August 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Configure and use dmix with a new ASoC driver
by Simon Tremblay 18 Aug '08

18 Aug '08

Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done?

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[alsa-devel] [PATCH] opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
by Krzysztof Helt 17 Aug '08

17 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove card pointer from the snd_opl3sa2 structure and break circular reference snd_card->snd_opl3sa2->snd_card. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A very small opl3sa2 clean up. diff -urp linux-ref/sound/isa/opl3sa2.c linux-2.6/sound/isa/opl3sa2.c --- linux-ref/sound/isa/opl3sa2.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/opl3sa2.c 2008-08-15 12:30:58.000000000 +0200 @@ -124,7 +124,6 @@ static int pnpc_registered; #define OPL3SA2_PM_D3 (OPL3SA2_PM_ADOWN|OPL3SA2_PM_PSV|OPL3SA2_PM_PDN|OPL3SA2_PM_PDX) struct snd_opl3sa2 { - struct snd_card *card; int version; /* 2 or 3 */ unsigned long port; /* control port */ struct resource *res_port; /* control port resource */ @@ -222,14 +221,13 @@ static void snd_opl3sa2_write(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_detect(struct snd_card *card) { - struct snd_card *card; + struct snd_opl3sa2 *chip = card->private_data; unsigned long port; unsigned char tmp, tmp1; char str[2]; - card = chip->card; port = chip->port; if ((chip->res_port = request_region(port, 2, "OPL3-SA control")) == NULL) { snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); @@ -298,12 +296,14 @@ static int __devinit snd_opl3sa2_detect( static irqreturn_t snd_opl3sa2_interrupt(int irq, void *dev_id) { unsigned short status; - struct snd_opl3sa2 *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_opl3sa2 *chip; int handled = 0; - if (chip == NULL || chip->card == NULL) + if (card == NULL || card->private_data == NULL) return IRQ_NONE; + chip = card->private_data; status = snd_opl3sa2_read(chip, OPL3SA2_IRQ_STATUS); if (status & 0x20) { @@ -327,8 +327,10 @@ static irqreturn_t snd_opl3sa2_interrupt snd_opl3sa2_read(chip, OPL3SA2_MASTER_RIGHT); snd_opl3sa2_read(chip, OPL3SA2_MASTER_LEFT); if (chip->master_switch && chip->master_volume) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } } return IRQ_RETVAL(handled); @@ -512,9 +514,9 @@ static void snd_opl3sa2_master_free(stru chip->master_volume = NULL; } -static int __devinit snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; + struct snd_opl3sa2 *chip = card->private_data; struct snd_ctl_elem_id id1, id2; struct snd_kcontrol *kctl; unsigned int idx; @@ -650,7 +652,6 @@ static struct snd_card *snd_opl3sa2_card chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; - chip->card = card; card->private_free = snd_opl3sa2_free; return card; } @@ -672,9 +673,12 @@ static int __devinit snd_opl3sa2_probe(s xdma2 = dma2[dev]; if (xdma2 < 0) chip->single_dma = 1; - if ((err = snd_opl3sa2_detect(chip)) < 0) + err = snd_opl3sa2_detect(card); + if (err < 0) return err; - if (request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, "OPL3-SA2", chip)) { + err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + "OPL3-SA2", card); + if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); return -ENODEV; } @@ -694,7 +698,8 @@ static int __devinit snd_opl3sa2_probe(s err = snd_wss_mixer(wss); if (err < 0) return err; - if ((err = snd_opl3sa2_mixer(chip)) < 0) + err = snd_opl3sa2_mixer(card); + if (err < 0) return err; err = snd_wss_timer(wss, 0, NULL); if (err < 0) ---------------------------------------------------------------------- Prosty kredyt na wszystkie potrzeby. Od 1000 zl do 120 000 zl. Bez poreczycieli i zabezpieczen >>> http://link.interia.pl/f1edf

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[alsa-devel] Problem with latest alsa-info.sh patch
by Travis Place 17 Aug '08

17 Aug '08

The latest patch to allow the uploading to alsa-project.org seems to work fine, for the most part, unless you want to use the --pastebin option for the legacy upload to pastebin.ca . It seems that the uploaded data gets truncated. Not sure if this is the script, or the pastebin having problems. Ill look into it a bit more when i have time (its 5:30am here, and i need sleep). Just thought id point this out. I realise the 'default' behaviour is to use alsa-project.org for uploading, but i feel the other pastebin.ca mode should still function(I personally get people to upload there first, and then 'submit' card details to alsa-project.org/db/ afterwards. Thanks, Travis Place (wishie)

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[alsa-devel] [SOUND/SOC] removed unused #include <version.h>
by Huang Weiyi 17 Aug '08

17 Aug '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/wm8753.c sound/soc/codecs/wm9712.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809..dc7b18f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -34,7 +34,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fb7f9a..2f1c91b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h>

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Re: [alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 16 Aug '08

16 Aug '08

Hello Stan, This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" 2. aplay does not produce sound, period! 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: sudo cat /var/log/messages | grep ALSA Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped Let me know if I should send something else. // Salvador // Northwestern University ----- Original Message ---- From: stan <ghjeold_i_mwee(a)cox.net> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Friday, August 15, 2008 11:54:12 AM Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues S. Aguinaga wrote: > Hello, > > I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. > > // Salvador > Fedora uses pulseaudio by default. How are you determining that there is no output? i.e. what program are you using, and how Try aplay -vv some.wav Try aplay -vv -Dhw:0,0 some.wav Try aplay -vv -Dplughw:0,0 some.wav Are there error messages or does it look like it is playing without sound? Are there messages in /var/log/messages regarding sound at boot or while trying to play. I don't use pulseaudio and don't know it very well, but there is more information here. http://www.pulseaudio.org Could you run a script which provides more information about your sound setup and post the link back here? The script is found at http://www.alsa-project.org/alsa-info.sh You can run the script so the results occur locally, by specifying the ---no-upload option. alsa-info.sh --no-upload You would then paste the output into your email instead of providing a link. This information will enable me to determine if your setup is OK, and look for other problems.

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[alsa-devel] [PATCH 1/2] wm8990: Fix routing of left DAC to speaker mixer
by Mark Brown 15 Aug '08

15 Aug '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) This ought to go into 2.6.27 - it's a fairly straightforward bug fix. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 3ecce51..9505a18 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, - {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"}, /* LONMIX */ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, -- 1.5.6.3

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[alsa-devel] WM8750 utilization
by Harald Radke 15 Aug '08

15 Aug '08

Hi there! Sorry, if this is the wrong list for this, just tell me if so (and maybe point out where to go instead *g*). I am trying to improve usage of the wm8750 codec on my Loox720 PDA. Its a PXA272 device. So far WM8750 driver works quite ok (via I2S), however there are some things that are kinda strange. The Loox720 has a headphone jack, a earpiece and a speaker. When driving LOUT1/ROUT1, sound goes via headphones, LOUT2/ROUT2 drives the earpiece. When setting OUT3 on ROUT1, the speaker plays fine. Strange: - when disabling either LOUT2 or ROUT2 (while the other is enabled), the speaker plays the sound (not sure if the earpiece still is active as speaker output is much louder, but I think so) - when disabling ROUT1/LOUT1 with LOUT2/ROUT2 active, the earpiece plays fine, however there is still some output on the headphone (really really silently) - Trying to utilize headphone switch via the WM8750 registers doesn't really work ): depending on the polarity bit either one of the OUTs is either not muted at all or activated at all. Any ideas or hints? Thanks a lot! Harald Radke

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[alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 15 Aug '08

15 Aug '08

Hello, I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. // Salvador

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[alsa-devel] SPDIF audio / non-audio bit
by Ben Stanley 15 Aug '08

15 Aug '08

Dear alsa-devel, I recently advised this list of some problems I experienced with the SPDIF audio / non-audio bit being set inappropriately while using the ca0106 driver [1]. Seeing as I got no replies, I am trying again. When playing AC3 encoded streams (Dolby Digital, DTS), the SPDIF non-audio bit is set. Otherwise, for PCM streams, the SPDIF is set to audio. Could someone explain to me what part of the software is responsible for correctly setting the SPDIF audio / non-audio bit? Is it * alsa-driver * alsa-lib * application ? It seems to me that currently some applications (MythTV, xine) set it correctly, but most other applications ignore it. So if it is set to non-audio (because for example MythTV crashed before re-setting it to audio), then all applications that are not aware of the setting have their audio broken. Could someone set me straight here so that I can try to produce a permanent fix for this? Thanks, Ben Stanley. [1] http://thread.gmane.org/gmane.linux.alsa.devel/54398/focus=55188

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.27-rc3-74-g027ee62
by Takashi Iwai 15 Aug '08

15 Aug '08

At Fri, 15 Aug 2008 12:39:19 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > @@ -362,10 +363,8 @@ static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream) > } > if (substream->pcm->device & 2) > runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID); > - if ((err = add_playback_constraints(runtime)) < 0) { > - kfree(dpcm); > + if ((err = add_playback_constraints(runtime)) < 0) > return err; > - } Run checkpatch.pl please... Takashi

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