September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 3.2
by Mark Brown 27 Sep '11

27 Sep '11

The following changes since commit 0a742681e6072a71f30cfe6312f758f1cd185c21: ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path (2011-09-23 11:54:24 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (4): ASoC: Drop exporting sn95031_get_mic_bias ASoC: Drop exporting ad1980_dai ASoC: Remove unused fields in struct mfld_mc_private ASoC: Staticise sst_platform_dai Lars-Peter Clausen (3): ASoC: ssm2602: Re-enable oscillator after suspend ASoC: ssm2602: Set initial bias level to standby ASoC: ssm2602: Support setting the oscillator and the clock output state Mark Brown (9): ASoC: Support a wider range of sample rates on Speyside WM8962 ASoC: Rename WM8962 DMIC widget to DMIC_ENA ASoC: Convert WM8962 MICBIAS to a supply widget ASoC: Add support for on-board analogue microphones on Speyside WM8962 ASoC: Add DMIC control to Speyside WM8962 board ASoC: Don't force bias on ground referenced devices ASoC: Add platform data for WM1250 EV1 GPIOs ASoC: Set idle_bias_off for WM1250 EV1 Merge branch 'for-3.1' into for-3.2 Peter Ujfalusi (11): ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’ ASoC: twl6040: Read the TRIM values from the chip ASoC: twl6040: Function to fetch the TRIM values ASoC: omap-mcpdm: API to configure offset cancellation ASoC: sdp4430: Configure McPDM offset cancellation ASoC: twl6040: Rename pga_event to out_drv_event ASoC: twl6040: Combine the custom volsw get, and put functions ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data ASoC: twl6040: No need to change delay during HS ramp ASoC: twl6040: No need to change delay during HF ramp include/sound/wm1250-ev1.h | 27 ++++++ sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/sn95031.c | 1 - sound/soc/codecs/ssm2602.c | 73 +++++++++++++---- sound/soc/codecs/ssm2602.h | 6 +- sound/soc/codecs/twl6040.c | 155 ++++++++++++----------------------- sound/soc/codecs/twl6040.h | 13 +++ sound/soc/codecs/wm1250-ev1.c | 122 +++++++++++++++++++++++++++- sound/soc/codecs/wm8962.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 2 - sound/soc/mid-x86/sst_platform.c | 2 +- sound/soc/omap/omap-mcbsp.c | 3 +- sound/soc/omap/omap-mcpdm.c | 25 ++++++ sound/soc/omap/omap-mcpdm.h | 12 +++ sound/soc/omap/sdp4430.c | 10 ++- sound/soc/samsung/speyside_wm8962.c | 24 ++++-- sound/soc/soc-dapm.c | 7 +- 17 files changed, 350 insertions(+), 143 deletions(-) create mode 100644 include/sound/wm1250-ev1.h

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[alsa-devel] [PATCH 00/10] ASoC: twl6040: Gain ramp code cleanups
by Peter Ujfalusi 27 Sep '11

27 Sep '11

Hello, the following series cleans up the gain ramp code found in the twl6040 codec driver. Main changes: - use one workqueue for the twl6040 codec driver (instead of the original 3) - Delays between the steps does not need to be different among the range. I assume, that the original code contained copy-paste snippets from wm8350 for this part - Cleanups for the DAPM_OUT_DRV_E event handler code. - delayed_works moved within their corresponding struct. The series has been generated on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch + ASoC: omap-mcpdm/twl6040: Offset cancellation series. Side note: I have done this to better understand (while cleaning up the twl6040 driver) the requirements for the ramp code, and to study the possibility of adding support in the core for this (to handle the wm8350, and twl6040 in a generic way later). I'm still looking at the optimal implementation without ending up with too complicated code/structures... Regards, Peter --- Peter Ujfalusi (10): ASoC: twl6040: Rename pga_event to out_drv_event ASoC: twl6040: Combine the custom volsw get, and put functions ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data ASoC: twl6040: One workqueue should be enough ASoC: twl6040: correct loop counters for HS/HF ramp code ASoC: twl6040: No need to change delay during HS ramp ASoC: twl6040: No need to change delay during HF ramp ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event ASoC: twl6040: Simplify code in out_drv_event for pending work check sound/soc/codecs/twl6040.c | 203 ++++++++++++------------------------------- 1 files changed, 57 insertions(+), 146 deletions(-) -- 1.7.6.1

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[alsa-devel] [PATCH 1/3] ASoC: ssm2602: Re-enable oscillator after suspend
by Lars-Peter Clausen 27 Sep '11

27 Sep '11

Currently the the internal oscillator is powered down when entering BIAS_OFF state, but not re-enabled when going back to BIAS_STANDBY. As a result the CODEC will stop working after suspend if the internal oscillator is used to generate the sysclock signal. This patch fixes it by clearing the appropriate bit in the power down register when the CODEC is re-enabled. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- sound/soc/codecs/ssm2602.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cceb002..32d6c51 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -430,7 +430,8 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, static int ssm2602_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, SSM2602_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, SSM2602_PWR); + reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN); switch (level) { case SND_SOC_BIAS_ON: -- 1.7.2.5

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 27 Sep '11

27 Sep '11

The following changes since commit f0e8ed858edb327802ee65fd695cc1538286226f: ASoC: Ensure we generate a driver name (2011-09-21 14:54:23 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.1 Jarkko Nikula (1): ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active Lars-Peter Clausen (1): ASoC: ssm2602: Re-enable oscillator after suspend sound/soc/codecs/ssm2602.c | 3 ++- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 2 files changed, 8 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: Remove unused fields in struct mfld_mc_private
by Axel Lin 27 Sep '11

27 Sep '11

Both *socdev and *codec of struct mfld_mc_private are not being used in this driver, remove it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 429aa1b..1d818dc 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -54,9 +54,7 @@ static unsigned int hs_switch; static unsigned int lo_dac; struct mfld_mc_private { - struct platform_device *socdev; void __iomem *int_base; - struct snd_soc_codec *codec; u8 interrupt_status; }; -- 1.7.4.1

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[alsa-devel] [RFC][PATCH] ASoC: Make wm1250-ev1 depends on GPIOLIB
by Axel Lin 27 Sep '11

27 Sep '11

I got below build error if GPIO support is disabled. Make wm1250-ev1 depends on GPIOLIB to fix below build error. CC sound/soc/codecs/wm1250-ev1.o sound/soc/codecs/wm1250-ev1.c:32:14: error: array type has incomplete element type sound/soc/codecs/wm1250-ev1.c: In function ‘wm1250_ev1_pdata’: sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:126:18: error: negative width in bit-field ‘<anonymous>’ sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:134:42: error: negative width in bit-field ‘<anonymous>’ sound/soc/codecs/wm1250-ev1.c: In function ‘wm1250_ev1_free’: sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:155:34: error: negative width in bit-field ‘<anonymous>’ make[3]: *** [sound/soc/codecs/wm1250-ev1.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I got the build error while trying to build for SND_MFLD_MACHINE and SND_SST_PLATFORM. ( "Enable ALSA for SoC audio support" and "Build all ASoC CODEC drivers" on X86 ) Axel sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3449431..0ad80fe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -58,7 +58,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE - select SND_SOC_WM1250_EV1 if I2C + select SND_SOC_WM1250_EV1 if I2C && GPIOLIB select SND_SOC_WM2000 if I2C select SND_SOC_WM5100 if I2C select SND_SOC_WM8350 if MFD_WM8350 -- 1.7.4.1

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[alsa-devel] [patch] ALSA: usb-audio - NULL dereference on probe error
by Dan Carpenter 27 Sep '11

27 Sep '11

There was a potential NULL dereference introduced in 362e4e49abe "ALSA: usb-audio - clear chip->probing on error exit" Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/card.c b/sound/usb/card.c index cc08588..8db260f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -532,7 +532,8 @@ snd_usb_audio_probe(struct usb_device *dev, __error: if (chip && !chip->num_interfaces) snd_card_free(chip->card); - chip->probing = 0; + if (chip) + chip->probing = 0; mutex_unlock(&register_mutex); __err_val: return NULL;

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[alsa-devel] [PATCH] Set CLOEXEC flag for pipes in PulseAudio plugins
by Chris Rankin 27 Sep '11

27 Sep '11

Hi, I have noticed that Alsa / PulseAudio are leaking a couple of pipes after a fork/exec, and have traced the leak to the following two plugins: libasound_module_ctl_pulse.so libasound_module_pcm_pulse.so This is with Fedora 15. I have attached a patch for the problem. Cheers, Chris

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Re: [alsa-devel] [PATCH] sound: usb: mixer: increase control transfer timeout
by Takashi Iwai 27 Sep '11

27 Sep '11

At Mon, 29 Aug 2011 14:10:26 +0300, Felipe Balbi wrote: > > 100ms might not be enough for slower devices. Increasing > timeout to 1000ms doesn't seem like overkill. > > Signed-off-by: Felipe Balbi <balbi(a)ti.com> Felipe, what is the status of this patch? Could you resend a revised patch? thanks, Takashi

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Re: [alsa-devel] snd_pcm_pause kills plaback handle on OMAP4
by john＠johnandjuliet.com 27 Sep '11

27 Sep '11

Thanks for the response. Yea I guess my question is a bit thin and the setup is a bit more complex so I should list the particulars: * This is on an OMAP4 BLAZE ES2.2 or 2.3 processor * Running Android Linux 2.6.35 from the TI release version G.1 * I am actually configuring ALSA to utilize the MODEM audio path (which is hw:0,5 that routes McBSP2 audio to/from the 6040's speaker and mic.) * The McBSP is configured for slave mode network SSI with the audio flow on slots 0 and 1 (the default was I2S) * McBSP is configured for 8K sps two channel 32-bit samples (though it could be 16 bit samples too) I guess the way I wrote the first email that I made it sound like it is the McBSP that is not working however it could be that while I am sending in audio perhaps the McBSP2 is working (not sure where I could look at data comming in down in the kernel <help plz> but if the McBSP2 is working then it could be that the data is not getting to the 6040?! > > On Wednesday 14 September 2011 11:13:22 john(a)johnandjuliet.com wrote: > > I'm trying to control the McBSP2 that is used for the modem in an > > android environment (BLAZE OMAP4) > > What kernel are you using? 2.6.35 released by TI for Android Gingerbread. TI tags this as their G1 release (they are up to G3 now) > > > What I am trying to do is control the audio paths for capture and > > playback seperatly like in a SIMPLEX (half duplex) instead of full > > DUPLEX mode audio .. > > So either playback, or capture, but never both at the same time? Yes. The device cannot do full duplex. The ALSA config for hw:0,5 is to route audio from McBSP2 to/from the 6040 (ABE) The stock Android alsa lib package opens up handles for both capture and playback and then turns on audio hardware both MICs and Speakers. > > > I also want to stop the audio without total > > shutdown of the audio system. > > > > From what I can see that would be to call snd_pcm_pause(handle,bool) > > > > But here is what happens. With the handle forthe plaback path > > everything works OK... But if I call that then the capture path stops > > working... even if I try to turn on (unpause) both paths. > > So you have both playback and capture opened, if you pause the playback, and > release it, then the capture path stops working? > Or you mean that doing the pause/release with playback, then closing it, after > that the capture is not working? If they are both flowing and I pause capture then playback stops and though I can un-pause capture playback never comes back. I'll have to double check but I think if I pause playback and then un-pause it... that too breaks and playback never resumes. > > > This implies some sort of bug in the omap soc driver? > > Everything is possible. I have not heard such about such a problem myself. > If this issue exist on OMAP4, it should be present on OMAP3 as well... OK. perhaps though as I just noted I left out that this could actually be a problem on the speaker flow out and not the McBSP2 flow in.. What do you think? The OMAP4 audio system is much different than OMAP3 on the speaker/mic side so I guess it could be a problem there? > > Could you be more specific on the scenario? > What codec you have attached to McBSP2 port? > Are you using ABE, or are you using the McBSP2 port directly? > I have my own simplex modem connected and it is generating 8ksps audio on slot 0 (first 32 bits) and the McBSP is configured that way. I am using the ABE to turn on the speaker hardware and the 6040 side of things. Mostly I am using the TI ALSA layer they provide for Android ALSA audio control and simply placed into it at a key point a snd_pcm_pause() for when I want to turn the capture off or on and another place for the playback to be off or on. Typically capture and playback are not both on at the same time.. You are either speaking or you are listening. This is like a CB radio or a Walkie Talkie type of situation. To save power and also reduce static the speaker is only turned on when the modem on McBSP2 indicates that it wants to play audio and that is when I would do a snd_pcm_pause(playbackhandle,0); when the audio from the modem stops and I get the signal from the modem then I call snd_pcm_pause(playbackhandle,1); When I want to talk then I do the snd_pcm_pause(capture_handle,0); and when I stop talking then I do the pause on the capture_handle. I will have to check but I think if I pause EITHER one then playback stops working. Playback does work if all I do is an open and not mess with it and just do the ALSA set controls to turn on the speaker. > CC-ing Jarkko to catch his attention... > > -- > Péter JBP

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