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[alsa-devel] 1.0.15: volume levels drift on HDA with STAC92XX codec
by Chuck Ebbert 26 Nov '07

26 Nov '07

We have two reports now of unstable volume levels: https://bugzilla.redhat.com/show_bug.cgi?id=361051 https://bugzilla.redhat.com/show_bug.cgi?id=354981 This is with kernel 2.6.23 plus the two ALSA merge patches from 2.6.23-rc1.

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[alsa-devel] [PATCH] add number of periods constraint to snd-aoa [try 3]
by Heikki O Lindholm 26 Nov '07

26 Nov '07

From: Heikki Lindholm <holindho(a)cs.helsinki.fi> The aoa driver is not specifying constraints on number of periods, and, it seems, it might end with a non-integer number, which it cannot deal with. Fix by adding a proper constraint. Signed-off-by: Heikki Lindholm <holindho(a)cs.helsinki.fi> --- try 3: fixed coding style diff -r 1b54a8725ded aoa/soundbus/i2sbus/i2sbus-pcm.c --- a/aoa/soundbus/i2sbus/i2sbus-pcm.c Wed Nov 14 17:07:17 2007 +0100 +++ b/aoa/soundbus/i2sbus/i2sbus-pcm.c Fri Nov 23 14:16:07 2007 +0200 @@ -194,6 +194,12 @@ static int i2sbus_pcm_open(struct i2sbus hw->period_bytes_max = 16384; hw->periods_min = 3; hw->periods_max = MAX_DBDMA_COMMANDS; + err = snd_pcm_hw_constraint_integer(pi->substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + result = err; + goto out_unlock; + } list_for_each_entry(cii, &sdev->codec_list, list) { if (cii->codec->open) { err = cii->codec->open(cii, pi->substream);

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[alsa-devel] ALSA Leaking memory
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! Apparently alsa-libs .15 is leaking memory when using snd_pcm_open() on some devices that cannot be opened. On my USB wecam which has only a single mono audio stream, opening it as "front:3" (which of course fails) leaks a bit of memory. Consider this example: <snip> #include <asoundlib.h> #include <assert.h> int main(int argc, char *argv[]) { snd_pcm_t *pcm; if (snd_pcm_open(&pcm, "front:3", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK) == 0) { int r = snd_pcm_close(pcm); assert(r == 0); } snd_config_update_free_global(); return 0; } </snip> And there they are gone, the 10 bytes of memory. Here's a snippet from valgrinding this program: ==22198== 10 bytes in 1 blocks are definitely lost in loss record 1 of 3 ==22198== at 0x4023765: malloc (vg_replace_malloc.c:149) ==22198== by 0x417C44F: strdup (strdup.c:43) ==22198== by 0x4071954: snd_determine_driver (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406E80C: snd_config_hook_load_for_all_cards (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406EC7C: (within /usr/lib/libasound.so.2.0.0) ==22198== by 0x406F896: snd_config_searcha_hooks (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406FA74: snd_config_search_alias_hooks (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406FCD9: snd_config_search_definition (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x40952FE: (within /usr/lib/libasound.so.2.0.0) ==22198== by 0x80484D7: main (in /home/lennart/alsa-test) And there seem to be more. When PulseAudio starts up it probes for available audio devices. On my machine there are 5 sound cards, which are all probed with device strings like "surround51:0" and so on, and in the end some memory with a caller history like the one shown above is shown, and another one: ==22229== 235 (180 direct, 55 indirect) bytes in 5 blocks are definitely lost in loss record 6 of 6 ==22229== at 0x4023AA4: calloc (vg_replace_malloc.c:279) ==22229== by 0x46967B8: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x4696C48: snd_config_make (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x4696CD0: snd_config_imake_pointer (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46EA573: _snd_pcm_hook_ctl_elems_install (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46EAC10: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46EB2BC: _snd_pcm_hooks_open (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C0CD4: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C1467: snd_pcm_open_slave (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46D232B: _snd_pcm_route_open (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C0CD4: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C1467: snd_pcm_open_slave (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46E47CC: _snd_pcm_empty_open (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C0CD4: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C13D2: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x412E007: pa_alsa_open_by_device_id (alsa-util.c:529) ==22229== by 0x4135904: module_alsa_sink_LTX_pa__init (module-alsa-sink.c:781) ==22229== by 0x419CD2B: pa_module_load (module.c:115) ==22229== by 0x4164D33: hal_device_load_alsa (module-hal-detect.c:204) ==22229== by 0x4165313: hal_device_add (module-hal-detect.c:323) 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] snd_pcm_name() returns NULL is used on a surround51: device
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! snd_pcm_name() returns NULL if used shortly after opening a "surround51" device. ("surround51" is just an example, it returns NULL for a lot of other devices, too.) Also reported as bug in the BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3563 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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Re: [alsa-devel] How to cont the sound cards?
by Takashi Iwai 26 Nov '07

26 Nov '07

At Mon, 26 Nov 2007 12:53:52 +0200, Cosmin Humeniuc wrote: > > Thank you for the reply, Takashi. > > The problem is, I'm not sure what should I loop through. I saw this > list on the wiki ( > http://www.alsa-project.org/main/index.php/DeviceNames ) but I don't > know how it was obtained. The code would be like: int card = -1; while (!snd_card_next(&card) && card >= 0) { ... } Takashi > > Cosmin > > On 11/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 19 Nov 2007 09:12:51 +0200, > > Cosmin Humeniuc wrote: > > > > > > Hi, > > > > > > I am trying to develop an application using the ALSA library. I read > > > in a book about the snd_cards function, but it doesn't seem to be > > > defined anywhere. (The book is rather old, though, it talks about the > > > 0.5.x series). > > > > > > The compiler doesn't complain, but the linker gives me an "undefined > > > reference to `snd_cards'" error. I assume this function is deprecated, > > > but what can I use instead? > > > > Oh that myth. I myself already forgot about that function :) > > > > On ALSA 1.x, there is no corresponding function. You'll need to > > loop and check the presence of each card. > > > > > > Takashi > > >

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[alsa-devel] [PATCH] Use different buffer metrics in the PulseAudio plugin
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! Please commit this patch to alsa-plugins we ship in Fedora: http://cvs.fedoraproject.org/viewcvs/*checkout*/devel/alsa-plugins/1.0.14-b… It increases the "pre-buffering level" (i.e. start threshold) to the full buffer size minus one period. This makes PA work a little bit more like normal audio devices, and makes a few drop outs go away for software which uses very small period sizes. It also increases the initial maximum buffer size, which allows a small overcommit. That's not really an issue, but cleaner nonetheless so I smuggled it into this patch. Also reported in the ALSA BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3578 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] If /dev/snd/controlCx EACCES vs. ENODEV mixup
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! If /dev/snd/controlC0 is not accessible due to a permission problem (EACCES), then alsa-libs will lie and return ENODEV. This is quite a bit confusing: <snip> $ sudo chmod 000 /dev/snd/controlC0 $ aplay -D hw:0 ALSA lib pcm_hw.c:1207:(_snd_pcm_hw_open) Invalid value for card aplay: main:546: audio open error: No such device </snip> (and strace shows that the actual problem is EACESS, as one would assume. I also filed this as bug on the BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3600 BTW, what's the status of the BTS? Takashi asked me to post all my issues on the ML instead of on the BTS. So why have the BTS at all? It's even linked on the alsa-project.org front page under "I found a bug!", which is a bit misleading. If this Mantis thing is not liked at all, would it be possible to make some replacement available? Maybe just use the kernel bugzilla on bugzilla.kernel.org? It's very fast, and certainly more fun to work with than with Mantis. 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] add support for Kore controller 2
by Daniel Mack 26 Nov '07

26 Nov '07

Added support for Native Instrument's Kore controller 2. This device has no audio but MIDI, input devices and ALSA controllers only. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] [PATCH] soc/wm8731: Fix stereo mixer controls
by Ville Syrjälä 26 Nov '07

26 Nov '07

soc/wm8731: Fix stereo mixer controls Disable the simultaneous load feature for the line in and headphone out volume registers. This allows left and right volume levels to be controlled separately. Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7ca0b52..57fb95a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -562,13 +562,13 @@ static int wm8731_init(struct snd_soc_device *socdev) /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100); + wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100); + wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg | 0x0100); + wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg | 0x0100); + wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); wm8731_add_controls(codec); wm8731_add_widgets(codec); -- Ville Syrjälä syrjala(a)sci.fi http://www.sci.fi/~syrjala/

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[alsa-devel] Audigy LS driver
by Daniel Sabo 26 Nov '07

26 Nov '07

Hi, I'm interested in helping with the ca0106 driver. My understanding is that the dmix and multi plugins can't be used with each other, which means that right now it not possible to have a dmixed surround sound device with this card. The driver has 4 stereo devices on it (one for each DSP on the board?), would it be possible for me to merge devices 0,1, and 2 into a single 6 channel device? I have some experience with driver development doing ARM SoC stuff, but this is my first time messing with a linux driver. Thanks, - Daniel Sabo

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