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[alsa-devel] A few fix patches for ASoC
by Takashi Iwai 01 Oct '09

01 Oct '09

Mark, I applied a few fix patches to fix/asoc and topic/asoc branches for build errors I found with cross-build tests in this morning. Check below and sync to your tree. commit bb26276744a80d066681836f4d49c70010b129d6 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 07:39:45 2009 +0200 ASoC: Fix build errors of wm8711.c with SPI commit c877c25170e2655d519b29e91d6c91d5d1a72a6f Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:33:47 2009 +0200 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 commit 140318aaa924ce9664ff59366993228cf1547f1d Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:40:32 2009 +0200 ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c

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[alsa-devel] LEFT or RIGHT Channel in alsa mixer
by Navink Singh 01 Oct '09

01 Oct '09

Hi Developers, I need to configure my audio device such as to use only LEFT or RIGHT Channel. I have carried out following steps to proceed with 1) Opened PCM playback Device 2) Configured its hardware and Software parameters 3) Open mixer 4) Attached Mixer to playback device 5) Registered Mixer 6) Loaded the mixer 7) Get the first element of mixer A) Now when i use following API, snd_mixer_selem_is_playback_mono(aElem); i found mixer is using MONO channel. But i am configuring my PCM Device for 2 channels I am using snd_mixer_selem_set_playback_switch() API to set right or Left channel to MUTE {Does not work} If i set all channels to Mute {Works fine} snd_mixer_selem_set_playback_switch_all (aElem, 0) Also i am able to set SND_MIXER_SCHN_MONO to mute {works fine} snd_mixer_selem_set_playback_switch(aElem, SND_MIXER_SCHN_MONO, 0) B) There is also an API in my MIXER driver to set the right and left balance, but i did not find any API in Application to do so {Is there any such API in Application}. Thanks in Advance, Navin Kumar Singh Tata Consultancy Services DLF Akruti Info Park, PN 28, Rajiv Gandhi Infotech Park Hinjewadi, PHASE II Pune - 411057,Maharashtra India Ph:- +91 20 66752458 Cell:- +919860227357 Mailto: navink.singh(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 01 Oct '09

01 Oct '09

The following changes since commit 140318aaa924ce9664ff59366993228cf1547f1d: Takashi Iwai (1): ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Mark Brown (6): ASoC: Clean up WM8974 PLL configuration Merge branch 'upstream/wm8974' into for-2.6.33 ASoC: Factor out analogue platform data from WM8993 Merge remote branch 'takashi/fix/asoc' into for-2.6.32 Merge remote branch 'takashi/topic/asoc' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 include/sound/soc-dai.h | 1 + include/sound/soc.h | 1 + sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++------------------ sound/soc/codecs/wm8974.c | 23 +++++---- sound/soc/codecs/wm8993.c | 36 +++---------- sound/soc/codecs/wm_hubs.c | 35 +++++++++++++ sound/soc/codecs/wm_hubs.h | 5 ++ sound/soc/davinci/davinci-i2s.c | 2 + sound/soc/davinci/davinci-mcasp.c | 17 +++---- sound/soc/davinci/davinci-pcm.c | 21 +++++++- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/pxa/Kconfig | 2 +- sound/soc/soc-core.c | 26 +++++++--- 13 files changed, 161 insertions(+), 107 deletions(-)

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[alsa-devel] Tascam
by humbert.olivier.1＠free.fr 01 Oct '09

01 Oct '09

Hi all in first thanks for your work, it permit us to make music with GNU Linux :) I'm Olivier from France. I work on linuxMAO.org website. It's a french website for music assisted by computer. In this website, i make an article for make work the tascam us-224 on GNU Linux. Sometime I was angry because all is'nt recognize on my sound card. (Like 4 faders for 4 tracks). So I make a letter in what i said something like that : "Hey Tascam, it's cool for you ? For me it's cool, because i buy your product and it's a good one, and now, i play music with my friend and children and many people. But, all doesn't work with my OS, GNU/Linux it's damage because you don't see that many people on linux and other OS, have to go to your concurent (like M-audio or RME) because they're more open than you. blablabla blablabla ..." They don't answer. ... :/ Another boy make another one few month ago. ... They don't answer too. ... :/ ... Last week, a new boy email me that, he was very happy because my website page for the TASCAM us 224 help him to make work on linux. (it was great, i'm happy) So i speak with him and said to him "hey, you can make an email to Tascam (in fact sennheiser in france, the official distributor for tascam products) and say to them "hello boys, can you give us the source code or specifications blablabla" " ... and this one, he answered positively ! The sennheiser man said to us that he speak with japanese guys who said "what they needs?". So, i'm not developper I search for the (one or many) people who made the tascam us 224 (and 122 and 448 i think) driver for ALSA system. because, we've got an open way to make it better. Or another one motived ! The ball is here now. Thanks for read and for your work. Olivier

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[alsa-devel] [PATCH 0/1 - try2] ASoC: debugfs support improvement
by Peter Ujfalusi 01 Oct '09

01 Oct '09

Hello, I have noticed that the support for multiple cards/codecs via the debugfs is kind of broken. The second codec, which tries to create the debugfs files would emit the following messages: ASoC: Failed to create codec register debugfs file Failed to create pop time debugfs file ... The following patch fixes this by adding a directory per codec to host the files. The additional directory is named as: {dev_name(socdev->dev)}-{codec->name} for example: debugfs/asoc/soc-audio.0-twl4030/... So it is easier to find the correct codec we are looking for. Note: The patch accidentally fixes one trailing space in the soc_pcm_apply_symmetry function. --- Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs include/sound/soc.h | 1 + sound/soc/soc-core.c | 28 ++++++++++++++++++++-------- 2 files changed, 21 insertions(+), 8 deletions(-)

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[alsa-devel] [PATCH] ASoC: Use codec mutex in dapm_set_pga()
by ext-eero.nurkkala＠nokia.com 01 Oct '09

01 Oct '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> codec->mutex seems required if widget->* values are altered. Moreover, snd_soc_dapm_put_volsw() may alter widget->saved_value. dapm_set_pga() uses widget->saved_value in a for loop which now has a distant chance of getting out of control. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> --- sound/soc/soc-dapm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f79711b..d6fb6c4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -349,6 +349,7 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + mutex_lock(&widget->codec->mutex); if (power) { int i; /* power up has happended, increase volume to last level */ @@ -373,6 +374,7 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) } widget->muted = 1; } + mutex_unlock(&widget->codec->mutex); } return 0; } -- 1.6.0.4

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[alsa-devel] adding hrtimer based delay to ioplug ALSA plugin on ARM9 (AT91) platform ?
by Stefan Schoenleitner 01 Oct '09

01 Oct '09

Hi, I'm currently developing an ALSA ioplug plugin which should transfer audio samples over UART to a DSP (I already wrote that in a previous post). The DSP wants to receive 156 - 164 audio samples with a sampling rate of 8 kHz and a format of S16_LE each 20 msec (which is more or less the number of audio samples multiplied by the period length of the sampling rate). My idea is to add a high precision delay to the transfer callback of the ALSA plugin. Unfortunately, I found out that the hrtimer resolution on the system is as low as 10ms: ------------------------------------------------------------------- kernel: hrtimer resolution: 0 sec, 10000000 nsec, loops_per_jiffy: 448512 ------------------------------------------------------------------- * Does this mean that if I use nanosleep for example each delay has to be an integral multiple of the hrtimer precision and for example a delay of 15ms would not be supported (since it is no integral multiple of 10ms) ? * It is possible to increase the precision (e.g. by changing the kernel configuration) ? * Do you know a better solution for the task, where the task is to send and receive audio samples to/from the DSP over UART and the timing is critical ? cheers, Stefan

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[alsa-devel] ioplug plugin programming: HW_PERIOD_BYTES and HW_PERIODS constraints ?
by Stefan Schoenleitner 01 Oct '09

01 Oct '09

Hi, I'm currently developing an ALSA ioplug plugin that should transfer audio samples to a DSP over UART. In my plugin code I set up the hardware constrains (HW_PERIOD_BYTES and HW_PERIODS being among them) and then I utilize the transfer callback to send the audio chunk over UART. The DSP application would like to have exactly 156 - 164 audio samples at a time, where the sampling rate has to be 8 kHz and the format is S16_LE. In between there has to be a delay of 20ms (which is equal to 164 audio frames at a sampling rate of 8 kHz (i.e. 164 * 125us = 20,5 ms). For this reason I configured HW_PERIOD_BYTES with these values. The result is that each time the transfer callback gets called, exactly 312 bytes (thus 164 frames with each one having 2 bytes) are transfered. However, what I do not understand is the setup of HW_PERIODS which is AFAIK the number of periods being transfered at a time ? * What is the influence of this constraint on my ioplug plugin ? * How does it change the way data is being transfered with the transfer callback ? * What would be the correct setup for the mentioned application ? cheers, stefan

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[alsa-devel] [PATCH] ASoC: fix kconfig order of Blackfin drivers
by Mike Frysinger 01 Oct '09

01 Oct '09

From: Barry Song <barry.song(a)analog.com> Some of the Blackfin options don't directly follow the kconfig options they depend on, so kconfig is unable to display the proper tree. So sort the options such they expand/collapse properly. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++++++---------------------- 1 files changed, 49 insertions(+), 49 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index ac927ff..f91dcb1 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,15 +7,6 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. -config SND_BF5XX_TDM - tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" - depends on (BLACKFIN && SND_SOC) - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in TDM - mode. - You will also need to select the audio interfaces to support below. - config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE Enter the GPIO used to control AD73311's SE pin. Acceptable values are 0 to 7 +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + +config SND_BF5XX_SOC_AD1938 + tristate "SoC AD1938 Audio support for Blackfin" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1938 + help + Say Y if you want to add support for AD1938 codec on Blackfin. + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN @@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT Say y if you want AC97 driver to support up to 5.1 channel audio. this mode will consume much more memory for DMA. +config SND_BF5XX_HAVE_COLD_RESET + bool "BOARD has COLD Reset GPIO" + depends on SND_BF5XX_AC97 + default y if BFIN548_EZKIT + default n if !BFIN548_EZKIT + +config SND_BF5XX_RESET_GPIO_NUM + int "Set a GPIO for cold reset" + depends on SND_BF5XX_HAVE_COLD_RESET + range 0 159 + default 19 if BFIN548_EZKIT + default 5 if BFIN537_STAMP + default 0 + help + Set the correct GPIO for RESET the sound chip. + +config SND_BF5XX_SOC_AD1980 + tristate "SoC AD1980/1 Audio support for BF5xx" + depends on SND_BF5XX_AC97 + select SND_BF5XX_SOC_AC97 + select SND_SOC_AD1980 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_SPORT tristate @@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1980 - tristate "SoC AD1980/1 Audio support for BF5xx" - depends on SND_BF5XX_AC97 - select SND_BF5XX_SOC_AC97 - select SND_SOC_AD1980 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 - help - Say Y if you want to add support for AD1938 codec on Blackfin. - config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) @@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM default 0 help Set the correct SPORT for sound chip. - -config SND_BF5XX_HAVE_COLD_RESET - bool "BOARD has COLD Reset GPIO" - depends on SND_BF5XX_AC97 - default y if BFIN548_EZKIT - default n if !BFIN548_EZKIT - -config SND_BF5XX_RESET_GPIO_NUM - int "Set a GPIO for cold reset" - depends on SND_BF5XX_HAVE_COLD_RESET - range 0 159 - default 19 if BFIN548_EZKIT - default 5 if BFIN537_STAMP - default 0 - help - Set the correct GPIO for RESET the sound chip. -- 1.6.5.rc2

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[alsa-devel] [PATCH 24/34] don't use __devexit_p to wrap snd_sgio2audio_remove
by Uwe Kleine-König 01 Oct '09

01 Oct '09

The function snd_sgio2audio_remove is defined using __exit, so don't use __devexit_p but __exit_p to wrap it. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Figo.zhang <figo1802(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/mips/sgio2audio.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index e497525..e24ba5a 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -984,7 +984,7 @@ static int __exit snd_sgio2audio_remove(struct platform_device *pdev) static struct platform_driver sgio2audio_driver = { .probe = snd_sgio2audio_probe, - .remove = __devexit_p(snd_sgio2audio_remove), + .remove = __exit_p(snd_sgio2audio_remove), .driver = { .name = "sgio2audio", .owner = THIS_MODULE, -- 1.6.4.3

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