-devel] Mic stops working after installation (ALC861)
by remrot＠gmail.com 25 Sep '09

25 Sep '09

/* Sorry for posting this to the devel list, but the problem seems to be trickier than I first thought * * and I don't have much time left to fix it (not my machine)! Thanks... */ Hi! I've got a very strange problem with a ALC861 soundchip on a friend's Toshiba A100-507 Laptop. Sound output and input work perfectly within a live session using the Ubuntu Jaunty and Karmic (Alpha6) CDs but immediately stops working after installing the system. Currently I've installed the latest Karmic containing kernel 2.6.31 and ALSA 1.0.20. I've attached the output of the alsa-info script as well as the output of 'pacmd list' for both systems (live and installed). Right after the installation the sound stopped working and I couldn't record anything. Alsamixer showed only output devices, no capture devices at all (everything has been correct within the live session). In 'dmesg' I found the following error messages: [ 279.924019] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x301f000a [ 280.928019] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x301f000a After doing some research I found this guide (thx Mr. Iwai): http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… And indeed, adding 'probe_mask=8' fixed the sound output and brought back the capture devices (remember that they were there within the live session without that option). I've also tried the 'model=toshiba' option (combined with 'probe_mask' and standalone) but that didn't help at all (no capture devices and no sound). The current situation: I've already posted to the PulseAudio mailing ist and got some help from Mr. Colin Guthrie but we haven't found a solution (everything seems OK regarding PA). I still can't record anything within the installed system. I tried audacity, alsarec and the gnome-sound-recorder without luck. When recording in audacity, the timeline cursor doesn't move, it only flickers. In gnome-recorder, recording stops right after it has started and the application hangs on exit. PA's volume meter doesn't show anything either, it is empty. I compared the configurations on both systems but I see no differences that are not related to the 'probe_mask' option... I really believe that there is a solution because everything works in a live session. Can anybody help me with this? Regards JK

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[alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi, I'm working on getting a WM8731 (in SLAVE) working using I2S on omap3 McBSP2 (in MASTER). Here is the hardware: http://www.efn.org/~rick/pub/wm8731.jpg Here are the PADCONF's and the pins they hook to on the WM8731: /*Audio Interface for csb740 */\ MUX_VAL(CP(McBSP2_FSX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_FSX -> I2S_FRM -> LRCLK -> DACLRC and ADCLRC*/\ MUX_VAL(CP(McBSP2_CLKX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_CLKX -> I2S_CLK -> I2S_BCLK -> BCLK */\ MUX_VAL(CP(McBSP2_DR), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP2_DR <- I2S_RXD <- ACDDAT */\ MUX_VAL(CP(McBSP2_DX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_DX -> I2S_TXD -> DACDAT */\ MUX_VAL(CP(McBSP_CLKS), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP_CLKS <- I2S_MCLK <- CLKOUT */\ My kernel is "2.6.29-rc3-omap1", here's what I see at bootup: 1 WM8731 Audio Codec 0.13<6>asoc: WM8731 <-> omap-mcbsp-dai-0 mapping ok 2 ALSA device list: 3 #0: omap3csb740 (WM8731) ... 4 beagle:~# aplay test48000.wav 5 Playing WAVE 'test48000.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo 6 aplay: pcm_write:1269: write error: Input/output error I get no audio output, other than a "click" when the 8731 is initialized upon boot. When I look at the signals the McBSP2_DX line just goes low for the duration of the "aplay", it does not toggle. The other lines look fine. The file is at http://www.efn.org/~rick/pub/omap3csb740.c but roughly, here's what I'm doing: static struct snd_soc_dai_link omap3csb740_dai = { ... .cpu_dai = &omap_mcbsp_dai[0], /* use 1st one */ static int __init omap3csb740_soc_init(void) ... *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ *(unsigned int *)omap3csb740_dai.cpu_dai->private_data = 1; /* McBSP2 debug only? */ static int omap3csb740_hw_params(struct snd_pcm_substream *substream, ... ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 256); Any help is greatly appreciated. Rick Bronson

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[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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Re: [alsa-devel] sound issues on an HP Pavilion DV3-1075 (solved)
by Liviu Andronic 23 Sep '09

23 Sep '09

On 9/23/09, Daniel Chen <seven.steps(a)gmail.com> wrote: > Yes, both of these options are necessary. What's your lspci -nv|grep -A1 0403? > debian-liv:/home/liviu# lspci -nv|grep -A1 0403 00:14.2 0403: 1002:4383 Subsystem: 103c:1506 -- 01:05.1 0403: 1002:960f Subsystem: 1002:960f Best Liviu

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[alsa-devel] [PATCH 1/2] ASoC: Blackfin I2S: use dai state rather than local counter
by Mike Frysinger 23 Sep '09

23 Sep '09

From: Barry Song <barry.song(a)analog.com> Since the active field of the dai already tells us the stream activity, the local counter variable is redundant and can be replaced. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-i2s.c | 15 +-------------- 1 files changed, 1 insertions(+), 14 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abad..66e7333 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -49,7 +49,6 @@ struct bf5xx_i2s_port { u16 rcr1; u16 tcr2; u16 rcr2; - int counter; int configured; }; @@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - - /*this counter is used for counting how many pcm streams are opened*/ - bf5xx_i2s.counter++; - return 0; -} - static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); - bf5xx_i2s.counter--; /* No active stream, SPORT is allowed to be configured again. */ - if (!bf5xx_i2s.counter) + if (!dai->active) bf5xx_i2s.configured = 0; } @@ -288,7 +276,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, -- 1.6.5.rc1

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[alsa-devel] sound issues on an HP Pavilion DV3-1075 (solved)
by Liviu Andronic 23 Sep '09

23 Sep '09

Dear ALSA devels I am writing on an ALSA-related issue that I managed to solve, on Debian testing with kernel 2.6.30 and ALSA 1.0.21. For background and exact symptoms please see these two threads. [1] [2] The first issue was that there was no sound. I solved it by manually configuring it as follows: debian-liv:/home/liviu# cat /etc/modprobe.d/sound ## ALSA portion alias char-major-116 snd alias snd-card-0 snd-hda-intel alias snd-card-1 snd-usb-audio ## module options should go here options snd-hda-intel index=0 model=hp-dv5 enable_msi=1 options snd-usb-audio index=1 The second issue concerned the external speakers/headphones: when such devices were plugged in, the internal speakers were not muted and the audio stream was not redirected to the external headset (that is, there was no sound in the headphones); although I didn't check, the microphone was probably not working, too. The issue was solved by adding position-fix=1. Thus, the following line was used options snd-hda-intel index=0 model=hp-dv5 enable-msi=1 position-fix=1 I have similar issues on Ubuntu Jackalope, and it seems that on Ubuntu it is quite often that audio cards on HP Pavilions misbehave by default (no sound, no sound in headphones, etc.). Best regards Liviu [1] http://www.mail-archive.com/debian-user@lists.debian.org/msg555984.html [2] http://www.pubbs.net/debian/200909/102780/ ### Debug messages ### liviu@debian-liv:~$ cat /proc/asound/card*/codec* | grep Codec Codec: IDT 92HD71B7X Codec: ATI RS690/780 HDMI liviu@debian-liv:~$ cat /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xd2400000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xd2310000 irq 29 liviu@debian-liv:~$ cat /proc/asound/devices 0: [ 0] : control 1: : sequencer 4: [ 0- 0]: hardware dependent 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 32: [ 1] : control 33: : timer 36: [ 1- 0]: hardware dependent 51: [ 1- 3]: digital audio playback liviu@debian-liv:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 liviu@debian-liv:~$ aplay -L default:CARD=SB HDA ATI SB, STAC92xx Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) hdmi:CARD=HDMI HDA ATI HDMI, ATI HDMI HDMI Audio Output -Computer- Processor : 2x AMD Turion(tm) X2 Dual-Core Mobile RM-72 Memory : 3802MB (907MB used) Operating System : Debian GNU/Linux squeeze/sid User Name : liviu (Liviu) Date/Time : Tue 22 Sep 2009 12:58:23 BST -Display- Resolution : 1280x800 pixels OpenGL Renderer : ATI Radeon HD 3200 Graphics X11 Vendor : The X.Org Foundation -Multimedia- Audio Adapter : HDA-Intel - HDA ATI SB Audio Adapter : HDA-Intel - HDA ATI HDMI -Input Devices- Macintosh mouse button emulation AT Translated Set 2 keyboard Video Bus Power Button Power Button Lid Switch PC Speaker ST LIS3LV02DL Accelerometer HP Webcam SynPS/2 Synaptics TouchPad HDA Digital PCBeep HDA ATI SB Mic at Ext Front Jack HDA ATI SB HP Out at Ext Front Jack -Printers (CUPS)- psc_1200_first -Version- Kernel : Linux 2.6.30-1-amd64 (x86_64) Compiled : #1 SMP Sat Aug 15 18:09:19 UTC 2009 C Library : GNU C Library version 2.9 (stable) Default C Compiler : GNU C Compiler version 4.3.4 (Debian 4.3.4-2) Distribution : Debian GNU/Linux squeeze/sid -Current Session- Computer Name : debian-liv User Name : liviu (Liviu) Home Directory : /home/liviu Desktop Environment : XFCE 4 -Misc- Uptime : 2 hours, 19 minutes Load Average : 0.57, 0.67, 0.73 liviu@debian-liv:~$ lspci -vv 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Subsystem: Hewlett-Packard Company Device 1506 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin ? routed to IRQ 16 Region 0: Memory at d2400000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel [..] 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller Subsystem: ATI Technologies Inc RS780 Azalia controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 19 Region 0: Memory at d2310000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel liviu@debian-liv:~$ lsmod | grep -i snd snd_hda_codec_atihdmi 3952 1 snd_hda_codec_idt 62480 1 snd_hda_intel 26680 9 snd_hda_codec 75248 3 snd_hda_codec_atihdmi,snd_hda_codec_idt,snd_hda_intel snd_hwdep 8152 1 snd_hda_codec snd_pcm_oss 37200 0 snd_mixer_oss 15072 2 snd_pcm_oss snd_pcm 78504 4 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_midi 6976 0 snd_rawmidi 23008 1 snd_seq_midi snd_seq_midi_event 7712 1 snd_seq_midi snd_seq 51392 2 snd_seq_midi,snd_seq_midi_event snd_timer 21824 3 snd_pcm,snd_seq snd_seq_device 7476 3 snd_seq_midi,snd_rawmidi,snd_seq snd 63912 25 snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 7984 2 snd snd_page_alloc 10000 2 snd_hda_intel,snd_pcm -- Do you know how to read? http://www.alienetworks.com/srtest.cfm Do you know how to write? http://garbl.home.comcast.net/~garbl/stylemanual/e.htm#e-mail -- Do you know how to read? http://www.alienetworks.com/srtest.cfm Do you know how to write? http://garbl.home.comcast.net/~garbl/stylemanual/e.htm#e-mail

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[alsa-devel] [PATCH 1/4] ASoC: ALSA ASoC CQ0093 Voice Codec Driver
by miguel.aguilar＠ridgerun.com 22 Sep '09

22 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds support for the Voice Codec CQ0093. This driver was tested on a DM355 EVM rev C. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 377 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 23 +++ 4 files changed, 406 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..55d1287 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_CQ0093VC if ARCH_DAVINCI_DM365 select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..a3e713c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..e64af25 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,377 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +#define AUDIO_NAME "cq0093" +#define CQ93VC_VERSION "0.1" + +/* + * Voice codec registers + * + * The base address for the registers used by this driver is shifted + * 0x80 from the original Voice Codec base adress since the first region + * of the Voice Codec registers is used by the Voice Codec Interface driver. + */ +#define VC_REG05 0x14 /* PGA and MIC Gain Control */ +#define VC_REG09 0x24 /* Digital Soft Mute/Attenuation Control */ +#define VC_REG12 0x30 /* Voice Codec Up/Down */ + +/* Registers bits */ +#define PGA_GAIN 0x07 +#define DIG_ATTEN 0x3F +#define MUTE 0x40 +#define POWER_ALL_ON 0xFD + +/* Codec private data */ +struct cq93vc_priv { + void __iomem *base; + unsigned int sysclk; + resource_size_t pbase; + size_t base_size; +}; + +/* + * Read to the cq0093 register space + */ +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct cq93vc_priv *cq93vc = codec->private_data; + + return __raw_readb(cq93vc->base + reg); +} + +/* + * Write to the cq0093 register space + */ +static int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct cq93vc_priv *cq93vc = codec->private_data; + + __raw_writeb(value, cq93vc->base + reg); + return 0; +} + +static int cq93vc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, VC_REG09) & ~MUTE; + + if (mute) + cq93vc_write(codec, VC_REG09, reg | MUTE); + else + cq93vc_write(codec, VC_REG09, reg); + + return 0; +} + +static int cq93vc_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(cq93vc_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&cq93vc_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cq93vc_priv *cq93vc = codec->private_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + cq93vc->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* all power is driven by DAPM system */ + cq93vc_write(codec, VC_REG12, POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* + * all power is driven by DAPM system, + * so output power is safe if bypass was set + */ + cq93vc_write(codec, VC_REG12, 0x0); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, VC_REG12, 0x0); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .hw_params = cq93vc_hw_params, + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "cq93vc", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_device *cq93vc_socdev; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct cq93vc_priv *cq93vc; + struct snd_soc_codec *codec; + struct resource *res, *mem; + int ret; + + dev_info(dev, "CQ0093 Voice Codec %s\n", CQ93VC_VERSION); + + cq93vc = kzalloc(sizeof(struct cq93vc_priv), GFP_KERNEL); + if (cq93vc == NULL) { + dev_dbg(dev, "%s: could not allocate memory for private data\n", + pdev->name); + return -ENOMEM; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(dev, "%s: could not allocate memory for codec data\n", + pdev->name); + ret = -ENOMEM; + goto fail1; + } + + res = cq93vc_resources; + + cq93vc->pbase = res->start; + cq93vc->base_size = resource_size(res); + + mem = request_mem_region(cq93vc->pbase, + cq93vc->base_size, "cq93vc"); + if (!mem) { + dev_dbg(dev,"%s: voice codec registers at %08x are not free\n", + pdev->name, cq93vc->pbase); + ret = -EBUSY; + goto fail2; + } + + cq93vc->base = ioremap(cq93vc->pbase, cq93vc->base_size); + if (cq93vc->base == NULL) { + dev_dbg(dev,"%s: can't ioremap mem resource.\n", + pdev->name); + ret = -ENOMEM; + goto fail3; + } + + mutex_init(&codec->mutex); + codec->private_data = cq93vc; + codec->name = "cq93vc"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + socdev->card->codec = codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + goto fail4; + } + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, VC_REG05, PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, VC_REG09, DIG_ATTEN); + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Register sound card */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(dev, "%s: failed to register card\n", pdev->name); + goto fail5; + } + + cq93vc_socdev = socdev; + + return 0; +fail5: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +fail4: + iounmap(cq93vc->base); +fail3: + release_mem_region(cq93vc->pbase, cq93vc->base_size); +fail2: + kfree(codec); +fail1: + kfree(cq93vc); + + return ret; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct cq93vc_priv *cq93vc = codec->private_data; + + /* Power down chip */ + if (codec->control_data) + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + iounmap(cq93vc->base); + release_mem_region(cq93vc->pbase, cq93vc->base_size); + + kfree(codec); + kfree(cq93vc); + + cq93vc_socdev = NULL; + + snd_soc_unregister_dai(&cq93vc_dai); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .suspend = cq93vc_suspend, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_init(void) +{ + return snd_soc_register_dai(&cq93vc_dai); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + snd_soc_unregister_dai(&cq93vc_dai); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..8d37763 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; +extern struct resource cq93vc_resources[]; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH 4/4] DaVinci: DM365: Add voice codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 22 Sep '09

22 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables the voice codec support for the DM365 EVM. For the DM365 EVM only one codec is able to be enabled at time, external (AIC3101) or internal (Voice Codec CQ0093). Either internal or external codec can be selected in the configuration menu. This patch was created based on the temp/asoc branch of the DaVinci git and it assumes that the patch "[PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM" is applied. This patch was tested on the DM365 EVM rev C. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 +++ sound/soc/davinci/davinci-evm.c | 42 +++++++++++++++++++++++++++++- 2 files changed, 44 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 77f815c..083b8d7 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -482,7 +482,11 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) dm365_init_asp(&dm365_evm_snd_data); +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + dm365_init_vcif(&dm365_evm_snd_data); +#endif } static __init void dm365_evm_irq_init(void) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..e826a80 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,9 +28,11 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/cq93vc.h" #include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-vcif.h" #include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -151,6 +153,15 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +#ifdef CONFIG_SND_DM365_INTERNAL_CODEC +static struct snd_soc_dai_link cq93vc_evm_dai = { + .name = "CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, +}; +#endif + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +188,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +196,19 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver external or internal */ +static struct snd_soc_card dm365_snd_soc_card_evm = { +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) + .name = "DaVinci DM365 EVM", + .dai_link = &evm_dai, +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + .name = "On-chip Voice Codec", + .dai_link = &cq93vc_evm_dai, +#endif + .platform = &davinci_soc_platform, + .num_links = 1, +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +241,17 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + .codec_dev = &soc_codec_dev_cq93vc, +#endif +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +279,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; -- 1.6.0.4

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[alsa-devel] [PATCH 3/4] DaVinci: DM365: Add resources for voice codec and platform device for VCIF
by miguel.aguilar＠ridgerun.com 22 Sep '09

22 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Add the resources needed by the CQ0093 voice codec. 2) Add platform device for the voice codec interface. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/dm365.c | 46 ++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/dm365.h | 5 +++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 3 files changed, 53 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index 2674438..80dc35b 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -605,6 +605,8 @@ INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif @@ -835,6 +837,42 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vcif_resources[] = { + /* The first are of Voice Codec registers before 0x80 belongs to VCIF */ + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + (SZ_1K>>4) - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vcif_device = { + .name = "voice_codec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vcif_resources), + .resource = dm365_vcif_resources, +}; + +struct resource cq93vc_resources[] = { + /* The base address for the Voice Codec registers is shifted by 0x80 */ + { + .start = DAVINCI_DM365_VC_BASE + 0x80, + .end = DAVINCI_DM365_VC_BASE + 0x80 + (SZ_1K>>3) - 1, + .flags = IORESOURCE_MEM, + }, +}; +EXPORT_SYMBOL_GPL(cq93vc_resources); + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -950,6 +988,14 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) platform_device_register(&dm365_asp_device); } +void __init dm365_init_vcif(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_EVT2_VC_TX); + davinci_cfg_reg(DM365_EVT3_VC_RX); + dm365_vcif_device.dev.platform_data = pdata; + platform_device_register(&dm365_vcif_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 2291c0d..b78ec55 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -25,7 +25,12 @@ #define DM365_EMAC_MDIO_OFFSET (0x4000) #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vcif(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index 7732832..b01b5c4 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -311,6 +311,8 @@ enum davinci_dm365_index { DM365_INT_NSF_DISABLE, /* EDMA event muxing */ + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, DM365_EVT26_MMC0_RX, -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 22 Sep '09

22 Sep '09

The following changes since commit 0c31cf3e4af79ea18bbd365b07ef0de207673894: Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration 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 Barry Song (1): ASoC: some minor changes for AD1836 and AD1938 codec drivers Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- sound/soc/codecs/wm8753.c | 1 + 3 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [PATCH 0/4] Add Voice Codec support for DM365
by Miguel Aguilar 22 Sep '09

22 Sep '09

Hi, The following series of patches adds the following support: 1) CQ0093 Voice Codec driver. 2) Voice Codec Interface. 3) Voice Codec resources and Voice Codec Interface platform device. 4) Voice Codec on the DM365 EVM. Notes: The patch series was created based on the current temp/asoc branch of the DaVinci git. It assumes that the patch "[PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM" is applied. This patch series was tested on a DM365 EVM rev C. Regards, -- Miguel Angel Aguilar Ulloa Embedded Software Engineer RidgeRun Embedded Solutions miguel.aguilar(a)ridgerun.com

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[alsa-devel] (AT91RM2000 - ARM) Driver that puts data over SSC to FPGA
by Michel Müller 22 Sep '09

22 Sep '09

Hello, I am trying to write an ALSA-driver that puts the audio data over the SSC channel to the FPGA. This FPGA routes the Data to the AD/DA chips. I tried to figure out how I can do this with the ASoC Interfaces but I am currently stuck. Currently there is an Atmel SSC and PCM Interface that I can probably use. If I want to use the ASoC API I need to have a Codec Driver. But I don't have direct access to the codec. How should I proceed? I have 2 possibilities I think: Writing a Codec Driver / Writing an Alsa Driver as described in the guide of Takashi Iwai. If I write a character device that transfers data to the FPGA over the SSC I can hear sound, but I want to use ALSA because the use of audio applications will be easy and flexible. Thank you in advance! Regards, Michel

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Re: [alsa-devel] Application and hw mixing for VoIP (modified)
by Takashi Iwai 22 Sep '09

22 Sep '09

At Mon, 21 Sep 2009 15:48:39 +0200, Fabien COMTE wrote: > > Hi, > I need some help please. > I have an application that does VoIP and alsa interface. > > I want to : > - send through VoIP application a mixed sound from mic and an another > application (aplay should be ok for testing). > > mic ---------- soundcard 0 -------------> > mix ---- alsa interface -----> > voip application > application 1 --- alsa interface ------> > (aplay) > > - use sound received from VoIP with HP output and an another application > (arecord should be ok for testing). > > ----- alsa interface > ------> application 2 (arecord) > voip application ----- alsa interface ----> share > ------ soundcard 0 > --------> HP output > > > And a little question : > How to define an application interface in asoundrc ? Well, this isn't so trivial as it sounds. If the application 1 (aplay) needs to be output actually, it means to split the signal to two interfaces, one to voip program, and another to the output channel. If app1 doesn't have to be heard, then you can use a loopback device (snd-aloop driver included in alsa-driver tree) and bind with the mic device using multi plugin. The latter case, split from voip to app2 and HP output, can be also done via multi plugin + aloop device. But, this kind of infrastructure could be better done with a proper daemon, such as JACK, I guess... Takashi

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