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

[alsa-devel] Problem setting mixer switch with amixer/alsactl.
by Oliver Ford 30 Oct '08

30 Oct '08

Hi, I have a system with a wm9713 ac97 codec. It has a single speaker connected to SPKL and OUT4 in BTL config. Currently I'm setting up the audio path with a small script using amixer: amixer cset numid=90 2 # Out4 Mux = Inv2 amixer cset numid=95 3 # Left Speaker Out Mux = Speaker amixer cset numid=88 2 # DAC Inv Mux 2 = Speaker amixer cset numid=31 1 # Out4 Playback Switch = on amixer cset numid=2 1,0 # Speaker Playback Switch = L:on, R:off and also: amixer cset numid=62 1 # Speaker Mixer PCM Playback Switch = on If I do all of the above except for setting 'Speaker Mixer PCM Playback Switch' and then un-mute that in the full alsamixer program, the audio works through the speaker. If I run the 'Speaker Mixer PCM Playback Switch' amixer command (either before or after setting it with alsamixer), the audio stops working. Once that's happened I can't get it back at all without a full reboot (even suspend/resume doesn't help). Setting it with alsamixer so it works and then using alsactl to save and restore the config also breaks it. I'm not sure how to go about debugging this one, can anyone help? Also, is there a way to (and should I) set the DAC -> SPKMIX -> INV2 -> OUT4 etc in my platform definition in the kernel code, rather than doing it with amixer? Oliver Ford

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[alsa-devel] [PATCH 0/8] ASoC updates for next
by Mark Brown 30 Oct '08

30 Oct '08

The following changes since commit 15ad9a531a64d46af54839ab13542a81ad4c82a5: Takashi Iwai (1): Merge branch 'topic/fix/asoc' into topic/asoc are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (6): ASoC: Fix handling of DAPM suspend work ASoC: Convert core to use standard debug print macros ASoC: Remove DAPM restriction on mixer control name lengths ASoC: Add PXA SSP support ASoC: Do a warm reset after cold when resetting the WM9713 ASoC: Use finer grained dependencies in SND_SOC_ALL_CODECS Timur Tabi (1): ASoC: Disable automatic volume control in the CS4270 sound driver Troy Kisky (1): ASoC: Allow setting codec register with debugfs filesystem include/sound/soc.h | 4 + sound/soc/codecs/Kconfig | 44 ++- sound/soc/codecs/cs4270.c | 13 + sound/soc/codecs/wm9713.c | 2 + sound/soc/pxa/Kconfig | 13 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa-ssp.c | 929 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa-ssp.h | 47 +++ sound/soc/soc-core.c | 207 ++++++++--- sound/soc/soc-dapm.c | 44 +-- 10 files changed, 1200 insertions(+), 105 deletions(-) create mode 100644 sound/soc/pxa/pxa-ssp.c create mode 100644 sound/soc/pxa/pxa-ssp.h

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[alsa-devel] [PATCH 0/3] ASoC updates for 2.6.28
by Mark Brown 30 Oct '08

30 Oct '08

These patches contain a few updates for 2.6.28. The fix for mono controls is very important, the other two changes less so.

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Re: [alsa-devel] [PATCH] alsa: add beagleboard SoC configuration.
by Steve Sakoman 30 Oct '08

30 Oct '08

On Wed, Oct 22, 2008 at 12:36 PM, David Brownell <david-b(a)pacbell.net> wrote: > On Wednesday 22 October 2008, Tony Lindgren wrote: >> This should go to alsa-devel list with l-o list cc'd. AFAIK, all >> the omap ASoC stuff is in mainline already. I will not push it >> to l-o until we have an ack from alsa-devel. > > There are multiple ASoc bits ... they're not all there yet. > The TWL4030 codec support should be ready to go, now that its > core has merged, plus Overo (and "soon" Beagle) board support. > > The twl4030-core support was the blocking issue; at least some > of the other patches were even acked. > > http://marc.info/?t=122304728600007&r=1&w=2 > > Maybe Steve would resubmit his patches, to merge through ALSA > into 2.6.28-rc? I presume all the feedback from his first round > (beginning of September) got addressed. Yes, all of the feedback from the first round of submissions was addressed. I received an ack for the twl4030 codec driver as well as the Overo driver. At the time, the feedback from the alsa-devel list was that they were OK with these patches living in linux-omap until the twl4030 driver was mainline ready. So Tony merged them into linux-omap at that point. I am happy to resubmit the patches, if that is what folks want. I'm copying alsa-devel so they can weigh in on the decision. Steve

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[alsa-devel] ASoC: pxa2xx-ssp
by Daniel Ribeiro 30 Oct '08

30 Oct '08

Hi! What is needed for [0] to be accepted on mainline linux? [0]http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.gi… -- Daniel Ribeiro

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[alsa-devel] Automatic assignment/configuration of model-id?
by Klaus Schulz 30 Oct '08

30 Oct '08

Hi gents. To take full advantage of the driver I need to assign the right model option to the module - as far as I understand - manually. Is there a way to automatically assign the "model=" option to the driver module. If not, how is a "normal" user supposed to configure the proper model-id to be able to get access to the full feature-set of the particular soundcard? Background: The manual configuration is not known to many people. And many people seem to get quite annoyed about the lack of functionality caused by a generic driver. THX. Cheers Klaus

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[alsa-devel] [PATCH] Fix some minor errors in mpc5200 psc i2s driver
by Jon Smirl 30 Oct '08

30 Oct '08

Fix missing unsigned for irqsave flags in psc i2s driver Make attribute visiblity static Collect all sysfs errors before checking status Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/mpc5200_psc_i2s.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 8692329..809c5c4 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -277,7 +277,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; u16 imr; u8 psc_cmd; - long flags; + unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_i2s->capture; @@ -699,9 +699,9 @@ static ssize_t psc_i2s_stat_store(struct device *dev, return count; } -DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); -DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); +static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); +static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -819,8 +819,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, /* Register the SYSFS files */ rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_i2s->dev, "error creating sysfs files\n");

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[alsa-devel] [PATCH] ALSA: Warn when control names are truncated
by Mark Brown 29 Oct '08

29 Oct '08

This is likely to confuse user interfaces since the end of the control name is interpreted (eg, "Volume", "Switch"). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/core/control.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 6d71f9a..b0bf426 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -225,8 +225,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.id.iface = ncontrol->iface; kctl.id.device = ncontrol->device; kctl.id.subdevice = ncontrol->subdevice; - if (ncontrol->name) + if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); + if (strcmp(ncontrol->name, kctl.id.name) != 0) + snd_printk(KERN_WARNING + "Control name '%s' truncated to '%s'\n", + ncontrol->name, kctl.id.name); + } kctl.id.index = ncontrol->index; kctl.count = ncontrol->count ? ncontrol->count : 1; access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : -- 1.5.6.5

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Re: [alsa-devel] sbxfi - What works/doesn't summary
by Takashi Iwai 29 Oct '08

29 Oct '08

At Wed, 29 Oct 2008 15:54:00 +0000, Jason Harvey wrote: > > The Source wrote: > > The Source пишет: > > > >> Jason Harvey пишет: > >> > >>> Takashi Iwai wrote: > >>> > >>> > >>>>> Sorry that was a bit ambiguous, I'd never tried 96kHz. > >>>>> Have just changed base_rate to 96kHz and it performs identically to > >>>>> 48kHz. > >>>>> Pulse mangles the sound in just the same manner at 48kHz or 96kHz. > >>>>> > >>>>> > >>>> Ah, so you meant mplayer/flash with PA backend? > >>>> > >>>> > >>> Yes. > >>> I also tried mplayer using -ao=alsa or -ao=oss and the sound is still > >>> corrupted, even after killing the pulse daemon. > >>> > >>> If I get the chance I will put the card into a 32bit machine tomorrow > >>> and see how it sounds there. > >>> > >>> Thanks, Jason > >>> > >>> > >>> _______________________________________________ > >>> Alsa-devel mailing list > >>> Alsa-devel(a)alsa-project.org > >>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >>> > >>> > >>> > >> Ok, looks like with the latest snapshot (28.10.2008 15:19:00) > >> pulseaudio works fine. But wine not. The bad thing is I can't get > >> hw_params somehow. When pulseaudio is started hw_params always say > >> 'closed' even if sound is played at this moment. I'll try to remove > >> pulseaudio and test wine again. > >> > >> > > Well, I wasn't lucky. Even without pulseaudio hw_params are 'closed' > > nomatter what. > With the latest unstable I grabbed this afternoon everything is sounding > pretty good right now! Hmm... I haven't changed sbxfi code since yesterday. The last change was: commit 941ec54e47f635f63621d3852253d9ef9b11132f Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Oct 28 08:00:24 2008 +0100 sbxfi - Use 48kHz as default Though, I'm glad to hear that it's working, of course :) Or, did you apply my last patch I posted? > pulseaudio is now working. > > [jason@quad alsa-kernel]$ cat HEAD > 2f280189c7779e0efbeead8eb03b179d141dd196 Merge commit 'stable/master' > > I can get hw_params when there is something playing. > > cat /proc/asound/card0/pcm0p/sub0/hw_params > > access: MMAP_INTERLEAVED > format: S16_LE > subformat: STD > channels: 2 > rate: 96000 (96000/1) > period_size: 2048 > buffer_size: 16384 > > Can't get mplayer to play at the same time as flash in firefox but I > don't really know if I've ever managed to do that under pulse... The flash with PA can work with libflashsupport. It'd be better to use it with flash-10, though. thanks, Takashi

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[alsa-devel] sbxfi - What works/doesn't summary
by Jason Harvey 29 Oct '08

29 Oct '08

Takashi Iwai wrote: > It'd be helpful if someone can summarize the working and non-working > cases. For example, > > global info: > 0. SBXFI model (product name, PCI ID, PCI SSID, cat /proc/asound/cards > entry) > 1. sbxfi driver version (date & HEADs) > 2. base_rate value > 3. system details (x86-64, distro, kernel version, etc) > > for each app: > A. Application name / version > B. working or not-working, problem descriptions > C. ALSA or OSS (you can see it in /proc/asound/card*/pcm0/sub0/hw_params) > D. period_size and buffer_size (ditto, or in kernel message) > E. any special options > > Maybe also nice on Wiki... Hi, Here's my setup and what's currently working. 0. [SB0550 ]: SBXFi - SB XFi (SB0550) SB XFi (SB0550) at 0x3040 irq 28 1. sbxfi driver version (date & HEADs) 26 Oct 2008 ./HEAD - 8682b8b4ced855b01639d85098ca7de23db87c3d Merge branch 'topic/snd-hrtimer' ./alsa-kernel/HEAD - 8f5d017c10e9da0b1e06f674bfc2c764192d090b Merge branch 'topic/sbxfi' 2. base_rate - 48000 3. x86_64, Fedora 9, 2.6.26.6-79.fc9.x86_64 APPS TESTED Mplayer - dev-SVN-r27841-4.3.0 WORKING cat /proc/asound/card0/pcm0p/sub0/hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 1024 buffer_size: 16384 OSS format: S16_LE OSS channels: 2 OSS rate: 48000 OSS period bytes: 4096 OSS periods: 16 OSS period frames: 1024 Firefox 3.0.2 flash-plugin-10.0.12.36-release.i386 WORKING cat /proc/asound/card0/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 1024 buffer_size: 16384 Enemy Territory Quake Wars worked but crashed after a couple of minutes. Graphics card is new so can't be sure what happened. If I set Gnome to use OSS I can get sound from Rhythmbox. Sorry to tag this on to the end of this message, the problems below are likely of my own causing rather than the sbxfi driver... I'm investigating why when I select alsa or pulse in Gnome I get an error on clicking the test button. "audiotestsrc wave=sine freq=512 ! audioconvert ! audioresample ! gconfaudiosink: Could not open audio device for playback." Selecting OSS and clicking test I hear the sine wave. But I cannot hear any desktop effect noises or the system bell. Same issue with KDE. I hear the log on tune but nothing else from the desktop itself. Possibly my machine is just too messed up and needs a re-install, it's been upgraded continuously from around FC3 and could do with an overhaul. Regards, Jason

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