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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 26 Sep '09

26 Sep '09

Hi Peter, I forgot to tell you that if I deliberately misconfigure the McBXP_CLKS pin so that it's an output, the FSX -> DACLRC/ADCLRC signal stops. This should indicate that the CLKOUT signal is actually making it to the CLKS pin on the OMAP. Right? Rick

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#527074: /etc/init.d/alsa-utils: `/etc/init.d/alsa-utils stop` triggers amixer error
by Elimar Riesebieter 26 Sep '09

26 Sep '09

* Piotr Lewandowski [090925 20:54 +0200] > Hi alsa-devel@! > > One of the maintainers of alsa-utils package in Debian advised me to > ask for your help. I've encountered following problem: > > #v+ > $ amixer -c 0 -q set Mic 0% mute > amixer: Invalid command! > #v- > > This command is invoked from /etc/init.d/alsa-utils script during > system shutdown. Do you have any ideas how to fix that? > > Here is some basic info about my hardware: > > #v+ > $ lspci -v | sed -ne '/Audio/,/^$/p' > 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High > Definition Audio Controller (rev 01) > Subsystem: Hewlett-Packard Company Device 30a2 > Flags: bus master, fast devsel, latency 0, IRQ 16 > Memory at f4580000 (64-bit, non-prefetchable) [size=16K] > Capabilities: <access denied> > Kernel driver in use: HDA Intel > > $ cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.18a. 1. It would be an option to insert: 'echo option "snd-hda-intel enable_msi=1"' >> /etc/modprobe.d/local.conf 2. It is a good idea to update to alsa-source 1.0.21-2 and build biuld your driver up from there. Elimar -- Never make anything simple and efficient when a way can be found to make it complex and wonderful ;-)

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[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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