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[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: Burst mode BCLK divider fixes
by Peter Ujfalusi 20 Jan '10

20 Jan '10

Hello, The following patches provides fixes for the BCLK clock divider configuration in tlv320dac33 codec. BCLK divider is used to specify the speed of the serial bus. The speed of the bus can be (must be) faster than the actual stream's sample rate in order to use the system in burst mode. With the introduction of mode7 FIFO mode, the BCLK handling got broken in this mode, the first patch fixes this issue. The second patch adds support for configuring the burst mode BCLK through platform data. --- Peter Ujfalusi (2): ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 22 +++++++++++++++++----- 2 files changed, 18 insertions(+), 5 deletions(-)

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[alsa-devel] Can we add optional parameters in snd_pcm_hw_params
by Raul Xiong 20 Jan '10

20 Jan '10

Hi all, In my current ALSA driver, I have defined a mode which will impact the configuration of the audio driver. The requirment is to ask application users to set this mode to driver. So I want to add a parameter in 'hwparams' of snd_pcm_hw_params(pcm_handle, hwparams). But it seems that ALSA have no such expansibility. Basically, my requirment is: There are two modes: ModeA and ModeB. I'd like application to set this mode before it begin playing. If you have other better methods to reach this requirment, it's also welcome. Thanks.

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[alsa-devel] State of the bugtracker
by Colin Guthrie 20 Jan '10

20 Jan '10

Hi, I remember a while back I asked about the bugtracker over at https://bugtrack.alsa-project.org/alsa-bug/my_view_page.php At the time it was generally regarded that the bugtracker was a bit of a black hole. I've generally avoided using it to report bugs, but I'd like to ask what it's current state is as I often have to direct users to *something* and generally go for the mailing list. So really this is just a "what is the state of the bug tracker" question. Do the main devs check it and triage it or is there no specific process? Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] No alsa-driver snapshots since 2009-12-19?
by Aleksander Adamowski 20 Jan '10

20 Jan '10

Hi! Seems that alsa-driver snapshots have stopped being updated on http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ - the last one is from 2009-12-19. What happened? -- Best Regards, Aleksander Adamowski http://olo.org.pl

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[alsa-devel] [PATCH] Fix automute issue on HP T5735
by Lee Trager 20 Jan '10

20 Jan '10

The following patch fixes an issue where when you plug something into the headphone jack the internal speaker does not automute. This also will close alsa bug #0004766 Lee --- alsa-driver-orig/alsa-kernel/pci/hda/patch_realtek.c 2009-08-28 07:41:39.000000000 -0400 +++ alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-10-13 14:24:40.000000000 -0400 @@ -10028,7 +10028,9 @@ struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + spec->autocfg.speaker_pins[0] = 0x14; + + alc_automute_amp(codec); } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {

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Re: [alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by Miguel Aguilar 19 Jan '10

19 Jan '10

Hi Mark, I have been trying to test the AIC3x and the Voice Codec in the same kernel, however only the AIC3x is working, I'm having problem when I try to use the Voice Codec, I already have verified that the Voice Codec works since I tested alone. Please check the following output: root@dm365-evm:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: EVM [DaVinci DM365 EVM], device 0: AIC3X tlv320aic3x-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: EVM [DaVinci DM365 EVM], device 1: CQ93 CQ93VC-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 root@dm365-evm:~# aplay -L default:CARD=EVM DaVinci DM365 EVM, Default Audio Device null Discard all samples (playback) or generate zero samples (capture) root@dm365-evm:~# cat /proc/asound/pcm 00-00: AIC3X tlv320aic3x-0 : : playback 1 : capture 1 00-01: CQ93 CQ93VC-1 : : playback 1 : capture 1 root@dm365-evm:~# cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 33: : timer root@dm365-evm:~# cat /proc/asound/cards 0 [EVM ]: tlv320aic3x - DaVinci DM365 EVM DaVinci DM365 EVM (tlv320aic3x) root@dm365-evm:~# cat /proc/asound/timers G0: system timer : 10000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-1-0: PCM playback 0-1-0 : SLAVE P0-1-1: PCM capture 0-1-1 : SLAVE root@dm365-evm:~# arecord -D plughw:0,0 test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono *** This is the error that I have. I not sure if it has to do with the root@dm365-evm:~# arecord -D plughw:0,1 test.wav asoc: can't open platform davinci-audio arecord: main:608: audio open error: No such device The following is the patch to the davinci-evm.c please check it if you see something wrong there: diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..951d730 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -29,8 +29,10 @@ #include "../codecs/tlv320aic3x.h" #include "../codecs/spdif_transciever.h" +#include "../codecs/cq93vc.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 +153,23 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm365_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, + }, +}; + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +196,7 @@ 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 +204,14 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &dm365_evm_dai, + .num_links = ARRAY_SIZE(dm365_evm_dai), +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +244,13 @@ 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, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +}; + +/* 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 +278,15 @@ 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; Thanks, Miguel Aguilar

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[alsa-devel] [PATCH] ASoC: fix compile breakage - add a missing header include
by Guennadi Liakhovetski 19 Jan '10

19 Jan '10

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- It might well have been reported and / or fixed in some trees, sorry for the noise, if that's the case. include/sound/soc-dai.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ca24e7f..061f16d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -16,6 +16,8 @@ #include <linux/list.h> +#include <sound/soc.h> + struct snd_pcm_substream; /* -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: tlv320dac33: Correct the prefill number of samples
by Peter Ujfalusi 19 Jan '10

19 Jan '10

Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2df9c20..65683aa 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -559,7 +559,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(20)); + DAC33_THRREG(10)); break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", -- 1.6.6

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[alsa-devel] About audio driver architecture
by Michael Qiu 19 Jan '10

19 Jan '10

Hi, I'm going to write an audio driver for a developping soc used for dvb set top box. As I've never done such things before, so I've some questions. 1. Seems the "ALSA Driver Documentation" is mainly for PCI(and for PC) based audio card. Where can I got a document for SOC which has no PCI bus. If there are some existing drivers it will be even better. 2. Is there any standard for such a audio driver which upper applications will depend on it? Or I can just follow ALSA? Best regards Michael Qiu

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Re: [alsa-devel] snd-hdsp driver error?
by Raymond Yau 19 Jan '10

19 Jan '10

2010/1/16 Lars Zobbe <fedesensfar(a)gmail.com> > Thansk a lot Raymond. That actually works! My HDSP now loads without mayor > problems. This is brilliant. I've been struggling for so long to make this > work and finally... > I recommend everyone with a RME HDSP, who wants to use pulseaudio, to check > out ticket 764. > Thanks again Raymond. > Lars. > > 2010/1/14 Raymond Yau <superquad.vortex2(a)gmail.com> > >> >> Refer to >> >> http://pulseaudio.org/ticket/764#comment:1 >> >> >> load-module module-alsa-sink device_id=0 tsched=0 >> > load-module module-alsa-source device_id=0 tsched=0 >> > if your hdsp works , this mean that this is a bug introduced at pulseaudio since 0.9.15 ( working profile ) I guess none of PA develpers had your card , if you want the pulseaudio bug fixed , you will need to post the output of full pulseaudio log of module-alsa-sink ( pulseaudio -k;pulseaudio -vvvvv ) and the outout of aplay -v -Dplughw:0,0 any.wav in the pulseaudio bug tracker so that PA developer can compare with the non-working pulseaudio log

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