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

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[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/]

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[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(-)

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[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

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[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. --------------------------------

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