June 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH v2] alsa-plugins/pulse: Implement 'pause'.
by Lennart Poettering 18 Jun '09

18 Jun '09

On Thu, 18.06.09 14:55, Troy Moure (twmoure(a)szypr.net) wrote: > > pa_stream_cork() will return a pa_operation object on success. The > > least you need to do is call pa_operation_unref() on it, which you can > > safely do right-away. Otherwise this will leak memory. > > > > Lennart > > Thanks, Lennart, for looking at the first verson I sent. I've added a > call to pa_operation_unref() as you suggested. Looks good to me now. > > pulse/pcm_pulse.c | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c > index db8d1e1..c276839 100644 > --- a/pulse/pcm_pulse.c > +++ b/pulse/pcm_pulse.c > @@ -739,6 +739,30 @@ static int pulse_close(snd_pcm_ioplug_t * io) > return 0; > } > > +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) > +{ > + snd_pcm_pulse_t *pcm = io->private_data; > + int err = 0; > + > + assert (pcm); > + assert (pcm->p); > + > + pa_threaded_mainloop_lock(pcm->p->mainloop); > + > + if (pcm->stream) { > + pa_operation *o; > + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); > + if (o) > + pa_operation_unref(o); > + else > + err = -EIO; > + } > + > + pa_threaded_mainloop_unlock(pcm->p->mainloop); > + > + return err; > +} > + > static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .start = pulse_start, > .stop = pulse_stop, > @@ -750,6 +774,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .prepare = pulse_prepare, > .hw_params = pulse_hw_params, > .close = pulse_close, > + .pause = pulse_pause > }; > > Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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Re: [alsa-devel] Regarding internal mic on Aspire 6935G
by Karthik Ramgopal 18 Jun '09

18 Jun '09

Hi, How to check if it works in stereo or mono mode? I remember reading hector's entry on his blog that on account of some wierd sum/difference problem he had duplicated the + signal across both outputs so it was actually mono? Regards, Karthik

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[alsa-devel] preventing SNDRV_PCM_STATE_XRUN
by Sergey Lapin 18 Jun '09

18 Jun '09

Hi, all! I have uncontrollable data stream which I need to play through kernel via ALSA snd_pcm_lib_write call. Everything is fine, but once a few seconds I get SNDRV_PCM_STATE_XRUN on stream and thus, -EPIPE. Sound glitches are minor, but annoying. Is it possible to somehow prevent this from happenning? Is it possible to distinguish if my next chunk of data will fit? Thanks a lot, S.

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[alsa-devel] Regarding internal mic on Aspire 6935G
by Karthik Ramgopal 18 Jun '09

18 Jun '09

Hi, Sorry for the confusion. My internal mic does work, but wierdly not in Gnome Sound recorder. Skype detects it just fine and I am able to talk without making any changes to the code. So, I guess 8930G and 6935G are same after all. Thanks and Regards, Karthik PS: Please also add the PCI ID's of 6920G (0x1025 0146) and 8920G (0x1025 0145) under the same quirk. They also use ALC889 chipset with the same pinconfig.

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Re: [alsa-devel] trace PCM data from usb bus to alsa lib
by Takashi Iwai 18 Jun '09

18 Jun '09

At Wed, 17 Jun 2009 16:10:26 +0800, Zero Luo wrote: > > Dear Takashi, > > Data from usb dongle is 16K*16 bit PCM data. And what is the format and channels you are using in alsa-lib? alsa-lib itself can convert formats and channels automatically if needed. Takashi > 2009/6/17, Takashi Iwai <tiwai(a)suse.de>: > > At Wed, 17 Jun 2009 07:25:25 +0000 (UTC), > ZeroCluo wrote: > > > > Hi > > > > I am trying to trace the Audio data from usb bus. > > in kernel: > > dump data in retire_capture_urb() ,found data format is like: > > AABB 0000 CCDD 0000 ; > > > > but in alsa library , > > dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", > > > > Do you know when and where those "0000" is cut? > > Are you using the native format that the hardware runs? > > Takashi > >

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Re: [alsa-devel] snd-ctxfi oops
by The Source 18 Jun '09

18 Jun '09

On 15.06.2009 17:01, Takashi Iwai wrote: > At Mon, 15 Jun 2009 15:00:03 +0200, > I wrote: > >> At Mon, 15 Jun 2009 16:59:53 +0400, >> The Source wrote: >> >>> On 15.06.2009 16:56, Takashi Iwai wrote: >>> >>>> At Sat, 13 Jun 2009 10:07:49 +0200, >>>> I wrote: >>>> >>>> >>>>> >>>>> >>>>>> Tested the patch more. Really no oopses. Instead of them I get rare >>>>>> system lockups. >>>>>> >>>>>> >>>>> In which situation, exactly? >>>>> Could you check whether it happens with use_system_timer=1 module >>>>> option, too? >>>>> >>>>> >>>> Or try the very latest alsa-driver-snapshot (not daily tarball). >>>> I fixed a deadlock issue in cttimer.c now. >>>> >>>> >>>> thanks, >>>> >>>> Takashi >>>> >>>> >>>> >>> Should I apply that patch to this snapshot? >>> >> Which patch do you mean? >> Just grab the latest alsa-driver-snapshot.tar.gz from >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ >> > Also, check alsa-driver/alsa-kernel/HEAD whether the commit > 8dca419721d188bfee5f19fad45275856c619a5c > ALSA: ctxfi - Fix deadlock with xfi-timer > is included. > > > Takashi > > Tested the driver and it's fine so far. No craches with any application including Doom 3, which caused system lockups often with earlier versions of the driver.

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[alsa-devel] [PATCH] alsa-plugins/pulse: Implement 'pause'.
by Troy Moure 17 Jun '09

17 Jun '09

Just cork or uncork the stream to pause or unpause it. Signed-off-by: Troy Moure <twmoure(a)szypr.net> --- I'm not very familiar with alsa or pulseaudio, so this might not be right. If it is, feel free to apply it, though. It *does* fix my problem, which was that pause and unpause didn't work in 'cmus' after I upgraded to Fedora 11. Since the pulse plug-in didn't support snd_pcm_pause, pausing in cmus would just cause cmus to stop writing to the buffer, leading to buffer underruns in pulseaudio. With this patch, pausing in cmus works perfectly for me. If this should be fixed another way, please let me know what it is! Thanks. pulse/pcm_pulse.c | 20 ++++++++++++++++++++ 1 files changed, 20 insertions(+), 0 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index db8d1e1..c322884 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -739,6 +739,25 @@ static int pulse_close(snd_pcm_ioplug_t * io) return 0; } +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) +{ + snd_pcm_pulse_t *pcm = io->private_data; + int err = 0; + + assert (pcm); + assert (pcm->p); + + pa_threaded_mainloop_lock(pcm->p->mainloop); + + if (pcm->stream) + if (!pa_stream_cork(pcm->stream, enable, NULL, NULL)) + err = -EIO; + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + + return err; +} + static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .start = pulse_start, .stop = pulse_stop, @@ -750,6 +769,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .prepare = pulse_prepare, .hw_params = pulse_hw_params, .close = pulse_close, + .pause = pulse_pause }; -- 1.6.2.2

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[alsa-devel] ICE1712 softdevices fixes
by Stefan Kost 17 Jun '09

17 Jun '09

hi, > The following patches fixes the default device as they appear for > ICE1712-based cards. Without this cards, 'aplay -L' will show a series of > devices like surround40 and surround71 that are not meaningful for this card, > and a front device that would not work. > > Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound > devices on ICE1712. Make the front:idx device work for ICE1712. thanks for making those. this was exactly what I needed to get pulseaudio working with my card again. I also agree that even though that my card (Hoontech DSP 24) has 8/10 physical in/outs, those are essentially mono channels. The surround mapping seem to be more harm that use to me. If a musician indeed wires 6 3.5 jack outs somehow to his 5.1 amp, then its probably okay to uncomment such mappings. So what about making diegos patches the default? Stefan

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[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: fix legacy input streaming
by Daniel Mack 17 Jun '09

17 Jun '09

Seems that nobody recently tried the input on the very first supported sound card model, RK2. This patch fixes the byte offset to make it running again. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 5 +++-- sound/usb/caiaq/device.c | 2 +- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b144513..8f9b60c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; } else { - dev->period_in_count[index] = BYTES_PER_SAMPLE; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; + int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; + dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; + dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; } if (dev->streaming) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 2240624..0e5db71 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.6.3.1

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[alsa-devel] sound device is buzy to run pcm.c
by Guilherme Raymo Longo 17 Jun '09

17 Jun '09

I am getting this problem when running the pcm.c demo program at http://alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html This is the execution of the program: guilherme@lap:/home/guilherme/Experimentos# ./PCM Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_LE, 1 channels sine wave rate is 0 0.0000Hz Using transfer method: write Playback open error: Device or resource busy I tried to check if the device is really buzy: root@lap:/home/guilherme/Experimentos# lsof |grep snd lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system */ home/guilherme/*.gvfs Output information may be incomplete. esd 5433 guilherme mem CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 5r CHR 116,33 6787 /dev/snd/timer esd 5433 guilherme 6w CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 7r CHR 116,0 6987 /dev/snd/controlC0 and it seems to be ok. That's my card root@lap:/home/guilherme/Experimentos# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Is there a way to free the device or smth is wrong with my code?? I am using slackware 12.2 kernel 2.6 Great Regads;

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