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Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Here I send the two also-info files as attachments. Initial problem was: Under kernel 2.6.30, Sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Kernel 2.6.24 with the same files has no issues. >> Then give as attachments later (and don't forget Cc to alsa-devel ML) >> these two files, not URL. It's important because ML is archived >> while pastebin can be removed at any time. Thanks, Christian Steffen -----Ursprüngliche Nachricht----- Von: Takashi Iwai [mailto:tiwai@suse.de] Gesendet: Mittwoch, 29. Juli 2009 17:32 An: Christian Steffen Cc: 'Takashi Iwai' Betreff: Re: AW: [alsa-devel] clicking noise with Sigmatel STAC9202 At Wed, 29 Jul 2009 17:11:18 +0200, Christian Steffen wrote: > > Thanks, > > >> Well, then at least, the contents of /proc/asound/card0/codec#*, > >> /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file > >> generated via "alsactl -f somefile store". These are the most > >> important information. > > There is no pinconfig file and there is also no alsactl tool present. > I could obtain the output from the codec stuff: > > http://pastebin.com/m1b81f47d (old kernel 2.6.24, only the codec info) > http://pastebin.com/m5baeca9a (new 2.6.30) > > that's all I could get for the older kernel as I only have access using > telnet. Then give as attachments later (and don't forget Cc to alsa-devel ML) these two files, not URL. It's important because ML is archived while pastebin can be removed at any time. Takashi

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[alsa-devel] backward/forward binary compatibility checker
by Andrey Ponomarenko 29 Jul '09

29 Jul '09

Colleagues, I'm software engineer from Institute for System Programing of Russian Academy of Sciences and we are developing a free lightweight tool for checking backward/forward binary compatibility of shared C/C++ libraries in OS Linux. It checks interface signatures and data type definitions in two library versions (headers and shared objects) and searches differences that may lead to incompatibility according to ABI standards. We have released beta version of this tool and we'd like you to consider its usefulness for your project. The package with the latest version of tool is in the attachment. The wiki-page with the latest release of binary compatibility checker is http://ispras.linux-foundation.org/index.php/ABI_compliance_checker Andrey Ponomarenko <susanin(a)ispras.ru>

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[alsa-devel] Master channel is only mono
by Sune Mølgaard 29 Jul '09

29 Jul '09

As per gnubien's advice on #alsa on irc.freenode.net, I have run alsa-info.sh - the URL is http://www.alsa-project.org/db/?f=fbe93f301ca214007e66c521cd6c41a17a604c79 . My problem is that the master channel for my card is shown to be mono only, which I don't think seems right... Also, sound in general is rather low. My system is a very new Dell Studio laptop - I have loaded snd-hda-intel with mode=dell-m6 as per the kernel documentation. Best regards, Sune Mølgaard

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[alsa-devel] pulseaudio workaround for alsa-driver bug
by jack craig 29 Jul '09

29 Jul '09

Hi alsa-developers, I wanted to report my recent issues with audio on fc11. after an update from fc10, my audio started failing. not the failure to year any audio, but instead, it would start off fine and then tail spin. i am using mplayer to play an internet asf stream to the local audio. the stream processing failure is described in https://bugzilla.redhat.com/show_bug.cgi?id=514213 i was able to get a workaround from the #pulseaudio guys, but! they encouraged me to report my experience and its fix to you to encourage the fix of this snd_pcm_delay() defect. if you get a driver fix for this, i am happy to try it out when its ready. thx, jackc... -- jack craig jackc(a)LinuxLightHouse.com 831-684-1375 (Office) 831-596-6924 (cell) IM: jackcraigaptos (AIM) _________________________________ This email has been ClamScanned ! www.LinuxLightHouse.com

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Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Takashi Iwai 29 Jul '09

29 Jul '09

At Wed, 29 Jul 2009 16:46:34 +0200, Christian Steffen wrote: > > >>>Is the older kernel as is even with the same user-space stuff? > > Probaly yes, I mean I can start the on-device linux with the newer kernel > and also get clicking noise. > But this way I can't run alsa-info.sh as this system has not all the apps it > needs. > > So I started another ubuntu linux on the device to run alsa-info.sh on the > 2.6.30 kernel. > > I've uploaded the result to: > http://pastebin.com/m5baeca9a Please run it with --no-upload on both kernels, and attach both generated files. > Does this already help? > It's difficult to install apps to the linux which is on the device as it is > very very limited. Well, then at least, the contents of /proc/asound/card0/codec#*, /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file generated via "alsactl -f somefile store". These are the most important information. Takashi

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[alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Hello to all subscribers, I'm new to this list and will start directly with my problem: I'm running some ubuntu linux on a device called "o2 Joggler", which is a digital picture frame with touchscreen control. Its chipsets are from intel, the cpu is an intel atom, and the other hardware devices are controlled by an Intel System Controller Hub "Poulsbo". I'm running kernel 2.6.30 and the newest alsa version 1.0.20. This is lscpi output: >>00:1b.0 Audio Device: Intel Corporation System Controller Hub (SCH Poulsbo) HD Audio Controller (rev 07) cat proc/asound/cards: >>0 [MID ]: HDA-Intel - HDA Intel MID >> HDA Intel MID at 0xd01c000 irq 16 The Codec is a >>Sigmatel STAC9202 My problem is, that sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Has someone any idea to get this working? The device comes with an own (hardly) closed linux which runs a 2.6.24 kernel - here sound is working with no problems at all. If you need further info, let me know. Thanks in advance, best! Christian

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[alsa-devel] [PATCH] ASoC: MPC5200: Changed how appl_ptr is initialized
by John Bonesio 29 Jul '09

29 Jul '09

Hello, I've been working in the ALSA SOC area for the MPC5200. I've noticed that in the routine psc_dma_trigger() in sound/soc/fsl/mpc5200_dma.c there are times when the computed value for s->appl_ptr ends up wrapping negative. This occurs when s->runtime->control->appl_ptr is less than runtime->period_size * runtime->periods It seemed like this would occur at then of the data stream or when the data stream is small (smaller than runtime->period_size * runtime->periods). As far as I can tell, s->appl_ptr is supposed to be the previous value of s->runtime->control->appl_ptr, so the driver can send out the right number of bytes (from previous appl_ptr to current appl_ptr) to avoid pops and clicks at the end of the audio. So it appears the assumption in this code is wrong, that the previous s->runtime->control->appl_ptr is always rumtime->period_size * runtime->periods bytes prior to the current s->runtime->control->appl_ptr. I took out this code and instead added code to intialize s->appl_ptr to 0 in psc_dma_open(). Does this fix seem right? - John The computation for appl_ptr in the trigger() routine made a bad assumption when the stream is at the end of the file and there is not (periods * period_size) bytes remaining in the data. This causes playback issues when the total stream size is less than this amount, as the s->appl_ptr wrapps negative and the data is never played. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index b2eba27..cfe0ea4 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -181,8 +181,6 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) * end of stream and not over running it. */ s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. @@ -287,6 +285,7 @@ static int psc_dma_open(struct snd_pcm_substream *substream) } s->stream = substream; + s->appl_ptr = 0; return 0; }

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[alsa-devel] [PATCH] aedsp16: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ on line 498: static char DSPVersion[CARDVERLEN + 1] __initdata = {0, }; diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index 3ee9900..35b5912 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -325,8 +325,9 @@ /* * Size of character arrays that store name and version of sound card */ -#define CARDNAMELEN 15 /* Size of the card's name in chars */ -#define CARDVERLEN 2 /* Size of the card's version in chars */ +#define CARDNAMELEN 15 /* Size of the card's name in chars */ +#define CARDVERLEN 10 /* Size of the card's version in chars */ +#define CARDVERDIGITS 2 /* Number of digits in the version */ #if defined(CONFIG_SC6600) /* @@ -410,7 +411,7 @@ static int soft_cfg __initdata = 0; /* bitmapped config */ static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */ -static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver: +static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver: hi->ver[0] lo->ver[1] */ #if defined(CONFIG_SC6600) @@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port) * string is finished. */ ver[len++] = ret; - } while (len < CARDVERLEN); + } while (len < CARDVERDIGITS); sprintf(DSPVersion, "%d.%d", ver[0], ver[1]); DBG(("success.\n"));

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[alsa-devel] [PATCH] mpu401.c: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

mpu_synth_info[m].name is a char[30], and the minimum length of the data written by sprintf is 31 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 1b2316f..734b8f9 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name); else sprintf(mpu_synth_info[m].name, - "MPU-401 %d.%d%c Midi interface #%d", + "MPU-401 %d.%d%c MIDI #%d", (int) (devc->version & 0xf0) >> 4, devc->version & 0x0f, revision_char,

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[alsa-devel] my udev rules are breaking my dmixer setup why?
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am trying to use udev to get fixed device names depending on the location of the device on the usb bus. The udev rules seems to be working fine, however i cant get dmix to work with the udev rules using only hw dies work. If i remove the udev rules the dmixer works fine. Can somebody look at the attachment, and maybe do some testing, i would highly appreciate it. I am also on chat.freenode.org and #alsa as tuxcrafter Thanks in advance, Jelle ls -hal /dev/snd/ total 0 drwxr-xr-x 2 root root 320 2008-11-09 15:44 . drwxr-xr-x 12 root root 4.8K 2008-11-09 15:50 .. crw-rw---- 1 root user0 116, 0 2008-11-09 15:44 controlC0 crw-rw---- 1 root user1 116, 32 2008-11-09 15:44 controlC1 crw-rw---- 1 root user2 116, 64 2008-11-09 15:44 controlC2 crw-rw---- 1 root user3 116, 96 2008-11-09 15:44 controlC3 crw-rw---- 1 root user0 116, 24 2008-11-09 15:44 pcmC0D0c crw-rw---- 1 root user0 116, 16 2008-11-09 15:49 pcmC0D0p crw-rw---- 1 root user1 116, 56 2008-11-09 15:44 pcmC1D0c crw-rw---- 1 root user1 116, 48 2008-11-09 15:44 pcmC1D0p crw-rw---- 1 root user2 116, 88 2008-11-09 15:44 pcmC2D0c crw-rw---- 1 root user2 116, 80 2008-11-09 15:44 pcmC2D0p crw-rw---- 1 root user3 116, 120 2008-11-09 15:44 pcmC3D0c crw-rw---- 1 root user3 116, 112 2008-11-09 15:44 pcmC3D0p crw-rw---- 1 root audio 116, 1 2008-11-09 15:44 seq crw-rw---- 1 root audio 116, 33 2008-11-09 15:44 timer ------------------------------------------------------------------------ udevinfo info --attribute-walk --name=/dev/snd/controlC0 ------------------------------------------------------------------------ echo 'SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="controlC[0-9]*", NAME="snd/controlC4" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="controlC[0-9]", NAME="snd/controlC5" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="controlC[0-9]", NAME="snd/controlC6" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="controlC[0-9]", NAME="snd/controlC7" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC4D0p" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC5D0p" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC6D0p" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC7D0p"' | tee /etc/udev/rules.d/10-persistent-sound.rules chmod 600 /etc/udev/rules.d/10-persistent-sound.rules ------------------------------------------------------------------------ # without dmixer # this works .... echo 'pcm.softvol { type softvol slave.pcm hw:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.softvol { type softvol slave.pcm dmix:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.!default { type plug slave.pcm dmixer } pcm.dmixer { type dmix ipc_key 2048 slave.pcm hw:1 } ctl.dmixer { type hw card 1 } pcm.dsp { type plug slave.pcm dmixer } ctl.mixer { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------

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