January 2010 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Reg: Audio recording with twl4030 codec
by Peter Ujfalusi 25 Jan '10

25 Jan '10

Hello, Let's put alsa-devel back to the CC... On Monday 25 January 2010 09:32:45 ext shariff wrote: > Hi Peter, > > 1) As we only have one Main MIC when I am recording in stereo mode by > specifying "-c 2", one channel is silent and one channel is recorded > properly. As it should. Just as a sidenote, I'm usually using -f dat for the format, which gives 48KHz, stereo 16LE > > 2) But still the recorded channel is noisy, how can I reduce the noise of > the recorded channel? Hmm, what kind of noise? How are you configuring the dai with the snd_soc_dai_set_fmt? For stereo it should be (if OMAP is slave, TWL is master): SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; For TDM mode (four channel): SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM; See omap3beagle.c how it is done. > > I have one more question. ( Regarding Audio path) > > ---> If I want to avoid that one channel silent, do I need to connect "SUB > MIC or AUXR"?( as these are connected right ADC). If you want audio data there, than yes you need another mic for the right channel. > ----> If I need to record 4 channel audio without any silent channel do I > need to connect digital microphones also? You can duplicate the channel 1 and 3 to channel 2 and 4, meaning that you route the ADCL and ADCR to these channels (TX1 and TX2 path are both analog). You can also hook up a stereo digital mic to the digimic1 interface, but remember, that the digimic needs 48KHz sampling rate. -- Péter > > Regards, > Shariff > > On Fri, Jan 22, 2010 at 6:29 PM, shariff > <shariffsinbox(a)gmail.com<mailto:shariffsinbox@gmail.com>> wrote: Hi Peter, > > Thank you very much for your suggestions. > > Regards, > Shariff > > > On Fri, Jan 22, 2010 at 6:11 PM, Peter Ujfalusi > <peter.ujfalusi(a)nokia.com<mailto:peter.ujfalusi@nokia.com>> wrote: Hello, > > On Friday 22 January 2010 11:57:23 ext shariff wrote: > > Hi all, > > > > I am working with OMAP3 based custom board that has twl4030 codec > > & using linux-2.6.29 kernel from opensource. > > This is kind of old... > > > I have some queries regarding Audio recording. > > > > Audio codec is connected to twl4030 as bellow > > > > TWL4030 > > > > |--------------------| > > | > > | |------ MAINMIC > > > > McBSP2 ------ |i2s/tdm | > > > > | ___________| > > > > --> One Analog MIC is connected to Main Mic and I am using Audio > > path not the Voice path. > > > > After selecting Main Mic through amixer control. > > > > and recording audio using "arecord -c 1 -r 8000 -f S16_LE -t > > wav -d 20 record-file.wav" > > The hardware should have been opened in stereo mode, since it is not > supporting the mono, you can verify it by appending -v to arecord and see > how the HW has been opened at the end. > > > --> Here record is successful but with lot of noise. > > > > 1) Can we record Mono(1 channel) through I2S from TWL4030 > > codec? > > The TWL codec "Audio" path is either stereo or 4 channel. > You need to record stereo/4 channel, and than extract/use the data from the > valid channel. > In case of Main mic it is the left channel, or Channel 1 or/and 2 > (depending on the routings) in four channel mode. > > > 2) To record stereo(2 channels), do we need two Mic's ? (i.e do > > I need Sub Mic also). > > No you don't need. The right channel will not contain valid data (most > likely it is going to be silent). > > You need to enable the Mic bias 1 for the main mic in your machine driver, > if you have not done that yet: > > static const struct snd_soc_dapm_widget your_widgets[] = { > ... > SND_SOC_DAPM_MIC("Your Mic", NULL), > ... > }; > > static const struct snd_soc_dapm_route your_audio_map[] = { > ... > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Your Mic"}, > ... > }; > > I'm not sure about the state of the twl soc codec driver in 2.6.29, but my > guess is that it should work. > > Are there any capture routing option for TX1 path, which selects between > Analog and Digimic0? If there is than that has to be set to analog as > well. > > > Regards, > > Shariff > > -- > Péter >

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 25 Jan '10

25 Jan '10

The following changes since your topic/asoc branch are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Since they include (via the 2.6.33 merge) a merge of Linus' tree I've only included the changes that aren't already in your tree or mainline in the summary below, though it looks like the diffstat picked up some extra changes due to the way I tried to do that. Barry Song (4): ASoC: ad1836: reset and restore clock control mode in suspend/resume entry ASoC: ad1938: fix typo, rename mask to rx_mask for ad1938_set_tdm_slot ASoC: ad1938: let soc-core dapm handle PLL power ASoC: soc-cache: cleanup training whitespace and coding style Ilkka Koskinen (2): ASoc: tpa6130a2: Remove unnecessary variable ASoC: tpa6130a2: Support for tpa6140's regulators Kuninori Morimoto (3): ASoC: fsi: Add over_period flag to prevent the misunderstanding ASoC: fsi: Add fsi_get_dai to get snd_soc_dai ASoC: fsi: Add over/under run error settlement Mark Brown (9): ASoC: Remove unneeded suspend checks from CODEC drivers ASoC: Use snprintf() when generating stream names ASoC: Fix variable shadowing warning in TLV320AIC3x ASoC: Only restore non-default registers for WM8731 ASoC: Only restore non-default registers for WM8776 ASoC: Only restore non-default registers for WM8961 ASoC: Implement suspend and resume for WM8993 ASoC: Fix WM8350 DSP mode B configuration Merge branch 'for-2.6.33' into for-2.6.34 Peter Ujfalusi (5): ASoC: tlv320dac33: Change nsample switch to FIFO mode enum ASoC: tlv320dac33: Introduce prefill and playback state handlers ASoC: tlv320dac33: Clean up the hardware configuration code ASoC: tlv320dac33: Add new FIFO mode: mode 7 ASoC: tlv320dac33: Safety check for codec slave mode include/sound/soc.h | 3 + include/sound/tpa6130a2-plat.h | 6 + include/sound/wm8904.h | 57 + include/sound/wm8955.h | 26 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ad1836.c | 32 + sound/soc/codecs/ad1938.c | 64 +- sound/soc/codecs/cs4270.c | 43 +- sound/soc/codecs/da7210.c | 589 +++++++++ sound/soc/codecs/da7210.h | 24 + sound/soc/codecs/tlv320aic3x.c | 75 +- sound/soc/codecs/tlv320dac33.c | 282 ++++- sound/soc/codecs/tpa6130a2.c | 109 ++- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8727.c | 66 +- sound/soc/codecs/wm8731.c | 3 + sound/soc/codecs/wm8753.c | 8 - sound/soc/codecs/wm8776.c | 2 + sound/soc/codecs/wm8904.c | 2556 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8904.h | 1681 ++++++++++++++++++++++++ sound/soc/codecs/wm8955.c | 1151 +++++++++++++++++ sound/soc/codecs/wm8955.h | 489 +++++++ sound/soc/codecs/wm8961.c | 3 + sound/soc/codecs/wm8990.c | 8 - sound/soc/codecs/wm8993.c | 67 + sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 120 +-- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 ++ sound/soc/sh/fsi.c | 164 ++- sound/soc/soc-cache.c | 2 +- sound/soc/soc-core.c | 11 +- sound/soc/soc-dapm.c | 37 +- 37 files changed, 7457 insertions(+), 359 deletions(-) create mode 100644 include/sound/wm8904.h create mode 100644 include/sound/wm8955.h create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h create mode 100644 sound/soc/codecs/wm8904.c create mode 100644 sound/soc/codecs/wm8904.h create mode 100644 sound/soc/codecs/wm8955.c create mode 100644 sound/soc/codecs/wm8955.h create mode 100644 sound/soc/sh/fsi-da7210.c

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[alsa-devel] ALSA: hda - add possibility to choose speakers configuration for 4930g
by Łukasz Wojniłowicz 25 Jan '10

25 Jan '10

I've got such possibility on MS Windows XP with Realtek drivers (prove is at http://img694.imageshack.us/i/choosechannelmod.png/ ) so I should have possibility to do it also on Linux e.g. here http://img704.imageshack.us/i/screenshotsoundpreferen.png/ Patch given in attachment should do the thing. patch_realtek.c was downloaded from http://git.alsa-project.org/?p=alsa-kernel.git on Thu, 21 Jan 2010 13:58:25 +0000

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Re: [alsa-devel] snd_pcm_hw_params_free() immediately after snd_pcm_hw_params() ???
by Clemens Ladisch 24 Jan '10

24 Jan '10

Raymond Yau wrote: > Is there any flags in substream which can indicate snd_xxx_hw_params() was > called at first time since snd_xxx_hw_params() were called many times > during OSS emulation (probing the supported rates , channels , .... ) ? There is no such flag. It is the responsibility of the driver to track the resources it has allocated. If a resource does not depend on the hardware parameters, it might be a better idea to allocate/free it in the open/close callbacks. Best regards, Clemens

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[alsa-devel] [PATCH] add simplified versions of widget macros
by Guennadi Liakhovetski 23 Jan '10

23 Jan '10

Many macros from include/sound/soc-dapm.h take an array and a number of elements in it as arguments, whereas most users use static arrays and use "x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of those macros, calling ARRAY_SIZE() internally. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c5c95e1..c0922a0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -95,6 +95,21 @@ .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ .num_kcontrols = 1} +/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ +#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\ + wcontrols) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols)\ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols)\ +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = ARRAY_SIZE(wcontrols)} + /* path domain with event - event handler must return 0 for success */ #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ @@ -126,6 +141,23 @@ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} +/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ +#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} +#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} +#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags} + /* events that are pre and post DAPM */ #define SND_SOC_DAPM_PRE(wname, wevent) \ { .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \

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[alsa-devel] [PATCH] ASoC: fix a memory-leak in wm8903
by Guennadi Liakhovetski 23 Jan '10

23 Jan '10

Remember to free the temporary register-cache. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- should go in -rc, maybe also for stable diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce5515e..3595bd5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev) struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; - u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults), + u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults), GFP_KERNEL); /* Bring the codec back up to standby first to minimise pop/clicks */ @@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev) for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) if (tmp_cache[i] != reg_cache[i]) snd_soc_write(codec, i, tmp_cache[i]); + kfree(tmp_cache); } else { dev_err(&i2c->dev, "Failed to allocate temporary cache\n"); }

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[alsa-devel] ca0106 broken (fwd)
by Russell Whitaker 23 Jan '10

23 Jan '10

Hi Sent msg below and no responce. As a work-around replaced ng code with know good code into 2.6.29.6 and it worked. Tried same trick with 2.6.31.12 and it won't compile. If you send me something to try will do it and report back. Russ ---------- Forwarded message ---------- Date: Wed, 21 Oct 2009 14:54:45 -0700 (PDT) From: Russell Whitaker <russ(a)ashlandhome.net> To: James(a)superbug.demon.co.uk Subject: ca0106 broken Hi ca0106 is known to work in these releases: linux-2.6.27.20 linux-2.6.27.23 linux-2.6.28.10 It is broken in: linux-2.6.29.6 linux-2.6.30.9 For the broken releases dmesg reports: unknown hardware "CA0106" "" "" "" "" Anything I can do to help? Russ

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Re: [alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 23 Jan '10

23 Jan '10

Hello Giuliano, No luck, same behavior. I edited the first file listed listed below and recompiled it and installed it. Then I restarted the system with the indigo card in place, and the VU meters show levels in the -120 to -100 dB? I do have alsa-driver-1.0.22 my kernel is 2.6.31 // Salvador ________________________________ From: Giuliano Pochini <pochini(a)shiny.it> To: S. Aguinaga <sa_jr(a)yahoo.com> Sent: Sun, January 17, 2010 4:04:38 PM Subject: Re: [alsa-devel] Echo Indigo card on Sabayon Gentoo On Thu, 14 Jan 2010 12:13:55 -0800 (PST) "S. Aguinaga" <sa_jr(a)yahoo.com> wrote: > Hello Fellows, > Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. First install a recent version of echomixer. If you are using alsa-driver 1.0.22 or kernel 2.6.32, try to add this line: #define ECHOCARD_HAS_LINE_OUT_GAIN in alsa-driver-1.0.22/alsa-kernel/pci/echoaudio/indigoio.c or in linux-2.6.32/sound/pci/echoaudio/indigoio.c and recompile. It re-enables the line-out control that I removed last year because it was not supposed to exist in that card. -- Giuliano.

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[alsa-devel] [PATCH] modem.conf Off-hook improve behavior
by David Fries 23 Jan '10

23 Jan '10

From: David Fries <david(a)fries.net> Only restore the old value if it differs from the requested value, because if it has changed restoring the old value overrides the change. Take for example, a voice modem with a .conf that sets preserve off-hook. Start playback (on-hook to off-hook), start record (off-hook to off-hook), stop playback (off-hook to restore on-hook), stop record (on-hook to restore off-hook), Clearly you don't want to leave the modem "on the phone" now that there isn't any playback or recording active. Signed-off-by: David Fries <david(a)fries.net> --- Comments welcome. The default modem.conf has, hooks.0 { type ctl_elems hook_args [ { name "Off-hook Switch" preserve true value "on" lock false optional true } ] } I wrote the patch to be, if you didn't modify it, don't restore the value. I also thought that it might work to check to see if the current mixer value matches the requested value (which is the value the program set on open), and only set the old value if it hasn't been externally modified. The theory being if something else changed the mixer value, we don't want to by default overwrite that change. Some additional flags could also be added to leave the current behavior alone and add the if you didn't modify it, don't restore it, or if it changed after you set it, don't restore it. Here are some flag names if that would be preferred, preserve_if_different restore_if_unmodified no_squash restore_overrides preserve_if_changed. Setting the lock isn't really an option, because then you can't manually "hang up" the modem if you needed to. On a side note, do any of the alsa voice/software modems support notifying userspace when the line is ringing? I'm using intel8x0m, not that I think it matters here. diff --git a/include/control.h b/include/control.h index 2361dc3..ded884a 100644 --- a/include/control.h +++ b/include/control.h @@ -419,6 +419,7 @@ int snd_ctl_elem_value_malloc(snd_ctl_elem_value_t **ptr); void snd_ctl_elem_value_free(snd_ctl_elem_value_t *obj); void snd_ctl_elem_value_clear(snd_ctl_elem_value_t *obj); void snd_ctl_elem_value_copy(snd_ctl_elem_value_t *dst, const snd_ctl_elem_value_t *src); +int snd_ctl_elem_value_compare(snd_ctl_elem_value_t *left, const snd_ctl_elem_value_t *right); void snd_ctl_elem_value_get_id(const snd_ctl_elem_value_t *obj, snd_ctl_elem_id_t *ptr); unsigned int snd_ctl_elem_value_get_numid(const snd_ctl_elem_value_t *obj); snd_ctl_elem_iface_t snd_ctl_elem_value_get_interface(const snd_ctl_elem_value_t *obj); diff --git a/src/control/control.c b/src/control/control.c index df249dd..159b09a 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2256,6 +2256,18 @@ void snd_ctl_elem_value_copy(snd_ctl_elem_value_t *dst, const snd_ctl_elem_value } /** + * \brief compare one #snd_ctl_elem_value_t to another + * \param dst pointer to destination + * \param src pointer to source + * \return 0 on match, less than or greater than otherwise, see memcmp + */ +int snd_ctl_elem_value_compare(snd_ctl_elem_value_t *left, const snd_ctl_elem_value_t *right) +{ + assert(left && right); + return memcmp(left, right, sizeof(*left)); +} + +/** * \brief Get CTL element identifier of a CTL element id/value * \param obj CTL element id/value * \param ptr Pointer to returned CTL element identifier diff --git a/src/control/setup.c b/src/control/setup.c index 408244e..3606daf 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -192,7 +192,17 @@ int snd_sctl_remove(snd_sctl_t *h) return err; } } - if (elem->preserve) { + /* Only restore the old value if it differs from the requested + * value, because if it has changed restoring the old value + * overrides the change. Take for example, a voice modem with + * a .conf that sets preserve off-hook. Start playback (on-hook + * to off-hook), start record (off-hook to off-hook), stop + * playback (off-hook to restore on-hook), stop record (on-hook + * to restore off-hook), Clearly you don't want to leave the + * modem "on the phone" now that there isn't any playback or + * recording active. + */ + if (elem->preserve && snd_ctl_elem_value_compare(elem->val, elem->old)) { err = snd_ctl_elem_write(h->ctl, elem->old); if (err < 0) { SNDERR("Cannot restore ctl elem");

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[alsa-devel] default alsa pcm improvement
by Aleksandar Kostadinov 23 Jan '10

23 Jan '10

Hello, _From some time now ALSA is a real pleasure to use. Namely it works by OOB and has lots of predefined useful PCMs so one doesn't need to be a guru to do advanced operations. I think that a small improvement in how default PCM is defined can help users with multiple sound cards that like to use pure ALSA and not some of the many available sound servers. I want to be able to switch the default sound device easily in asoundrc but the way the default pcm is defined makes this not very straightforward. I like to use the definitions of default:<CARD> but if I define "pcm.!default" in .asoundrc I lose the "default:<CARD>" definition. I now can rename the "default" definition in default.conf to someting like "defaultorg" and then reference it asoundrc but that requires root as well change can possibly disappear after an alsa-lib package upgrade. Or I can use the dmix and dsnoop predefined PCMs to put together PCMs that I can later reference in my "pcm.!default" definition but in this case one needs to know if the card needs dsnoop/dmix and probably other tweaks that default.conf contains. Additionally this needs to be done for every sound card one owns. So I would suggest the "default" PCM as defined now in the default.conf file to be renamed to something like "alsadefault". Then "default" PCM should be an alias to the alsadefault pcm. This will be backwards compatible to what users have now and owners of multiple sound devices can define their "default" pcm in asoundrc referencing the alsadefault pcm benefiting from the good work already done and not have to mess with a complex asoundrc file or make changes to the system configurations files. Please CC me in replies because I'm not subscribed. Thanks again for the cool work!

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[alsa-devel] [PATCHv2 4/6] OMAP4: MCPDM: Register McPDM platform device
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Add platform device for McPDM interface in OMAP4 and register its memory and irq resources. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/devices.c | 37 +++++++++++++++++++++++++++++++++++++ 1 files changed, 37 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/devices.c b/arch/arm/plat-omap/devices.c index 30b5db7..572265c 100644 --- a/arch/arm/plat-omap/devices.c +++ b/arch/arm/plat-omap/devices.c @@ -28,6 +28,7 @@ #include <plat/menelaus.h> #include <plat/mcbsp.h> #include <plat/dsp_common.h> +#include <plat/omap44xx.h> #if defined(CONFIG_OMAP_DSP) || defined(CONFIG_OMAP_DSP_MODULE) @@ -192,6 +193,41 @@ void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, /*-------------------------------------------------------------------------*/ +#if defined(CONFIG_SND_OMAP_SOC_MCPDM) || \ + defined(CONFIG_SND_OMAP_SOC_MCPDM_MODULE) + +static struct resource mcpdm_resources[] = { + { + .name = "mcpdm_mem", + .start = OMAP44XX_MCPDM_BASE, + .end = OMAP44XX_MCPDM_BASE + SZ_4K, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcpdm_irq", + .start = INT_44XX_MCPDM_IRQ, + .end = INT_44XX_MCPDM_IRQ, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device omap_mcpdm_device = { + .name = "omap-mcpdm", + .id = -1, + .num_resources = ARRAY_SIZE(mcpdm_resources), + .resource = mcpdm_resources, +}; + +static void omap_init_mcpdm(void) +{ + (void) platform_device_register(&omap_mcpdm_device); +} +#else +static inline void omap_init_mcpdm(void) {} +#endif + +/*-------------------------------------------------------------------------*/ + #if defined(CONFIG_MMC_OMAP) || defined(CONFIG_MMC_OMAP_MODULE) || \ defined(CONFIG_MMC_OMAP_HS) || defined(CONFIG_MMC_OMAP_HS_MODULE) @@ -385,6 +421,7 @@ static int __init omap_init_devices(void) omap_init_dsp(); omap_init_kp(); omap_init_rng(); + omap_init_mcpdm(); omap_init_uwire(); omap_init_wdt(); return 0; -- 1.6.0.4

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[alsa-devel] [PATCHv2 3/6] ARM: OMAP4: Add McPDM base address
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Define McPDM physical and L3 base address for OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/omap44xx.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/omap44xx.h b/arch/arm/plat-omap/include/plat/omap44xx.h index 8e4c581..4c64f20 100644 --- a/arch/arm/plat-omap/include/plat/omap44xx.h +++ b/arch/arm/plat-omap/include/plat/omap44xx.h @@ -43,6 +43,8 @@ #define OMAP44XX_SCU_BASE 0x48240000 #define OMAP44XX_LOCAL_TWD_BASE 0x48240600 #define OMAP44XX_WKUPGEN_BASE 0x48281000 +#define OMAP44XX_MCPDM_BASE 0x40132000 +#define OMAP44XX_MCPDM_L3_BASE 0x49032000 #define OMAP44XX_MAILBOX_BASE (L4_44XX_BASE + 0xF4000) -- 1.6.0.4

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[alsa-devel] [PATCHv2 2/6] OMAP4: IRQ: Add McPDM IRQ definition.
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Support for McPDM IRQs in OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/irqs.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/irqs.h b/arch/arm/plat-omap/include/plat/irqs.h index c0ab7c8..bbb8eb6 100644 --- a/arch/arm/plat-omap/include/plat/irqs.h +++ b/arch/arm/plat-omap/include/plat/irqs.h @@ -432,7 +432,7 @@ #define INT_44XX_PARTHASH_IRQ (79 + IRQ_GIC_START) #define INT_44XX_MMC3_IRQ (94 + IRQ_GIC_START) #define INT_44XX_MMC4_IRQ (96 + IRQ_GIC_START) - +#define INT_44XX_MCPDM_IRQ (112 + IRQ_GIC_START) /* Max. 128 level 2 IRQs (OMAP1610), 192 GPIOs (OMAP730/850) and * 16 MPUIO lines */ -- 1.6.0.4

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[alsa-devel] Conexant 5051 subsystem 0x103c30d6
by Jacopo De Simoi 22 Jan '10

22 Jan '10

Dear all, this is a follow up to my previous posting to the ml about having CX 5051 working on my laptop; I found out that the patch provided in http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024284.ht… fits perfectly my model as well. Hence I suggest to add a quirk matching my subsystem id as follows: SND_PCI_QUIRK(0x103c, 0x30d6, "HP DV2700", CXT5051_F700) Thanks __J

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[alsa-devel] Reg: Audio recording with twl4030 codec
by shariff 22 Jan '10

22 Jan '10

Hi all, I am working with OMAP3 based custom board that has twl4030 codec & using linux-2.6.29 kernel from opensource. I have some queries regarding Audio recording. Audio codec is connected to twl4030 as bellow TWL4030 |--------------------| | |------ MAINMIC McBSP2 ------ |i2s/tdm | | ___________| --> One Analog MIC is connected to Main Mic and I am using Audio path not the Voice path. After selecting Main Mic through amixer control. and recording audio using "arecord -c 1 -r 8000 -f S16_LE -t wav -d 20 record-file.wav" --> Here record is successful but with lot of noise. 1) Can we record Mono(1 channel) through I2S from TWL4030 codec? 2) To record stereo(2 channels), do we need two Mic's ? (i.e do I need Sub Mic also). Regards, Shariff

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Re: [alsa-devel] [PATCH 1/4] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 22 Jan '10

22 Jan '10

On Tue, 19 Jan 2010, Mark Brown wrote: > On Tue, Jan 19, 2010 at 09:08:57AM +0100, Guennadi Liakhovetski wrote: > > This all looks pretty good - there's a few issues below but they're > largely stylistic rather than anything fundamental. > > > + 0x0000, 0x0000, 0x0000, 0x01ff, /* 0x08...0x0b */ /* 0x0b contains the VU bit */ > > + 0x01ff, 0x0000, 0x0100, 0x01ff, /* 0x0c...0x0f */ /* 0x0c and 0x0f contain the VU bit */ > > + 0x01ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ /* 0x10 contains the VU bit */ > > What do these comments refer to - do you mean to say that these are not > the actual chip register defaults? The normal way to deal with those is > to have a write (or other update of the cache defaults). This avoids > potential confusion later on if the chip updates the register defaults > or when reviewing against the datasheet. No, I meant, that there's a "Value Update" bit in that register, usually bit 8. So, that bit, of course, is not in default, but it was suggested to me on IRC to set it in cache to force automatic value update. > > +#define ARRAY_SINGLE(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ > > + ARRAY_SIZE(xtexts), xtexts) > > This looks generic - please namespace it and add it to the header file, > other drivers can benefit from it. SOC_ARRAY_SINGLE()? > > +static const struct soc_enum wm8978_enum[] = { > > + /* adc */ > > + ARRAY_SINGLE(WM8978_COMPANDING_CONTROL, 1, wm8978_companding), > > Please define individual variables for this - indexing into the array > gets hard to follow, it's far too easy to have an off by one errors > trying to match things up. > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > > > +/* Mixer #3: Boost (Input) mixer */ > > +static const struct snd_kcontrol_new wm8978_left_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Left PGA Mute", WM8978_LEFT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > +static const struct snd_kcontrol_new wm8978_right_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Right PGA Mute", WM8978_RIGHT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > These should have "Switch" at the end of the name. Also, the names are > going be concatenated with the mixer names so you'll end up with things > "Left Boost Mixer Left PGA Switch" - the individual controls should > probably be just "Switch" so you end up with "Left Boost Mixer Switch". > > > +/* Mic Input boost vol */ > > +static const struct snd_kcontrol_new wm8978_mic_boost_controls[] = { > > +SOC_DAPM_SINGLE("Left Mic Volume", WM8978_LEFT_ADC_BOOST_CONTROL, 4, 7, 0), > > +SOC_DAPM_SINGLE("Right Mic Volume", WM8978_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0), > > +}; > > This doesn't seem to be referenced anywhere? > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > Again, this should be done somewhere generic. Probably by going through > and changing the SND_SOC_DAPM_MIXER definition. Well, having discussed this privately and having looked at the heades, I'm not sure, this is a good idea. First, there are SND_SOC_DAPM_PGA() SND_SOC_DAPM_MIXER() SND_SOC_DAPM_MIXER_NAMED_CTL() SND_SOC_DAPM_PGA_E() SND_SOC_DAPM_MIXER_E() SND_SOC_DAPM_MIXER_NAMED_CTL_E() - 6 macros that follow the pattern .kcontrols = wcontrols, .num_kcontrols = wncontrols, and lots of macros like SOC_ENUM_DOUBLE() SOC_ENUM_SINGLE() SOC_ENUM_SINGLE_EXT() ... doing .max = xmax. BTW, many of the users even open-code array sizes instead of using ARRAY_SIZE() like static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", "Off"}; static const char *spk_function[] = {"On", "Off"}; static const struct soc_enum tosa_enum[] = { SOC_ENUM_SINGLE_EXT(5, jack_function), SOC_ENUM_SINGLE_EXT(2, spk_function), }; Changing only one of them would make the headers and the API look inconsistent. Changing all of them... I really don't think we want to endeavor this now, at least not me;) Besides, doing this locally is very convenient, and is available to everyone - there's no space science involved with this. One can be even as evil as doing +#define _A(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ + m, ARRAY_SIZE(m)) ... +#undef _A +#define _A(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ + ARRAY_SIZE(xtexts), xtexts) ... +#undef _A I think, it's best to leave these things as they are. Otherwise we can give this to kernel janitors as a small exercise;) > > + /* Output PLL to GPIO1 */ > > + snd_soc_write(codec, WM8978_GPIO_CONTROL, > > + snd_soc_read(codec, WM8978_GPIO_CONTROL) | 0x4); > > This should be being done unconditionally. Put a switch in via the > set_dai_clkdiv() call. Ok, I moved it to wm8978_set_dai_clkdiv() under "case WM8978_OPCLKDIV:" > > + /* Mic bias */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + (snd_soc_read(codec, 1) & ~4) | 0x10); > > + > > + /* Out-1 enabled, left/right input channel enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x1bf); > > + > > + /* Out-2 disabled, right/left output channel enabled, dac enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0x10f); > > These should all be being managed via DAPM. Yes, I'm having quite a bit of trouble with DAPM, but I'm getting through slowly. Will need some more time for a next version. > > > +static int wm8978_set_bias_level(struct snd_soc_codec *codec, > > + enum snd_soc_bias_level level) > > +{ > > + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; > > This bitmask maintains everything except the two LSB... > > > + switch (level) { > > + case SND_SOC_BIAS_ON: > > + case SND_SOC_BIAS_PREPARE: > > + power1 |= 1; /* VMID 75k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); > > + break; > > + case SND_SOC_BIAS_STANDBY: > > + power1 |= 0xC; > > ...but this is also managing other bits. Right, bits 7-6 are either kept or set, nothing wrong with that. > > + if (codec->bias_level == SND_SOC_BIAS_OFF) { > > + /* Initial cap charge at VMID 5k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + power1 | 0x3); > > + mdelay(100); > > + } > > > +/* Also supports 12kHz */ > > +#define WM8978_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ > > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | \ > > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) > > SNDRV_PCM_RATE_8000_48000 > > > +static int wm8978_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; > > + > > + /* we only need to suspend if we are a valid card */ > > + if (!codec->card) > > + return 0; > > Don't need to check for the card any more, the core will stop you > getting called without a card. > > > + /* Sync reg_cache with the hardware */ > > + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { > > + if (i == WM8978_RESET) > > + continue; > > You can also skip the write if the register has the default value (this > will speed up resume slightly). > > > + data = cpu_to_be16((i << 9) | (cache[i] & 0x1ff)); > > + codec->hw_write(codec->control_data, (char *)&data, 2); > > + } > > Just use snd_soc_write()? I'll take care about the rest. Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Add PCI quirk for 17" MacBook Pro 5, 2 model (July 2009) to use existing mb5 model
by murray.alex＠gmail.com 22 Jan '10

22 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 7e0ea37..d044db9 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -9223,6 +9223,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x10ec, 0x0885, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- 1.6.3.3

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[alsa-devel] Update information: HD-Audio-models list
by Mark Brown 22 Jan '10

22 Jan '10

Sir/Madam, I have an update for the HD-Audio-Models.txt file. Under ALC262, the toshiba-s06 model setting is applicable to the Sony Vaio VGN-CS60B laptop, and may also be applicable to similar models. It also needs configuring in alsamixer to enable the internal microphone afterwards. Best regards, Mark Brown.

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Re: [alsa-devel] Reforming s3c2443 ac97 driver
by SeungHyun Choi 22 Jan '10

22 Jan '10

3 2

[alsa-devel] [PATCH] ASoC: add helper macros to declare struct soc_enum instances
by Guennadi Liakhovetski 21 Jan '10

21 Jan '10

Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..9537144 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -169,6 +169,23 @@ .private_value = (unsigned long)&xenum } /* + * Simplified versions of above macros, declaring a struct and calculating + * ARRAY_SIZE internally + */ +#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ + struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ + ARRAY_SIZE(xtexts), xtexts) +#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ + SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) +#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \ + struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) +#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ + struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations.

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[alsa-devel] [PATCH 2/5] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 21 Jan '10

21 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 3 + sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 ++++ 4 files changed, 305 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..47e7cce 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,6 +12,9 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..03f3feb --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,272 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + davinci_vcif_dai.dev = &pdev->dev; + davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.private_data = davinci_vcif_dev; + + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + goto fail; + } + + return 0; + +fail: + kfree(davinci_vcif_dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci_vcif", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

1 0

[alsa-devel] Problem with PPH plugin
by Colin Guthrie 21 Jan '10

21 Jan '10

Hi, Just wanted to highlight a bug over on Mozilla relating to a crash in the PPH plugin. https://bugzilla.mozilla.org/show_bug.cgi?id=540460 It was initially though to be in PulseAudio, but on closer inspection this doesn't involve PA at all and in fact seems to relate to the PPH plugin from what I can gather. The backtrace is here: https://bug540460.bugzilla.mozilla.org/attachment.cgi?id=422264 The user did submit a bug report to the bugtracker (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4878) but given previous experience there (see other mail!) I figured I'd post a message here to highlight it seeing as the reporter has already done a lot to try and work out what is going on and posting a message on his behalf seemed like a good idea :) Cheers All the best. -- 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/]

3 9

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 21 Jan '10

21 Jan '10

The following changes since commit 9135f6db09aa9d186f05d41e9c516ec8e3550d78: Mark Brown (1): Merge branch 'mxc-audio' into for-2.6.34 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compile breakage - add a missing header include Mark Brown (3): ASoC: Remove console DAPM debug code ASoC: Support turning off bias when the CODEC is idle ASoC: Use BIAS_OFF when idle for wm_hubs devices Peter Ujfalusi (3): ASoC: tlv320dac33: Correct the prefill number of samples ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/soc-dai.h | 2 + include/sound/soc.h | 2 + include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 24 +++++++-- sound/soc/codecs/wm_hubs.c | 6 ++ sound/soc/soc-dapm.c | 105 ++++++++++---------------------------- 6 files changed, 56 insertions(+), 84 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Support turning off bias when the CODEC is idle
by Mark Brown 21 Jan '10

21 Jan '10

Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 25 ++++++++++++++++++++++++- 2 files changed, 26 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..a8768ea 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -405,6 +405,8 @@ struct snd_soc_codec { short reg_cache_size; short reg_cache_step; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + /* dapm */ u32 pop_time; struct list_head dapm_widgets; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d8e9374..6c33510 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1012,13 +1012,28 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + sys_power = 0; + break; + default: + sys_power = 1; + break; + } break; default: break; } } + if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to turn on bias: %d\n", ret); + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -1042,6 +1057,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply standby bias: %d\n", ret); } + /* If we're in standby and can support bias off then do that */ + if (codec->bias_level == SND_SOC_BIAS_STANDBY && + codec->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + if (ret != 0) + pr_err("Failed to turn off bias: %d\n", ret); + } + /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { ret = snd_soc_dapm_set_bias_level(socdev, -- 1.6.6

1 0

[alsa-devel] Reforming s3c2443 ac97 driver
by JASWINDERSINGH BRAR 21 Jan '10

21 Jan '10

Hi, Right now the S3C2443's AC97_ver2.0 controller driver isn't flexible enough to handle newer SoCs. In order to prepare ground for supporting 64xx and S5Pxxxx the extant driver needs to be reformed, and that is what i am planning to do next. I plan to do the following:- 1) User platform_driver structure to detect and manage AC97 controller platform devices. 2) Remove hardcoded parameters(DMA, IO address, IRQ etc) and pass via platform resources. 3) Initialize platform specific stuff by callback pointers to platform code(cfg_gpio) Objections, suggestions welcome. ------------------------------- Jaswinder Singh Brar Senior Engineer Linux OS Team Application Processor Development System LSI Division SAMSUNG Electronics Co., Ltd. --------------------------------

5 8

[alsa-devel] snd_pcm_hw_params_free() immediately after snd_pcm_hw_params() ???
by Raymond Yau 21 Jan '10

21 Jan '10

Refer to http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s06.html#pcm-… hw_free callback static int snd_xxx_hw_free(struct snd_pcm_substream *substream); This is called to release the resources allocated via *hw_params*. For example, releasing the buffer via snd_pcm_lib_malloc_pages() is done by calling the following: snd_pcm_lib_free_pages(substream); This function is always called before the close callback is called. Also, the callback may be called multiple times, too. Keep track whether the resource was already released. Nearly all the examples in those ALSA tutorials in the web sites have the code if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)); exit (1); } snd_pcm_hw_params_free (hw_params); This seem to violate "This function is always called before the close callback is called." and those hardware mixing sound cards (e.g cs46xx , au88x0, ... ) use snd_xxx_hw_free() to deallocate the hardware mixing path in the DSP. There are a number of audio package (e.g. gstreamer , wine , .... ) call snd_pcm_hw_params_free immediately after snd_pcm_hw_params() Is this a bug of snd_pcm_hw_params_free() not change the state of the PCM stream ? or snd_pcm_prepare() did not check the buffer is already released ?

2 2

[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA850/OMAP-L138 EVM
by Chaithrika U S 20 Jan '10

20 Jan '10

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

3 3

[alsa-devel] device naming
by Andrew Kohlsmith (Mailing List Account) 20 Jan '10

20 Jan '10

Is there a definitive guide to device naming with ALSA? I'm trying to use a single device for simultaneous capture and playback but opening different device names gives different results and I'm trying to determine which is the best to use. aplay -L output gives front, surround40, surround41, surround50, surround51, null and pulse. Trying to register a playback callback using "pulse" gives a "function not implemented" error, although I can open and set all the parameters I need (8kHz, 2 channels, 20ms buffers). I get the same result with "default". If I use "plughw:0,0" I can capture and play audio, but the driver (intel HDA) seems to be very prone to either not calling either the playback or capture callback, or throwing an xrun and not being able to recover. This is an otherwise idle core2 machine running Ubuntu 9.10. Playing with nice or ionice does not seem to affect this "brittleness." I do not have an ~/.asoundrc, nor do I have anything related to asound in /etc. Moving pulseaudio out of the way doesn't seem to do anything either. Is there a definitive list of devices and their device filenames so I can stop guessing? /proc/asound/cards just shows card 0 as an Intel HDA, and /proc/asound/devices shows the following: $ cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0- 1]: hardware dependent 8: [ 0- 0]: hardware dependent 9: [ 0] : control (0-0 is the only device showing both playback and capture, which is why I'm using it, but how do I know that plughw is the correct device filename? I just guessed at it due to previous experience trying to get bt audio working with ALSA.) I'm sure I'm doing something incredibly stupid, but I can't seem to find out what it is. -A.

1 0

[alsa-devel] [PATCH 2/2] Davinci: DM365: Enable DaVinci Voice Codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 20 Jan '10

20 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> It sets up one single card for the DM365 EVM with two subdevices: the AIC3x and the Voice Codec. The general structures are defined at DM365 SoC file and the specific platform data structure for the EVM is defined at board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 39 +++++++++++++++++++++++++++- arch/arm/mach-davinci/dm365.c | 37 ++++++++++++++++++++++++-- arch/arm/mach-davinci/include/mach/asp.h | 1 + arch/arm/mach-davinci/include/mach/dm365.h | 5 +++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 5 files changed, 80 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 8d23972..74f3519 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -175,7 +175,43 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; -static struct snd_platform_data dm365_evm_snd_data; +static int dm365_snd_dma_event_mux(int device, int enable, int playback) +{ + static int tx_dma_mux = 0; + static int rx_dma_mux = 0; + + if (!enable) { + if (playback) + tx_dma_mux = 0; + else + rx_dma_mux = 0; + return 0; + } + + if (playback) { + if (tx_dma_mux == 1) + return -EBUSY; + if (device == 0) + davinci_cfg_reg(DM365_EVT2_ASP_TX); + else + davinci_cfg_reg(DM365_EVT2_VC_TX); + tx_dma_mux = 1; + } else { + if (rx_dma_mux == 1) + return -EBUSY; + if (device == 0) + davinci_cfg_reg(DM365_EVT3_ASP_RX); + else + davinci_cfg_reg(DM365_EVT3_VC_RX); + rx_dma_mux = 1; + } + + return 0; +} + +static struct snd_platform_data dm365_evm_snd_data = { + .dma_event_mux = &dm365_snd_dma_event_mux, +}; static struct i2c_board_info i2c_info[] = { { @@ -513,6 +549,7 @@ static __init void dm365_evm_init(void) evm_init_cpld(); dm365_init_asp(&dm365_evm_snd_data); + dm365_init_vc(&dm365_evm_snd_data); dm365_init_rtc(); #ifdef CONFIG_KEYBOARD_DAVINCI diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index cc3bae4..024cbab 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -455,7 +455,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "timer3", &timer3_clk), CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), - CLK("voice_codec", NULL, &voicecodec_clk), + CLK("davinci_voicecodec", NULL, &voicecodec_clk), CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), @@ -604,6 +604,8 @@ 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 +837,31 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vc_resources[] = { + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + SZ_1K - 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_vc_device = { + .name = "davinci_voicecodec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vc_resources), + .resource = dm365_vc_resources, +}; + static struct resource dm365_rtc_resources[] = { { .start = DM365_RTC_BASE, @@ -985,12 +1012,16 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) davinci_cfg_reg(DM365_MCBSP0_BDR); davinci_cfg_reg(DM365_MCBSP0_R); davinci_cfg_reg(DM365_MCBSP0_BFSR); - davinci_cfg_reg(DM365_EVT2_ASP_TX); - davinci_cfg_reg(DM365_EVT3_ASP_RX); dm365_asp_device.dev.platform_data = pdata; platform_device_register(&dm365_asp_device); } +void __init dm365_init_vc(struct snd_platform_data *pdata) +{ + dm365_vc_device.dev.platform_data = pdata; + platform_device_register(&dm365_vc_device); +} + void __init dm365_init_ks(struct davinci_ks_platform_data *pdata) { davinci_cfg_reg(DM365_KEYSCAN); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..6a175eb 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -50,6 +50,7 @@ #define DAVINCI_ASP1_TX_INT IRQ_MBXINT struct snd_platform_data { + int (*dma_event_mux)(int, int, int); u32 tx_dma_offset; u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 3c07a88..50aeaec 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -31,8 +31,13 @@ #define DM365_RTC_BASE (0x01C69000) +#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_vc(struct snd_platform_data *pdata); void __init dm365_init_ks(struct davinci_ks_platform_data *pdata); void __init dm365_init_rtc(void); diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index b60c693..2937c51 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -325,6 +325,8 @@ 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

4 13

[alsa-devel] [PATCH] Obsolete config in kernel source (USE_MIXARTLOADER)
by Christoph Egger 20 Jan '10

20 Jan '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we're checking referential integrity between kernel KConfig options and in-code Conditional blocks. The config Variable USE_MIXARTLOADER has been non-existent for all git based linux kernels (2.6.12-rc) and seems to be no longer relevant. It is only referenced at one place in the Code (sound/pci/mixart/mixart_hwdep.c). The attached patch makes the behaviour more obvious as it is equivalent to the old one when assuming CONFIG_MIXARTLOADER as not defined. Please keep me informed of this patch getting confirmed / merged so we can keep track of it. Regards Christoph Egger

2 3

[alsa-devel] Conexant CX20561 on HP Pavilion dv2700
by Jacopo De Simoi 20 Jan '10

20 Jan '10

Dear alsa-devs I'm trying to have CX20561 in working order on my laptop; I'll be glad to send a patch once I figure out what is going on there, and for this I need your help. There are a number of issues, mostly regarding the internal mic, as I'm sure you know considering the bugs volume on the subject :) 1a) There are three capture sources listed in alsamixer (Digital excluded): Docking External Internal; the only one that has any influence in recording (both from internal and external mics) is "External" 1b) Why does Node 0x14 shows two volume controls? the first one doesn't seem to have any effect. 2) The input from the internal mic device has very low volume and is very noisy; I tried to optimize fiddling around with hda_verb and the values in alsa-info; the output of codec#0 below shows the optimal ones. (the only relevant ones were gains on node 0x17). I managed, however, to saturate it by blowing directly into it, which shows that it is not really an amplification issue. Rather the real issue here is noise; Audacity shows ~-24dB and in 16bits the noise background gets up to ~+/-500 versus +/-50 with an external mic. The internal mic is indeed a stereo mic; the left channel only produces noise, the right channel seems to work, with the caveats above. (this could be very well that my mic is broken, even if I remember that at some point during my fiddling with hda_verb it started to work properly :) in any case I noticed that the noise background is very much correlated between R and L; would it be possible to write an alsa-plugin to mix the two channels with an inversion to mix away the noise? It sounds fairly simple, but I couldn't find any documentation on how to make a plugin I'm willing to invest some time on fixing this, so maybe I can get some help here to start with. Judging from the patch_conexant code it should be at least possible to fix #1. I'd like to hear your thoughts about #2; especially about fiddling around with hda_verb or making the described alsa-plugin. Thanks a lot Jacopo De Simoi !!################################ !!ALSA Information Script v 0.4.58 !!################################ !!Script ran on: Wed Jan 20 08:49:43 UTC 2010 !!Linux Distribution !!------------------ Gentoo Base System release 2.0.0 !!DMI Information !!--------------- Manufacturer: Hewlett-Packard Product Name: HP Pavilion dv2700 Notebook PC !!Kernel Information !!------------------ Kernel release: 2.6.30-tuxonice-r9 Operating System: GNU/Linux Architecture: x86_64 Processor: AMD Turion(tm) 64 X2 TL-62 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.22.1 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfc480000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:07.0 0403: 10de:055c (rev a1) Subsystem: 103c:30d6 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=laptop snd-hda-intel: model=auto !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : laptop,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Conexant CX20561 (Hermosa) Address: 0 Function Id: 0x2 Vendor Id: 0x14f15051 Subsystem Id: 0x103c30d6 Revision Id: 0x100000 Modem Function Group: 0x2 Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x3b 0x3b] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x4a 0x4a] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Internal Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="External Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="External Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x00 0x00] [0x50 0x50] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x1d 0x17* Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Docking Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Docking Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01214030: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=37, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00001224: IN Detect Vref caps: 50 80 Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=38, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001224: IN Detect Vref caps: 50 80 Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=39, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x1a [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1b [Pin Complex] wcaps 0x400500: Mono Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="Conexant Digital", type="SPDIF", device=1 Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x1d [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00000020: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x1e [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jan 19 20:26 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jan 19 20:26 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 24 Jan 20 09:31 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 20 09:34 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 17 Jan 19 20:26 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 1 Jan 19 20:26 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jan 19 20:26 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: Conexant Digital [Conexant Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfc480000 irq 20' Mixer name : 'Conexant CX20561 (Hermosa)' Components : 'HDA:14f15051,103c30d6,00100000' Controls : 15 Simple ctrls : 8 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 59 [80%] [-15.00dB] [off] Front Right: Playback 59 [80%] [-15.00dB] [off] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Docking Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 0 [0%] [-74.00dB] Playback [on] Front Right: 0 [0%] [-74.00dB] Playback [on] Simple mixer control 'External Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 80 [100%] [6.00dB] Playback [on] Front Right: 80 [100%] [6.00dB] Playback [on] Simple mixer control 'Internal Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 0 [0%] [-74.00dB] Playback [on] Front Right: 0 [0%] [-74.00dB] Playback [on] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'Internal Mic Volume' value.0 0 value.1 0 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Internal Mic Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'External Mic Volume' value.0 80 value.1 80 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'External Mic Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'Docking Mic Volume' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Docking Mic Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 74' comment.dbmin -7400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 59 value.1 59 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value false } control.9 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.10 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.11 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.14 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } control.15 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module ndiswrapper bnep sco rfcomm l2cap snd_seq snd_seq_device snd_hda_codec_conexant uvcvideo videodev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_timer sdhci_pci sdhci mmc_core v4l1_compat nvidia forcedeth snd btusb bluetooth v4l2_compat_ioctl32 led_class snd_page_alloc !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x16 0x01214030 0x17 0x01a19020 0x18 0x400001f0 0x19 0x400001f0 0x1a 0x92170110 0x1b 0x400001f0 0x1c 0x400001f0 0x1d 0x400001f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ [ 8.457898] ACPI: PCI Interrupt Link [LAZA] enabled at IRQ 20 [ 8.457916] HDA Intel 0000:00:07.0: PCI INT A -> Link[LAZA] -> GSI 20 (level, low) -> IRQ 20 [ 8.457970] HDA Intel 0000:00:07.0: setting latency timer to 64 [ 8.499562] Linux video capture interface: v2.00 -- [ 8.548267] USB Video Class driver (v0.1.0) [ 9.660207] input: HDA NVidia Mic as /class/input/input10 [ 9.660575] input: HDA NVidia Mic as /class/input/input11 [ 9.660798] input: HDA NVidia Headphone as /class/input/input12 [ 10.842906] kjournald starting. Commit interval 5 seconds -- [18156.018398] forcedeth 0000:00:0a.0: PCI INT A disabled [18156.400042] HDA Intel 0000:00:07.0: PCI INT A disabled [18156.440030] ehci_hcd 0000:00:04.1: PCI INT B disabled -- [18157.038988] ehci_hcd 0000:00:04.1: restoring config space at offset 0x1 (was 0xb00006, writing 0xb00002) [18157.039014] HDA Intel 0000:00:07.0: restoring config space at offset 0xf (was 0x5020100, writing 0x502010a) [18157.039022] HDA Intel 0000:00:07.0: restoring config space at offset 0x4 (was 0x0, writing 0xfc480000) [18157.039027] HDA Intel 0000:00:07.0: restoring config space at offset 0x1 (was 0xb00000, writing 0xb00002) [18157.039057] ahci 0000:00:09.0: restoring config space at offset 0x1 (was 0xb00007, writing 0xb00407) -- [18157.150089] amd74xx 0000:00:06.0: BIOS didn't set cable bits correctly. Enabling workaround. [18157.150111] HDA Intel 0000:00:07.0: PCI INT A -> Link[LAZA] -> GSI 20 (level, low) -> IRQ 20 [18157.150115] HDA Intel 0000:00:07.0: setting latency timer to 64 [18157.850034] pci 0000:00:08.0: setting latency timer to 64

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[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

The same is true for USE_PCXHRLOADER and USE_VXLOADER Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- include/sound/vx_core.h | 2 +- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 5456343..c6835dd 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -28,7 +28,7 @@ #include <linux/interrupt.h> #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_VXLOADER) && !defined(CONFIG_SND_VX_LIB) /* built-in kernel */ +#if !defined(CONFIG_SND_VX_LIB) /* built-in kernel */ #define SND_VX_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 17cb123..16152ce 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -35,7 +35,7 @@ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */ +#if !defined(CONFIG_SND_PCXHR) /* built-in kernel */ #define SND_PCXHR_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --bCsyhTFzCvuiizWE--

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[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

The same is true for USE_PCXHRLOADER Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 17cb123..16152ce 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -35,7 +35,7 @@ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */ +#if !defined(CONFIG_SND_PCXHR) /* built-in kernel */ #define SND_PCXHR_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --liOOAslEiF7prFVr--

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[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- sound/pci/mixart/mixart_hwdep.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --+pHx0qQiF2pBVqBT--

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Re: [alsa-devel] [PATCH] ASoC: DaVinci: Fix stream restart error
by Chaithrika U S 20 Jan '10

20 Jan '10

All, Please ignore this patch, missed the 'To' address in the mail. I have re-submitted this patch. Regards, Chaithrika On Wed, Jan 20, 2010 at 17:04:45, alsa-devel(a)alsa-project.org wrote: > From: Chaithrika U S <chaithrika(a)ti.com> > > Sometimes after a suspend-resume cycle, the ALSA application > restarts the stream when resume fails and McASP fails to work > as the clock is not enabled. This patch corrects this bug. > > Testes on TI DA850/OMAP-L138 EVM. > > 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 | 5 ++--- > 1 files changed, 2 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c > index a613bbb..ab6518d 100644 > --- a/sound/soc/davinci/davinci-mcasp.c > +++ b/sound/soc/davinci/davinci-mcasp.c > @@ -768,13 +768,12 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, > > switch (cmd) { > case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > if (!dev->clk_active) { > clk_enable(dev->clk); > dev->clk_active = 1; > } > - /* Fall through */ > - case SNDRV_PCM_TRIGGER_START: > - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > davinci_mcasp_start(dev, substream->stream); > break; > > -- > 1.5.6 >

1 0

[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.

2 2

[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

2 1

[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

1 1

[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

2 1

[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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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by miguel.aguilar＠ridgerun.com 18 Jan '10

18 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> There are two clients for the MFD driver: * Voice Codec Interface. * Voice Codec itself, CQ0093. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 5 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 216 ++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 113 +++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 342 ++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/davinci/Kconfig | 10 +- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.h | 28 +++ 11 files changed, 750 insertions(+), 2 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..efeba26 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,11 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + bool "DaVinci Voice Codec" + help + This driver supports the DaVinci voice codec and its interface. + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index ca2f2c4..f5c617f 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_MFD_SH_MOBILE_SDHI) += sh_mobile_sdhi.o obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 0000000..94dbeaa --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,216 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/core.h> +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int davinci_vc_client_dev_register(struct davinci_vc *davinci_vc, + const char *name, + struct platform_device **pdev) +{ + int ret; + + *pdev = platform_device_alloc(name, -1); + if (pdev == NULL) { + dev_err(davinci_vc->dev, "failed to allocate %s\n", name); + return -ENODEV; + } + + (*pdev)->dev.parent = davinci_vc->dev; + platform_set_drvdata(*pdev, davinci_vc); + ret = platform_device_add(*pdev); + if (ret != 0) { + dev_err(davinci_vc->dev, "failed to register %s: %d\n", name, + ret); + platform_device_put(*pdev); + *pdev = NULL; + return ret; + } + + return 0; +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + + /* Register voice codec interface */ + ret = davinci_vc_client_dev_register(davinci_vc, "davinci_vcif", + &(davinci_vc->davinci_vcif.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail4; + } + + /* Register voice codec */ + ret = davinci_vc_client_dev_register(davinci_vc, "cq93vc", + &(davinci_vc->cq93vc.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail5; + } + + return 0; + +fail5: + platform_device_unregister(davinci_vc->davinci_vcif.pdev); +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + platform_device_unregister(davinci_vc->cq93vc.pdev); + platform_device_unregister(davinci_vc->davinci_vcif.pdev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 0000000..6a1757a --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,113 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..a759d06 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 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 dbaecb1..674db53 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.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 @@ -64,6 +65,7 @@ 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_AK4671) += snd-soc-ak4671.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..f420e01 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,342 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#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 <linux/mfd/davinci_voicecodec.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" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_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, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_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 davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->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: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + 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 = { + .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_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, DAVINCI_VC_REG05, DAVINCI_VC_REG05_PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_DIG_ATTEN); + + /* 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); + return ret; + } + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +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; + + /* 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); + + 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_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + 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; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int cq93vc_codec_suspend(struct platform_device *pdev, pm_message_t m) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int cq93vc_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define cq93vc_codec_suspend NULL +#define cq93vc_codec_resume NULL +#endif + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), + .suspend = cq93vc_codec_suspend, + .resume = cq93vc_codec_resume, +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +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..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..3aa04e6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,15 +12,21 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

3 3

[alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 18 Jan '10

18 Jan '10

Hello Fellows, Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. Do you know what I might have to do to get it working on Gentoo (sabayon)? When I type the following, this is what I get: ~/Downloads/alsa-utils-1.0.22/alsactl $ ./alsactl init 2 Unknown hardware: "Echo_Indigo IO" "" "" "0xecc0" "0x00a0" Hardware is initialized using a guess method Using aplay to list the device: card 2: IO [Indigo IO], device 0: PCM [Indigo IO] Subdevices: 8/8 .... etc. Using aplay to test the card here is the command: aplay -D plughw:2,0 <wav file> And this does not work, I have checked that things are ok with alsamixer, but still! Any help will be appreciated or let me know if this is best addressed by submitting a bug. // Salvador

2 2

[alsa-devel] WARNING: at sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]()
by John Kacur 18 Jan '10

18 Jan '10

The following was in my 2.6.33-rc3 dmesg (of course that doesn't mean that the problem didn't exist earlier too) WARNING: at /home/jkacur/rt.linux.git/sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]() Hardware name: 2241B48 BUG? (minor < 0 || minor >= 128) Modules linked in: snd_mixer_oss thinkpad_acpi(+) sdhci hwmon i2c_i801 firewire_ohci snd_pcm mmc_core snd_timer cfg80211 firewire_core btusb joydev iTCO_wdt bluetooth ata_generic e1000e i2c_core ppdev snd battery ac iTCO_vendor_support parport_pc ricoh_mmc intel_agp pata_acpi pcspkr parport tpm_tis video wmi output soundcore button tpm rfkill sr_mod tpm_bios snd_page_alloc crc_itu_t cdrom sg ahci libata sd_mod scsi_mod crc_t10dif xfs exportfs uhci_hcd ohci_hcd ehci_hcd [last unloaded: scsi_wait_scan] Pid: 1317, comm: modprobe Not tainted 2.6.33-rc3 #1 Call Trace: [<ffffffffa027bb25>] ? snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffff8104338c>] warn_slowpath_common+0x7c/0xa9 [<ffffffff81043410>] warn_slowpath_fmt+0x41/0x43 [<ffffffffa027bb25>] snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffffa027bc7e>] snd_register_oss_device+0x2c/0x21a [snd] [<ffffffffa0271e1e>] snd_mixer_oss_notify_handler+0xa5/0x2e5 [snd_mixer_oss] [<ffffffff81375e00>] ? __mutex_unlock_slowpath+0x5c/0x141 [<ffffffff810710d5>] ? trace_hardirqs_on_caller+0x11f/0x14a [<ffffffff8107110d>] ? trace_hardirqs_on+0xd/0xf sdhci-pci 0000:15:00.2: SDHCI controller found [1180:0822] (rev 21) sdhci-pci 0000:15:00.2: PCI INT C -> GSI 18 (level, low) -> IRQ 18 [<ffffffff81375ed3>] ? __mutex_unlock_slowpath+0x12f/0x141 sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. Registered led device: mmc0:: mmc0: SDHCI controller on PCI [0000:15:00.2] using DMA [<ffffffffa027628a>] snd_card_register+0x135/0x14b [snd] [<ffffffffa0370c88>] volume_init+0x335/0x3d2 [thinkpad_acpi] [<ffffffffa037289b>] thinkpad_acpi_module_init+0x61f/0xa4f [thinkpad_acpi] [<ffffffffa037227c>] ? thinkpad_acpi_module_init+0x0/0xa4f [thinkpad_acpi] [<ffffffff810001fa>] do_one_initcall+0x5f/0x154 [<ffffffff8107ff34>] sys_init_module+0xd7/0x234 [<ffffffff81002d5b>] system_call_fastpath+0x16/0x1b ---[ end trace a6d53dc318f09cb5 ]---

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[alsa-devel] issue with sound stopping and not being able to restart
by Andrew Kohlsmith (Mailing List Account) 18 Jan '10

18 Jan '10

Good morning, I am new to ALSA development, but experienced in C. I wrote up a little test application which simply opens one playback and one capture device and spits noise into the playback and recording it on the capture device. I cross- correlate the data streams to determine the "depth" or latency of the pipe, end-to-end. I'm using playback and capture callbacks to handle the audio. The program seems to work reasonably well, but at seemingly random times either the playback or capture stream will "hiccup" (xrun) -- I am handling this case, but it's fairly often that the stream does NOT resume and I'm stuck with either no playback or no capture. I've looked at the example programs and the documentation and I do not see what it is I'm doing wrong. I'm posting my code here, hopefully someone can show me the error of my way. Capture/playback setup (error checking removed): snd_pcm_open(handle, fname, stream, SND_PCM_NONBLOCK); snd_pcm_hw_params_malloc(hwptr); snd_pcm_hw_params_any(h, hwp); snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(h, hwp, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_rate(h, hwp, 8000, 0); snd_pcm_hw_params_set_channels(h, hwp, 2); snd_pcm_hw_params_set_buffer_size(h, hwp, BUF_SIZE); snd_pcm_hw_params(h, hwp); At this point I have the capture or playback device set up for 16 bit signed linear data in interleaved format and two channels. I've set up the buffers to be 160 frames, or 20ms. Each of these calls is checked for errors, and I do verify that the rate, buffer size and buffer time are what I expect. There is nothing unusual going on with these calls that I can determine. I set up the playback and capture callbacks thusly (again checking for errors) snd_async_add_pcm_handler(&ahp, hp, playback_callback, &pb_data); snd_async_add_pcm_handler(&ahc, hc, capture_callback, &cap_data); At this point I believe I'm ready to go, all I have to do is start playback and capture: if((fcount = snd_pcm_writei(h[PLAYBACK], pb_data.buf, BUF_SIZE)) != BUF_SIZE) { if(fcount < 0) { if((e = snd_pcm_recover(h[PLAYBACK], fcount, 0)) < 0) fprintf(stderr, "initial: can't writei: %s\n", snd_strerror(e)); } else { fprintf(stderr, "initial: short write: %d instead of %d\n", fcount, BUF_SIZE); } } if((e = snd_pcm_start(h[CAPTURE])) < 0) { fprintf(stderr, "capture prepare error: %s\n", snd_strerror(e)); return 1; } Now the playback and capture handlers are called by ALSA as required. The capture handler looks like this: /* capture callback. Called by ALSA whenever a period time is up */ void capture_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; short *p; int i, fc, e, faloops; struct timeval tv; static int c=0; /* read in whatever is available */ fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; if((fc = snd_pcm_readi(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't readi: %s\n", snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short read: %d instead of %ld\n", fc, fa); } } for(i=0,p=&d->buf[0]; i<fc; i++) { *d->ptr++ = *p++; p++; if(d->ptr > &in[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&in[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; in_step++; if(in_step >= MAX_BUFS) in_step = 0; } frames_cap += fc; fa = snd_pcm_avail_update(h); faloops++; }; cap_faloops=faloops; } The playback callback is very similar: /* playback callback. Called by ALSA whenever a period time is up */ void playback_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; int i, fc, e, faloops; static int c=0; struct timeval tv; fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; for(i=0; i<fa; i++) { d->buf[i*2] = awgn(&noise_source); d->buf[i*2+1] = awgn(&noise_source); } if((fc = snd_pcm_writei(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't writei: error %x %s\n", e, snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short write: %d instead of %ld\n", fc, fa); } } for(i=0; i<fc; i++) { *d->ptr++ = d->buf[i*2]; if(d->ptr > &out[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&out[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; out_step++; if(out_step >= MAX_BUFS) out_step = 0; } frames_pb += fc; fa = snd_pcm_avail_update(h); faloops++; }; pb_faloops = faloops; } That's really it; the main loop keeps track of the in_step and out_step variables and does the cross-correlation as the data comes in. Now it seems that it's the capture playback that is getting xruns, but I when capture xruns occur my playback callback seems to stop, and sometimes the entire program just hangs as if it is waiting for ALSA to call one of the callbacks. Can anyone shed any light on this? I'm certain it's something very simple and braindead, but I can't find it. Regards, Andrew

1 0

Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 18 Jan '10

18 Jan '10

> Actually i am working on soc platform(android-S3C6410) and wanted to > use DMIX and some good resampler for my software mixing instead of > current software mixing and resampling in upper layer(Above > alsa-lib).DMIX is working fine but this resampling is causing some > jitter.During rate conversion it looks for > libasound_module_rate_speexrate.so which i don't have so its using > default one causing the jitter. >>Have you considered using PulseAudio for this? Mark thanks for the suggestion and i checked out PulseAudio for my requirements.I found out that i don't need all the things in PulseAudio a. I just need good resampling code.Can i just use just the resampler code of pulseaudio(mine is mobile enviroment)? b. Just want a good mixer at ALSA-LIB level.As of now this is provided by android framework but i want this to be done by ALSA-LIB so that i can replace with some good mixer later on(if i find that some other mixer is good).DMIX is providing that but it's not that good. c.Jaroslav suggested to use sox resamplers but for that to work i have to get the sox resampler code(proper license) and need to build it for my system and link it with corresponding functions of ALSA-LIB so that they can call relevant functions for resampling.As i am working on mobile domain i think i need to do the integration of sox resampler code with ALSA-LIB for that to work?? or Can i just use the integrated sox library(some .so i guess) and change the asound.conf for using that library??Will this work?? d. I saw some check-ins related to Scenario Manager.As of now scenario manager functionality is also given in Android(we open a differnet device and the corresponding hooks for changing the routes dynamically) but i think if it given in alsa-lib switching will be much faster and this will increase the system performance.

2 1