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

[alsa-devel] [PATCH] ASoC/pxa-ssp.c: fix build on pxa2xx
by Daniel Ribeiro 27 Oct '08

27 Oct '08

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[alsa-devel] [PATCH] ASoC/pxa-ssp.c: keep up with pxa api changes
by Daniel Ribeiro 27 Oct '08

27 Oct '08

1 0

[alsa-devel] problem persist on Mac Pro (quad pro) using fedora 9
by S. Aguinaga 27 Oct '08

27 Oct '08

Hello Team, I'm hoping that some of you folks might know the work-around and or fix to my Mac Pro (quad core) installation of alsa ( on fedora 9 ). I wanted to get the record to work copied the same thing I did on my laptop MacBook Pro, which is working ( to some extent, such as with audacity ), but on the Mac Pro tower it has gotten worse. I used to have audio on the front speaker, and now I have none. And I'm not sure what else to try. aplay 1.wav does not generate sound, but alsamixer seems to have all ports on and looking ok. 1. /etc/modprobe.d/sound reads: options snd-hda-intel model=macpro 2. ~/.asound reads ( but my gui volume controls tools don't work ) ... I'm wondering if there anyway to not rely on pulseaudio or if I need to have pulseaudio working on my fedora 9 installation. pcm.pulse{ type pulse } ctl.pulse { type pulse } pcm.phononpulse { type plug slave.pcm { type pulse } hint { show on description "PulseAudio" } } 3. the file /etc/asound.conf reads the following ... so I wonder if this is enough and should I erase .asoundrc ? If so, what else do I need to get audio working. pcm.!default { type hw card Intel } ctl.!default { type hw card Intel } I also have asound.state and where should I check to see if it is ok? // Sal // Northwestern University - HARP

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[alsa-devel] 'multi' plugin signals POLLOUT but snd_pcm_avail_update() returns 0
by Lennart Poettering 27 Oct '08

27 Oct '08

Heya! There's a apparently a bug in the 'multi' plugin. When it is used it will pretty often signal POLLOUT when used with snd_pcm_poll_descriptors_revents() and the subsequent call to snd_pcm_update_avail() will still return 0. Doesn't look right to me. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Recording problem with pulse plugin
by Takashi Iwai 27 Oct '08

27 Oct '08

Hi Lennart, I got a bug report about the recording with pulse alsa-plugin. It gets just -EIO error. I checked it, and can reproduce it actually. The problem appears to happen since 1.0.18rc1, and the affecting commit is: commit 90c32999189cf6f5c63f40dadb8076eca379713b Author: Lennart Poettering <mznyfn(a)0pointer.de> Date: Fri Jul 18 21:45:47 2008 +0200 send both an uncork and a trigger in _start() The problem is that pa_stream_trigger() returns NULL for capture. I made a quick fix for this (the patch attached below), but not entirely sure whether it's the right thing. Please check. thanks, Takashi diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index a5f794f..3d15ff0 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -145,18 +145,15 @@ static int pulse_start(snd_pcm_ioplug_t * io) u = pa_stream_trigger(pcm->stream, pulse_stream_success_cb, pcm->p); - if (!u) { - pa_operation_unref(o); - err = -EIO; - goto finish; - } pcm->underrun = 0; err_o = pulse_wait_operation(pcm->p, o); - err_u = pulse_wait_operation(pcm->p, u); + if (u) + err_u = pulse_wait_operation(pcm->p, u); pa_operation_unref(o); - pa_operation_unref(u); + if (u) + pa_operation_unref(u); if (err_o < 0 || err_u < 0) { err = -EIO;

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[alsa-devel] NVIDIA HDMI Support
by Robert Vincent Krakora 27 Oct '08

27 Oct '08

Hello, I see that NVIDIA HDMI support is in the tip but not in RC3. Will it be in RC4? Thanks in advance. Best Regards, -- Rob Krakora Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

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[alsa-devel] ASOC: For SND_SOC_DAIFMT_IB_IF what is invert baseed on?
by Richard Zhao 27 Oct '08

27 Oct '08

Hi, include/sound/soc.h /* * DAI hardware signal inversions */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */ #define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ #define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ What are frame cock and bit clock invert based on? I2S, PCM or some else bus protocols? Or just high level voltage or low level voltage? Thanks Richard

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[alsa-devel] ASoC: Merging at91 and avr32 dirs - test report
by ext-jukka.hynninen＠vaisala.com 27 Oct '08

27 Oct '08

I tested the patches and got everything working with minor changes. However, there seems to be some problem with PM, occasionally ASoC is stuck with Standby mode (looking at dapm_widgets) when I start the playback. Restarting the player brings the sound back. I'll carry out some further investigation and get back to this. My kernel is 2.6.27-rc3, and machine driver is adapted from playpaq_wm8510.c to run on at91rm9200 board. The SSC peripheral clock was disabled more often than it was enabled, and that catched the bug trap in arch/arm/mach-at91/clock.c. I'm not sure if this was the real problem, but it's the way I managed to solve it. -Jukka diff -uprN a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c --- a/sound/soc/atmel/atmel_ssc_dai.c 2008-10-10 17:23:46.000000000 +0300 +++ b/sound/soc/atmel/atmel_ssc_dai.c 2008-10-10 17:26:06.000000000 +0300 @@ -261,11 +261,11 @@ static void atmel_ssc_shutdown(struct sn spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - if (ssc_p->initialized) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dai: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + free_irq(ssc_p->ssc->irq, ssc_p); ssc_p->initialized = 0; }

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[alsa-devel] [PATCH 1/1] ASoC: Blackfin: update SPORT0 port selector (v2)
by Bryan Wu 27 Oct '08

27 Oct '08

From: Cliff Cai <cliff.cai(a)analog.com> - Setting the TFS pin selector for SPORT 0 based on whether the selected port id F or G. If the port is F then no conflict should exist for the TFS. When Port G is selected and EMAC then there is a conflict between the PHY interrupt line and TFS. Current settings prevent the conflict by ignoring the TFS pin when Port G is selected. This allows both ssm2602 using Port G and EMAC concurrently. - some code cleanup Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/bf5xx-i2s.c | 34 ++++++++++++++++++++++------------ 1 files changed, 22 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 827587f..e020c16 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = { } }; -static u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, -}; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) @@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } -- 1.5.6

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[alsa-devel] Only one element for multichannel mixer controls please!
by Lennart Poettering 27 Oct '08

27 Oct '08

Heya! It seems that quite a few ALSA drivers for multichannel sound cards export three seperate stereo mixer controls to control front, rear and center/lfe independantly. That sucks because a client (such as PulseAudio) cannot make use of this. Instead they should export a single 6ch mixer control! The "amixer scontents" output looks very strange for these drivers since it lists a "Rear" control that as a "Front Left" and a "Front Right" channel. So, what is it now: front or rear? Is there any reason for this current driver behaviour? I mean, the mixer knows channel ids like SND_MIXER_SCHN_REAR_LEFT perfectly well. Or is it that those devices have some devices joined and this cannot be expressd with a single boolean "joined" for each control? Please enlighten me! Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] patch for alsa-info.sh to quote [ -d $SYSFS ]
by Luis Yanes 27 Oct '08

27 Oct '08

Found a couple of bugs in http://www.alsa-project.org/alsa-info.sh while running it in my system. * old uname <= 2.0.11 doesn't accept -o argument * $SYSFS is not quoted while checking if directory exist and fails if empty Attached is a small patch for the first of them, but think that all variables also should be quoted to avoid future problems. Regards. --- alsa-info.sh.orig Sat Oct 25 14:16:43 2008 +++ alsa-info.sh Sat Oct 25 14:18:44 2008 @@ -373,7 +373,7 @@ echo "" >> $FILE fi -if [ -d $SYSFS ] +if [ -d "$SYSFS" ] then echo "!!Loaded sound module options" >> $FILE echo "!!--------------------------" >> $FILE

2 1

[alsa-devel] using dshare plugin to create multiple independent audio outputs for multiseat out of 7.1 internal audio card
by Jelle de Jong 27 Oct '08

27 Oct '08

Hello everybody, I am looking for ways to get good cheap audio outputs per user for multiseat systems. A nice good way would be to be able to use the available resources of motherboards than now come often with 8-channel audio cards with 6 or more audio outputs ... (6 users) I have seen something about dshare plugin that might be able to do this, but I have no idea how this works or how and if it is possible to configure this for multiple users? Would somebody be able to do some experimenting and maybe make some examples with some .asoundrc quotes? Chipset integrated by Realtekr ALC888/ALC888T - Flexible 8-channel audio with jack sensing - Compliant with Azalia 1.0 Spec Any ideas, solutions? Thanks in advance, Jelle

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[alsa-devel] [PATCH1/4] jack: lineout support to jack abstraction layer
by Matthew Ranostay 27 Oct '08

27 Oct '08

This patch introduces support for reporting SW_LINEOUT_INSERT detection events via the jack abstraction layer. Also adds a SND_JACK_LINEOUT define to the input system header. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/include/linux/input.h b/include/linux/input.h index b86fb55..fa9127d 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -657,6 +657,7 @@ struct input_absinfo { #define SW_RADIO SW_RFKILL_ALL /* deprecated */ #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ +#define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b..7cb25f4 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -35,6 +35,7 @@ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, + SND_JACK_LINEOUT = 0x0004, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index bd2d9e6..4978db1 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -102,6 +102,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_HEADPHONE) input_set_capability(jack->input_dev, EV_SW, SW_HEADPHONE_INSERT); + if (type & SND_JACK_LINEOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_LINEOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -153,6 +156,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_LINEOUT) + input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, + status & SND_JACK_LINEOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE);

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[alsa-devel] When running aplay, reported:ALSA lib pcm.c:2202:(snd_pcm_open_noupdate) Unknown PCM plughw:0, 0
by zhangal 26 Oct '08

26 Oct '08

Hi, everybody I am dealing with OMAP730 alsa driver based on kernel 2.6.25. The alsa core drivers are coming from linux-2.6.25/sound and I wrote my own alsa driver for OMAP730. Now I can insmod my driver correctly and the modules I installed are: Module Size Used by Not taintedsnd_omap_alsa_pers2 15868 0snd_pcm 66660 1 snd_omap_alsa_pers2snd_timer 19492 1 snd_pcmsnd 45972 3 snd_omap_alsa_pers2,snd_pcm,snd_timersnd_page_alloc 6152 1 snd_pcmsoundcore 6020 1 sndAnd device nodes are made automatically [root@/mnt/temp]#ls -l /dev/snd/crw-rw-r-- 1 root root 116, 5 Jan 1 00:16 controlC0crw-rw-r-- 1 root root 116, 4 Jan 1 00:16 pcmC0D0ccrw-rw-r-- 1 root root 116, 3 Jan 1 00:16 pcmC0D0pcrw-rw-r-- 1 root root 116, 2 Jan 1 00:16 timerIt seems the necessary nodes are all there, but after I set the ALSA_CONFIG_PATH=/mnt/temp and then run aplay I got errosrs: [root(a)/mnt/temp]#./aplay -D plughw:0,0 test.wavALSA lib pcm.c:2202:(snd_pcm_open_noupdate) Unknown PCM plughw:0,0aplay: main:583: audio open error: No such file or directory The PCM device can not be recognized. What's happening here? Did anyboady meet this problem before? NOTE: The aplay is compiled from alsa-utils-1.0.18rc3.tar.bz2 and alsa-lib-1.0.18rc3.tar.bz2 Thanks a lot for any help! Another question, why can not I see this email I post? 用MSN有问题？客服机器人小美帮你忙！立刻和小美聊天！ _________________________________________________________________ 一边聊天一边快速搜索，并把结果共享给好友，立刻试试！ http://im.live.cn/Share/18.htm

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[alsa-devel] When running aplay, reported:ALSA lib pcm.c:2202:(snd_pcm_open_noupdate) Unknown PCM plughw:0, 0
by zhangal 26 Oct '08

26 Oct '08

Hi, everybody I am dealing with OMAP730 alsa driver based on kernel 2.6.25. The alsa core drivers are coming from linux-2.6.25/sound and I wrote my own alsa driver for OMAP730. Now I can insmod my driver correctly and the modules I installed are: Module Size Used by Not taintedsnd_omap_alsa_pers2 15868 0snd_pcm 66660 1 snd_omap_alsa_pers2snd_timer 19492 1 snd_pcmsnd 45972 3 snd_omap_alsa_pers2,snd_pcm,snd_timersnd_page_alloc 6152 1 snd_pcmsoundcore 6020 1 snd And device nodes are made automatically [root@/mnt/temp]#ls -l /dev/snd/crw-rw-r-- 1 root root 116, 5 Jan 1 00:16 controlC0crw-rw-r-- 1 root root 116, 4 Jan 1 00:16 pcmC0D0ccrw-rw-r-- 1 root root 116, 3 Jan 1 00:16 pcmC0D0pcrw-rw-r-- 1 root root 116, 2 Jan 1 00:16 timer It seems the necessary nodes are all there, but after I set the ALSA_CONFIG_PATH=/mnt/temp and then run aplay I got errosrs: [root(a)/mnt/temp]#./aplay -D plughw:0,0 test.wavALSA lib pcm.c:2202:(snd_pcm_open_noupdate) Unknown PCM plughw:0,0aplay: main:583: audio open error: No such file or directory The PCM device can not be recognized. What's happening here? Did anyboady meet this problem before? NOTE: The aplay is compiled from alsa-utils-1.0.18rc3.tar.bz2 and alsa-lib-1.0.18rc3.tar.bz2. _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[alsa-devel] volume scaling of usb audio class headset
by Jelle de Jong 25 Oct '08

25 Oct '08

Hello everybody, I got a question, I got a few usb headsets that seem to apply to the usb audio class and the volume scale is way to big with alsa. with the control on 10 of the 100 you are almost getting deaf. So my question is how does alsa make an scaling for such devices and how to fix it so that the full range can be used in a normal volume spectrum. Thank in advance, Jelle

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Re: [alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Jelle de Jong 25 Oct '08

25 Oct '08

Sean McNamara wrote: > On Wed, Oct 22, 2008 at 1:48 PM, Jelle de Jong > <jelledejong(a)powercraft.nl> wrote: >> Hello everybody, >> >> I have a four to eighth muliseat debian linux system here and it is >> missing sound for the users. I was hoping the alsa team can help me with >> this issue. > > All sound issues on Linux are extremely application-specific. Getting > sound to work without any specific application in mind is asking for > trouble, because there is no single configuration of the sound stack > (which, if you expand into historical sound APIs and servers, contains > about 20 different ways to play sound) that will satisfy any > application without further configuration. > > Since this is not a really well-formed question, I will proceed with > the assumption that you want to be able to run, for example, > > aplay /usr/share/sounds/pop.wav > > in the console, and have it produce sound in the "appropriate" > headset. Here's how you proceed: > > 1. Each person on the multiseat box must have their own UNIX user account. > 2. Each user account must have an ~/.asoundrc file. > 3. Each ~/.asoundrc file must redefine the "default" device. How you > redefine it depends on a lot of things: > (a) Do you want _direct_ hardware access to the sound card, i.e. one > application at a time? > (b) Do you want to use ALSA's built-in software mixing, i.e. dmix? > (c) How many channels? > (d) Do you want to use PulseAudio? > (e) Do you want to use JACK? > (f) For each USB headset, what is its corresponding "raw device > string"? This will be something like hw:n, where n is a number > starting from 0, depending on the order in which the USB drivers > initialize each USB controller. > > All of the above questions, and others, will determine exactly how > each user configures their ~/.asoundrc file. > > The *extremely naive* (not recommended) example of an ~/.asoundrc is like: > > pcm.!default { > card 3 > } > > to have ALSA clients play sound, by default, to the fourth sound > device that happened to be initialized. > >> I have usb headsets and usb audio devices and all users have there own >> usb hub. There are no internal audio cards in the multiseat server. >> >> How can we configure an system that the user can listen to audio only to >> the devices connected to his usb hub. > > Well, this is guaranteed by design, isn't it? Without physical access > to someone else's headset, how would I get access to the audio streams > being played to that headset? I don't think this really expresses what > your problem is. > > If we flip your question around, it might be a little more > interesting: "How can we configure a system so that each user's > applications will only play audio to that user's corresponding > headset?" > > You can interpret this question in two ways: > 1. "How can we _prevent_ users from playing sound to each other's > sound cards?" [mutually untrust[ed/ing] users] > or > 2. "How can we configure the default settings so that, if untampered > with, a user's applications will play sound to that user's own headset > and not anyone else's?" [mutually trust[ed/ing] users] > > The lack of hardware mixing actually comes back to being a security > benefit to user space control in this kind of situation. Recall that > if I run > > aplay --device=hw:0 /usr/share/sounds/pop.wav > > then as long as the aplay program is playing sound to hw:0, it is > *impossible* (short of some really devious hacking in alsa-lib or the > kernel) for another application -- from this user, or another -- to > also play sound to hw:0 at the same time. I'll call this claim "No > Native SW Mixing". > > Now, let's look at what is required to satisfy the question in each of > the two above cases. > > Trusted: > 1. We know that users aren't going to maliciously try and hurt one > another's ears by setting someone else's mixer really loud and playing > static to them. > 2. We know that users won't try to tamper with eachother's processes > or configuration files; once things are configured correctly, it'll > stay that way. > 3. By 'No Native SW Mixing', we know that users can *theoretically* > hog another user's sound device whenever that user has zero > applications currently using it. > 4. Since users trust each other, they won't do this. > 5. We can use something like dmix, or just use hw:0 if users don't > need simultaneous app output, and things will be fine. > 6. dmix is a viable option to implement software mixing. > 7. PulseAudio is also a viable option to implement software mixing. > > Untrusted: > 1. Users may want to maliciously try and hurt one another's ears by > setting someone else's mixer really loud and playing static to them. > 2. Users may try to tamper with eachother's processes or configuration > files; once things are configured correctly, it could get messed up > again. [There's not much you can do to prevent this ... at least not > based on existing open source tools.] > 3. By 'No Native SW Mixing', we know that users can *theoretically* > hog another user's sound device whenever that user has zero > applications currently using it. > 4. Since users do not trust each other, there is potential for 3 to happen. > 5. We must use 'No Native SW Mixing' to our advantage, to create a > walled garden around each user's sound experience. We need a process > that, once started by a user, will permanently "hog" the sound card > that user wants to use. This process should, ideally, allow other > processes run by that user (and by that user only) to access their USB > headset for output or input. We need a "gatekeeper" that is sensitive > to the context in which an app is being run. > 6. dmix is a NOT viable option to implement software mixing, because: > (a) Once all dmix applications close their streams, the audio device > hw:n is available, and some malicious user can decide to acquire it, > abuse it, and prevent its rightful user from accessing it with his own > applications. > (b) dmix does not restrict users from outputting sound based on their > UID or session. It is NOT a gatekeeper, just an open gate. > 7. PulseAudio is [perhaps the best] viable solution to implement > software mixing for each respective user. It is also a viable solution > to preventing users from taking control of one another's sound > devices. > > Configuring pulseaudio for a multi-soundcard multi-user system is out > of the scope of this mailing list, but suffice it to say that it can > be done. You will want to run one PulseAudio daemon per user, and each > daemon will hold exclusive control over exactly one USB headset. Each > daemon will only accept clients' requests for audio I/O from the user > who started the daemon. Do make sure that you disable > module-suspend-on-idle, or PulseAudio will actually give up the sound > device after a period of inactivity, making it insecure in an > untrusted environment. > > BTW, if this project is for a commercial interest, you usually pay > someone for this kind of in-depth analysis. If you're that someone, > you might want to read up on some of the underlying technologies so > that you can develop this sort of reasoning on your own. I can > envision a very real situation for both the trusted and the untrusted > environment, so there is definitely a demand for this kind of > specialist. Trusted environments are popular at workplaces, and > untrusted multi-seat boxes are popular at computer labs, public > libraries, etc. > >> Any help is really appreciated. > > Please let me know if this information was helpful -- it will help me > gauge, in turn, whether I should spend my time writing up these sorts > of answers on the ML. FWIW, I searched around for existing articles > touching this subject, but I couldn't find anyone who addresses this > specific issue accurately or in enough detail. > > Sean > > P.S. - I once spoke to a user on IRC who was having intermittent > problems with multiple USB headsets. It turns out that the USB > controllers in the computer didn't have enough bandwidth, or power (or > both) to allow all of the headsets to play at once. If, after > configuring, you run into issues where only so many users can play > sound at once, then you should invest in additional PCI or > (preferably) PCI-Express USB controllers. Note that adding hubs > upstream of a USB controller does absolutely nothing to lessen the > load on the USB controllers that are integrated into your motherboard. > >> Thank in advance, >> >> Jelle Thank you Sean, this is by far the most extensive response i received in months with an "open" question on an mailinglist, this really is a good character, that I will reward if you sent an paypal address :-p. I have extensive experience configuring alsa and usb headset, little new info was delivered for me but i think others will really like your response too. My main issue is how to start alsa when there are no audio cards on the system using debian sid? so i can use bluetooth or hotplug usb audio devices...anybody? Then I think it will become an udev issue to regulate a fixed group and permissions for usb audio devices. I was kind of hoping somebody had done this kind of udev ruling for usb audio devices and can provide some examples this will save me time...anybody? PS. I have also noticed usb audio problems (also posted them to irc maybe 1 year ago) in the past with usb hubs and other kinds of things, these things can brake an design or make it:-p So thank you very much for this great response, and keep up doing this kind of work. Kind regards, Jelle

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[alsa-devel] bluetooth alsa and software mixing
by Jelle de Jong 25 Oct '08

25 Oct '08

Hello everybody, I have a bunch of bluetooth audio devices, and I can currently only play one audio stream as once, and i would like to be able to listen to radio and other sounds at the same time. So i think some sort of software mixing is needed to get this working. I attached my .asound / also configuration. How can I configure my bluetooth system so that I can listen to different sources of audio at the same time? Thank for any information in advance, Kind regards, Jelle

2 2

[alsa-devel] ALSA 1.0.18rc3 packages
by Jaroslav Kysela 25 Oct '08

25 Oct '08

Hello all, ALSA 1.0.18rc3 packages are available for download on http://www.alsa-project.org. I expect to release 1.0.18 final in two weeks. Changes from 1.0.18rc2: http://www.alsa-project.org/main/index.php/Changes_v1.0.18rc2_v1.0.18rc3 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH4/4] hda: add support for jack detection on IDT codecs.
by Matthew Ranostay 25 Oct '08

25 Oct '08

This patch adds support to the IDT codec families to report jack status to the jack abstraction layer. This required some reorganization in the stac92xx_unsol_event function in which the index value is changed to reporting the nid with the event. Also adds an sigmatel_jack struct to keep track of the nid relation to the jack abstraction layer instance. Also adds functions to set and retrieve data values for each nid, this is used in stac92xx_unsol_event to retrieve the GPIO mask for STAC_VREF_EVENT. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d106ea5..c24d22f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,7 @@ #include "hda_patch.h" #include "hda_beep.h" +#define STAC_INSERT_EVENT 0x10 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 #define STAC_VREF_EVENT 0x40 @@ -129,6 +130,17 @@ enum { STAC_927X_MODELS }; +struct sigmatel_event { + hda_nid_t nid; + int data; +}; + +struct sigmatel_jack { + hda_nid_t nid; + int type; + struct snd_jack *jack; +}; + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -161,6 +173,12 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; hda_nid_t *dac_list; + /* jack detection */ + struct snd_array jacks; + + /* events */ + struct snd_array events; + /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; @@ -216,9 +234,6 @@ struct sigmatel_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ - /* jack detection */ - struct snd_jack *jack; - /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; @@ -2458,13 +2473,15 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; spec->hp_switch = ucontrol->value.integer.value[0]; /* check to be sure that the ports are upto date with * switch changes */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, (STAC_HP_EVENT | nid) << 26); return 1; } @@ -2504,7 +2521,8 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ * appropriately according to the pin direction */ if (spec->hp_detect) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, + (STAC_HP_EVENT | nid) << 26); return 1; } @@ -3574,13 +3592,70 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } +static int stac92xx_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jack; + int def_conf = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int connectivity = get_defcfg_connect(def_conf); + char name[32]; + + if (connectivity && connectivity != AC_JACK_PORT_FIXED) + return 0; + + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + if (!jack) + return -ENOMEM; + jack->nid = nid; + jack->type = type; + + sprintf(name, "%s at %s %s Jack", + snd_hda_get_jack_type(def_conf), + snd_hda_get_jack_connectivity(def_conf), + snd_hda_get_jack_location(def_conf)); + + return snd_jack_new(codec->bus->card, name, type, &jack->jack); +} + +static int stac92xx_add_event(struct sigmatel_spec *spec, hda_nid_t nid, + int data) +{ + struct sigmatel_event *event; + + snd_array_init(&spec->events, sizeof(*event), 32); + event = snd_array_new(&spec->events); + if (!event) + return -ENOMEM; + event->nid = nid; + event->data = data; + + return 0; +} + +static int stac92xx_event_data(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_event *events = spec->events.list; + if (events) { + int i; + for (i = 0; i < spec->events.used; i++) + if (events[i].nid == nid) + return events[i].data; + } + return 0; +} + static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int event) { - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); + (AC_USRSP_EN | event | nid)); + } } static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) @@ -3623,27 +3698,36 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ - for (i = 0; i < cfg->hp_outs; i++) - enable_pin_detect(codec, cfg->hp_pins[i], - STAC_HP_EVENT); + for (i = 0; i < cfg->hp_outs; i++) { + int type = SND_JACK_HEADPHONE; + hda_nid_t nid = cfg->hp_pins[i]; + enable_pin_detect(codec, nid, STAC_HP_EVENT | nid); + /* jack detection */ + if (cfg->hp_outs == i) + type |= SND_JACK_LINEOUT; + err = stac92xx_add_jack(codec, nid, type); + if (err < 0) + return err; + + } /* force to enable the first line-out; the others are set up * in unsol_event */ stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], - AC_PINCTL_OUT_EN); - stac92xx_auto_init_hp_out(codec); - /* jack detection */ - err = snd_jack_new(codec->bus->card, - "Headphone Jack", - SND_JACK_HEADPHONE, &spec->jack); - if (err < 0) - return err; + AC_PINCTL_OUT_EN); /* fake event to set up pins */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, + (STAC_HP_EVENT | spec->autocfg.hp_pins[0]) << 26); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); } + for (i = 0; i < cfg->line_outs; i++) { + err = stac92xx_add_jack(codec, + cfg->line_out_pins[i], SND_JACK_LINEOUT); + if (err < 0) + return err; + } for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { @@ -3656,6 +3740,11 @@ static int stac92xx_init(struct hda_codec *codec) if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); + err = stac92xx_add_jack(codec, nid, + SND_JACK_MICROPHONE); + if (err < 0) + return err; + enable_pin_detect(codec, nid, STAC_INSERT_EVENT | nid); } } for (i = 0; i < spec->num_dmics; i++) @@ -3697,6 +3786,18 @@ static int stac92xx_init(struct hda_codec *codec) return 0; } +static void stac92xx_free_jacks(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + if (spec->jacks.list) { + struct sigmatel_jack *jacks = spec->jacks.list; + int i; + for (i = 0; i < spec->jacks.used; i++) + snd_device_free(codec->bus->card, &jacks[i].jack); + } + snd_array_free(&spec->jacks); +} + static void stac92xx_free_kctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3717,11 +3818,10 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - if (spec->jack) - snd_device_free(codec->bus->card, spec->jack); - if (spec->bios_pin_configs) kfree(spec->bios_pin_configs); + stac92xx_free_jacks(codec); + snd_array_free(&spec->events); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3804,8 +3904,6 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } - snd_jack_report(spec->jack, - presence ? SND_JACK_HEADPHONE : 0); if (presence) { /* disable lineouts, enable hp */ @@ -3862,24 +3960,57 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx) /* power down unused output ports */ snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); -}; +} + +static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int pin_ctl = + snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, + 0x00); + int type = jacks->type; + if (type == (SND_JACK_LINEOUT + | SND_JACK_HEADPHONE)) + type = (pin_ctl & AC_PINCTL_HP_EN) + ? SND_JACK_HEADPHONE : SND_JACK_LINEOUT; + snd_jack_report(jacks->jack, + get_hp_pin_presence(codec, nid) + ? type : 0); + } + jacks++; + } + } +} static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; - int idx = res >> 26 & 0x0f; + int event = (res >> 26) & 0x70; + int nid = res >> 26 & 0x0f; - switch ((res >> 26) & 0x70) { + switch (event) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); /* fallthru */ + case STAC_INSERT_EVENT: case STAC_PWR_EVENT: - if (spec->num_pwrs > 0) - stac92xx_pin_sense(codec, idx); + if (nid) { + if (spec->num_pwrs > 0) + stac92xx_pin_sense(codec, nid); + stac92xx_report_jack(codec, nid); + } break; case STAC_VREF_EVENT: { int data = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0); + int idx = stac92xx_event_data(codec, nid); /* toggle VREF state based on GPIOx status */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, !!(data & (1 << idx))); @@ -4402,8 +4533,11 @@ again: snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_VREF_EVENT | 0x01)); + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | codec->afg)); + err = stac92xx_add_event(spec, codec->afg, 0x02); + if (err < 0) + return err; spec->gpio_mask |= 0x02; break; } @@ -4802,8 +4936,11 @@ static int patch_stac9205(struct hda_codec *codec) snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_HP_EVENT)); + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | codec->afg)); + err = stac92xx_add_event(spec, codec->afg, 0x01); + if (err < 0) + return err; spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01;

1 0

[alsa-devel] [PATCH3/4] hda: add snd_hda_get_jack* functions
by Matthew Ranostay 25 Oct '08

25 Oct '08

This patch adds snd_hda_get_jack* functions for reporting jack location, device, and connectivity type. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9a8adc6..eaa8b56 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -107,6 +107,52 @@ static void hda_keep_power_on(struct hda_codec *codec); static inline void hda_keep_power_on(struct hda_codec *codec) {} #endif +const char *snd_hda_get_jack_location(u32 cfg) +{ + static char *bases[7] = { + "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", + }; + static unsigned char specials_idx[] = { + 0x07, 0x08, + 0x17, 0x18, 0x19, + 0x37, 0x38 + }; + static char *specials[] = { + "Rear Panel", "Drive Bar", + "Riser", "HDMI", "ATAPI", + "Mobile-In", "Mobile-Out" + }; + int i; + cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; + if ((cfg & 0x0f) < 7) + return bases[cfg & 0x0f]; + for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { + if (cfg == specials_idx[i]) + return specials[i]; + } + return "UNKNOWN"; +} + +const char *snd_hda_get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} + +const char *snd_hda_get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a77ba22..07ddd3d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -859,6 +859,13 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state); int snd_hda_resume(struct hda_bus *bus); #endif +/* + * get widget information + */ +const char *snd_hda_get_jack_connectivity(u32 cfg); +const char *snd_hda_get_jack_type(u32 cfg); +const char *snd_hda_get_jack_location(u32 cfg); + /* * power saving */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 64ab19f..b36d4d0 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -145,32 +145,6 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) -{ - static char *bases[7] = { - "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", - }; - static unsigned char specials_idx[] = { - 0x07, 0x08, - 0x17, 0x18, 0x19, - 0x37, 0x38 - }; - static char *specials[] = { - "Rear Panel", "Drive Bar", - "Riser", "HDMI", "ATAPI", - "Mobile-In", "Mobile-Out" - }; - int i; - cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; - if ((cfg & 0x0f) < 7) - return bases[cfg & 0x0f]; - for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { - if (cfg == specials_idx[i]) - return specials[i]; - } - return "UNKNOWN"; -} - static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +180,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,9 +241,9 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], - get_jack_location(caps)); + snd_hda_get_jack_type(caps), + snd_hda_get_jack_connectivity(caps), + snd_hda_get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps));

1 0

[alsa-devel] [PATCH2/4] hda: dynamic jack id
by Matthew Ranostay 25 Oct '08

25 Oct '08

This patch duplicates the jack->id pointer with kstrdup() to prevent scoping issues from calling autoprobing functions from the HDA section. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/core/jack.c b/sound/core/jack.c index 4978db1..284432f 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device) else input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return 0; @@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (jack == NULL) return -ENOMEM; - jack->id = id; + jack->id = kstrdup(id, GFP_KERNEL); jack->input_dev = input_allocate_device(); if (jack->input_dev == NULL) {

1 0

Re: [alsa-devel] Problem setting mixer switch with amixer/alsactl.
by Mark Brown 24 Oct '08

24 Oct '08

On Thu, Oct 23, 2008 at 11:12:34PM +0100, Oliver Ford wrote: > Am I right in understanding that setting all the controls in amixer > doesn't actually enable the control in the codec chip at that time but > that the powering on/off is done by soc-dapm.c when the the DAC is > actually enabled when playing some audio? This appears to be what is > happening when it does work (looking at the log). I hadn't realised this. Sort of. The power control happens whenever any power-related control changes and whenever audio is started or stopped. > I've put some debugging in pxa2xx-ac97-write and in dapm_power_widgets() > and the log is attached (stripped of non-audio stuff). I'll have a look at these on Monday.

1 0

Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 24 Oct '08

24 Oct '08

At Wed, 22 Oct 2008 06:43:06 +0200, Nir Tzachar wrote: > > On Wed, Oct 22, 2008 at 4:42 AM, Wei Ni <wni(a)nvidia.com> wrote: > > Hi, Takshi > > We have no official plan to support this nforce 630i HDMI audio in linux > > now, > > and we didn't do any test on it yet. So we can't provide the official > > patch for this addition. > > > > I think if someone interest in it and do some test, he can generate > > patch and submit it. > > As I am interested, and willing o do the tests, I would be happy to > know which tests are needed. If you can play both audio streams and non-audio streams over the device by some applications, it's basically confirmed to work :) Takashi

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[alsa-devel] Audio on omap3+twl4030 codec
by naveen krishna ch 24 Oct '08

24 Oct '08

Hi all, I am working on omap3430 and twl4030 combination for audio. I am wokring on 2.6.26 whre the support was not there for omap3 mcbsp. so i ported some stuff from the 2.6.27 latest git tree and now audio is working on omap3 evm. with the very same kernel when i try with my custom board. I am unable to hear audoi. I get audio when i boot out of MMC (x-load and u-boot in mmc and fiel system is nfs). But, when i usse nand(x-load and u-boot in nand and file system is on nfs) i dont get audio out meams aplay and arecord seems working but i dont hear any audio. I probe and checked the Bit clock and Word clock of the I2S which seems are comming when i try a aplay and arecord.. I am sure that hardware has no problem. Can any one tell me what might be the issue and where i have to start debugging the issue. If this is not the right palce to put my question can any one tell me the correct mailing list. -- Thanks, NeyO

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