July 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] Support for muting channels DAC with CA0106 - SB0570
by Trent Piepho 24 Jul '07

24 Jul '07

This patch adds mute controls for the analog output channels. It does this by sending SPI commands to the DAC on cards like the SB live 24-bit / Audigy SE that have an SPI DAC.

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[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

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[alsa-devel] SuSE 9.1 on Toshiba Satellite A105 - S4004
by Neeraj Vaidya 24 Jul '07

24 Jul '07

Hi, I've installed this distro on my toshiba laptop. In YaST , when I try to configure the sound card , it says no sound card detected. To cut a long story short , I do not get any sounds to work on my laptop. Sound works beautifully on XP , but I am trying to migrate all of my routine activities to linux ;) I am a newbie to the ALSA mailing list. Have a feeling that something to do with ALSA configuration can help me sort out this problem. When I run lspci , I do not see any audio card as being detected. Can someone please help me out ? Also , send me some diagnostic steps , if not the entire solution , so that we can sort this out in a series of emails , if the entire solution can not be provided in one email. Thanks in Advance! -Neeraj --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

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[alsa-devel] [PATCH] s3c24xx-pcm: fix hw_params dma handling
by Arnaud Patard 24 Jul '07

24 Jul '07

Hi, While testing the recently merged driver for the neo1973, I found out that it was not working due to dma allocation failure. The problem is that there's only one dma channel available on s3c2410 and there may be more than one call to pcm_ops->hw_params(), resulting in dma request failures. This patch from Harald Welte fix this. Arnaud

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[alsa-devel] Intel HDA digital out distortion
by Ingmar Stieger 24 Jul '07

24 Jul '07

Hi, a couple of weeks ago, I bought an ASUS mainboard with integrated Intel HDA and an AD1988B codec. Unfortunately, the digital / SPDIF output gets distorted with stock 1.0.14 drivers. It does not matter what the volume levels of the various mixers are set to. I think I might have found the source for this problem. Maybe it's only a workaround, but it solved the issue and digital out now works fine for me, so I thought I'd share. In patch_analog.c, in static struct hda_verb ad1988_spdif_init_verbs[], changeing the forth entry from {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, to {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)} mutes what might be an feedback loop from the analog inputs. The distortion is gone if the input to this widget is muted. On a side note, the settings for the ADCs with NID:08, 09, and 0F looks fishy, too. According to the functional diagram, those should have no amps, and thus should not be muteable. The driver mutes them anyway... I'd like to help somebody who really understands this stuff to solve the problem for good. What would the next step be ? Thanks, Ingmar

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[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

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[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

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[alsa-devel] SNDRV_PCM_FMTBIT_S24_LE vs. SNDRV_PCM_FMTBIT_S24_3LE
by Timur Tabi 24 Jul '07

24 Jul '07

What is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] Fix Kconfig entry for SND_S3C24XX_SOC_NEO1973_WM8753
by Arnaud Patard 23 Jul '07

23 Jul '07

SND_S3C24XX_SOC_NEO1973_WM8753 depends on MACH_GTA01 but the Kconfig entry which is going to be merged is MACH_NEO1973_GTA01. signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org>

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[alsa-devel] [PATCH] Fix tas_suspend/resume build warning
by Stephen Rothwell 23 Jul '07

23 Jul '07

sound/aoa/codecs/snd-aoa-codec-tas.c:750: warning: 'tas_suspend' defined but not used sound/aoa/codecs/snd-aoa-codec-tas.c:760: warning: 'tas_resume' defined but not used Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- sound/aoa/codecs/snd-aoa-codec-tas.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 2f771f5..3cbfe46 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -743,6 +743,7 @@ static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock return 0; } +#ifdef CONFIG_PM /* we are controlled via i2c and assume that is always up * If that wasn't the case, we'd have to suspend once * our i2c device is suspended, and then take note of that! */ @@ -768,7 +769,6 @@ static int tas_resume(struct tas *tas) return 0; } -#ifdef CONFIG_PM static int _tas_suspend(struct codec_info_item *cii, pm_message_t state) { return tas_suspend(cii->codec_data); @@ -778,7 +778,10 @@ static int _tas_resume(struct codec_info_item *cii) { return tas_resume(cii->codec_data); } -#endif +#else /* CONFIG_PM */ +#define _tas_suspend NULL +#define _tas_resume NULL +#endif /* CONFIG_PM */ static struct codec_info tas_codec_info = { .transfers = tas_transfers, @@ -791,10 +794,8 @@ static struct codec_info tas_codec_info = { .owner = THIS_MODULE, .usable = tas_usable, .switch_clock = tas_switch_clock, -#ifdef CONFIG_PM .suspend = _tas_suspend, .resume = _tas_resume, -#endif }; static int tas_init_codec(struct aoa_codec *codec) -- 1.5.2.3

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