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Re: [alsa-devel] snd-hdsp driver error?
by Raymond Yau 14 Jan '10

14 Jan '10

AFAIK , alsa-lib use the driver name to find the name of corresponding conf file from /usr/share/alsa/cards/aliases.conf In your case , H-DSP.conf does not exist and there is no aliases in /usr/share/alsa/cards/aliases.conf You have to ask Takashi or Jaroslav to fix this issue for your card open("/usr/share/alsa/cards/aliases.conf", O_RDONLY) = 3 access("/usr/share/alsa/cards/H-DSP.conf", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC1", O_RDONLY) = 3 close(3) = 0 write(2, "ALSA lib confmisc.c:1286:(snd_fu"..., 42ALSA lib confmisc.c:1286:(snd_func_refer) ) = 42 write(2, "Unable to find definition 'cards"..., 58Unable to find definition 'cards.H-DSP.pcm.front.0:CARD=0') = 58 write(2, "\n", 1 2009/11/7 Lars Zobbe <fedesensfar(a)gmail.com> > It is a clean install. > lars@lars-desktop:~$ lsb_release -a && uname -a > No LSB modules are available. > Distributor ID: Ubuntu > Description: Ubuntu 9.10 > Release: 9.10 > Codename: karmic > Linux lars-desktop 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:05:01 UTC > 2009 x86_64 GNU/Linux > > It didn't work with pulseaudio in any previos version (of ubuntu or any > other distro I've tried) either. > And the standard answer from people "out there" is "forget about pulse it > doesn't work with the HDSP" f.x. this guy > http://ubuntuforums.org/showpost.php?p=7578331&postcount=10 > And that's why I contacted pulseaudio developers first when I found out the > it was because snd_hdsp didn't load with the preferred udev but only with > module-detect (se my first post). They said that I should inform Alsa > developers as there seems to be a problem with the alsa driver. > > I've attached the strace output. > > Thanks. > Lars. > > > > > On Fri, Nov 6, 2009 at 2:53 AM, Raymond Yau <superquad.vortex2(a)gmail.com>wrote: > >> Is it a clean install of Ubuntu 9.10 or upgrade from previous Ubuntu ? >> >> you may need to find out why aplay -v -Dfront 32bit.wav fail ? >> >> strace aplay -v -Dfront 32bit.wav >> >> >> >> >> >> !!Linux Distribution >> >> !!------------------ >> >> Ubuntu 9.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.10" >> >> !!Sound Servers on this system >> !!---------------------------- >> >> Pulseaudio: >> Installed - Yes (/usr/bin/pulseaudio) >> >> >> Running - Yes >> >> ESound Daemon: >> Installed - Yes (/usr/bin/esd) >> Running - No >> >> Jack: >> Installed - Yes (/usr/bin/jackd) >> Running - No >> >> >> !!ALSA configuration files >> >> >> !!------------------------ >> >> !!User specific config file (~/.asoundrc) >> >> pcm.!default { >> type plug >> slave { pcm "jack" } >> } >> >> pcm.jack { >> type jack >> playback_ports { >> >> 0 alsa_pcm:playback_1 >> >> 1 alsa_pcm:playback_2 >> } >> capture_ports { >> 0 alsa_pcm:capture_1 >> 1 alsa_pcm:capture_2 >> } >> } >> >> ctl.mixer0 { >> type hw >> card 0 >> } >> >> >> lars@lars-desktop:~$ aplay -v -Dfront:0 32bit.wav >> ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition >> 'cards.H-DSP.pcm.front.0:CARD=0' >> ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer >> returned error: No such file or directory >> ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or >> directory >> ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM front:0 >> aplay: main:608: audio open error: No such file or directory >> >> >> Did PA developer tell you not to load oss module ? >> >> >> >> D: alsa-util.c: Checking for subset analog-mono (hw:%f) >> D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument >> I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid >> argument >> D: alsa-util.c: Trying hw:0 as last resort... >> D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: Maximum hw buffer size is 371 ms >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> D: alsa-util.c: Set neither period nor buffer size. >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy >> D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: Maximum hw buffer size is 371 ms >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> D: alsa-util.c: Set neither period nor buffer size. >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy >> I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Device or >> resource busy >> E: module.c: Failed to load module "module-alsa-source" (argument: >> "device_id=0"): initialization failed. >> W: module.c: module-oss is deprecated: Please use module-alsa-card instead >> of module-oss! >> D: oss-util.c: capabilities: MMAP REALTIME TRIGGER >> I: module-oss.c: Device opened in O_RDONLY mode. >> D: oss-util.c: Asking for 4 fragments of size 512 (requested 880) >> E: oss-util.c: SNDCTL_DSP_SETFMT: Device or resource busy >> E: module.c: Failed to load module "module-oss" (argument: >> "device=/dev/dsp"): initialization failed. >> W: module-detect.c: failed to detect any sound hardware. >> >> >> 2009/11/6 Lars Zobbe <fedesensfar(a)gmail.com> >> >> Sorry. I didn't realise that it was a bash script. Here is the link to my >>> alsa-info >>> >>> http://www.alsa-project.org/db/?f=d3728557be71067e2ed0920747118ae7efff2829 >>> >>> Thanks again. >>> Lars. >>> >>> >>> On Thu, Nov 5, 2009 at 5:29 AM, Raymond Yau <superquad.vortex2(a)gmail.com >>> > wrote: >>> >>>> Please provide the full pulseaudio log and output of alsa-info.sh >>>> >>>> http://www.pulseaudio.org/wiki/Troubleshooting >>>> >>>> pulseaudio -k; pulseaudio -vvv >>>> >>>> and >>>> >>>> aplay -v -Dfront:0 32bit.wav >>>> >>>> aplay -v -M -Dfront:0 32bit.wav >>>> >>>> aplay -v -Dplughw:0 any.wav >>>> >>>> aplay -v -M -Dplughw:0 any.wav >>>> >>>> >>>> 2009/11/4 Lars Zobbe <fedesensfar(a)gmail.com> >>>> >>>> > Hi Raymond. >>>> > >>>> > I so sorry Raymond. But I'm not sure what to do with this info. Is >>>> there >>>> > something more I have to tell you? Is there anything I can do to make >>>> > pulseaudio able to detect via udev? Can I copy-paste that bit of code >>>> you >>>> > sent me somewhere? >>>> > Please tell me if there's more you need to know. >>>> > Greetings. >>>> > Lars. >>>> > >>>> > >>>> > >>>> > >>>> > On Tue, Nov 3, 2009 at 7:48 AM, Raymond Yau < >>>> superquad.vortex2(a)gmail.com>wrote: >>>> > >>>> >> I guess route plugin used by RME96*.conf does not support mmap when >>>> there >>>> >> is >>>> >> channel conversion since minimum channels supported by hdsp is 14 >>>> >> >>>> >> >>>> >> static struct snd_pcm_hardware snd_hdsp_playback_subinfo = >>>> >> { >>>> >> .info = (SNDRV_PCM_INFO_MMAP | >>>> >> SNDRV_PCM_INFO_MMAP_VALID | >>>> >> SNDRV_PCM_INFO_NONINTERLEAVED | >>>> >> SNDRV_PCM_INFO_SYNC_START | >>>> >> SNDRV_PCM_INFO_DOUBLE), >>>> >> #ifdef SNDRV_BIG_ENDIAN >>>> >> .formats = SNDRV_PCM_FMTBIT_S32_BE, >>>> >> #else >>>> >> .formats = SNDRV_PCM_FMTBIT_S32_LE, >>>> >> #endif >>>> >> .rates = (SNDRV_PCM_RATE_32000 | >>>> >> SNDRV_PCM_RATE_44100 | >>>> >> SNDRV_PCM_RATE_48000 | >>>> >> SNDRV_PCM_RATE_64000 | >>>> >> SNDRV_PCM_RATE_88200 | >>>> >> SNDRV_PCM_RATE_96000), >>>> >> .rate_min = 32000, >>>> >> .rate_max = 96000, >>>> >> .channels_min = 14, >>>> >> .channels_max = HDSP_MAX_CHANNELS, >>>> >> .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, >>>> >> .period_bytes_min = (64 * 4) * 10, >>>> >> .period_bytes_max = (8192 * 4) * HDSP_MAX_CHANNELS, >>>> >> .periods_min = 2, >>>> >> .periods_max = 2, >>>> >> .fifo_size = 0 >>>> >> }; >>>> >> >>>> >> 2009/11/2 Lars Zobbe <fedesensfar(a)gmail.com> >>>> >> >>>> >> > Hi. >>>> >> > >>>> >> > Talking to one of the pulseaudio developers (*Colin Guthrie*) about >>>> a >>>> >> > problem with pulseaudio loading modules and my sound card (RME HDSP >>>> >> 9632), >>>> >> > I >>>> >> > was adviced to inform you about the problem. >>>> >> > >>>> >> > I'm on ubuntu 9.10, 64bit, kernel 2.6.31 with an amd64 x2 3800+ >>>> >> processor. >>>> >> > >>>> >> > >>>> >> > Symptoms are: >>>> >> > Can only make pulseaudio load modules with the deprecated >>>> load-module >>>> >> > module-detect command, module-udev-detect doesn't detect anything >>>> (no >>>> >> sound >>>> >> > - only null sink output in padevchooser). >>>> >> > I get terrible stuttering. >>>> >> > I can remove pulseaudio and make things work, more or less. >>>> However, >>>> >> that >>>> >> > leaves me without software mixing and apps fighting for the rights >>>> to >>>> >> make >>>> >> > sound. And more inconveniences (fx. no volume-control). >>>> >> > I can play sound from audacity without stuttering by choosing the >>>> >> cardname >>>> >> > directly in output HDSP, not ALSA, not pulseaudio. >>>> >> > >>>> >> > >>>> >> > When trying to run alsa-info.sh I get a lot of "not found" errors. >>>> So >>>> >> > you'll >>>> >> > get some manual info (they might have moved som files around in >>>> ubuntu >>>> >> > 9.10). >>>> >> > >>>> >> > This is part of my syslog during a boot. (I did the formatting to >>>> bold). >>>> >> > >>>> >> > Nov 2 10:57:32 lars-desktop pulseaudio[1524]: module.c: >>>> module-detect >>>> >> is >>>> >> > deprecated: Please use module-udev-detect instead of module-detect! >>>> >> > Nov 2 10:57:37 lars-desktop pulseaudio[1524]: ratelimit.c: 27 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 10:57:39 lars-desktop pulseaudio[1612]: module.c: >>>> module-detect >>>> >> is >>>> >> > deprecated: Please use module-udev-detect instead of module-detect! >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: ratelimit.c: 23 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: ALSA >>>> woke us >>>> >> up >>>> >> > to write new data to the device, but there was actually nothing to >>>> >> write! >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: *Most >>>> likely >>>> >> > this is a bug in the ALSA driver 'snd_hdsp'. Please report this >>>> issue to >>>> >> > the >>>> >> > ALSA developers.* >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: We were >>>> >> woken >>>> >> > up >>>> >> > with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 >>>> or >>>> >> > another value < min_avail. >>>> >> > Nov 2 10:57:49 lars-desktop pulseaudio[1612]: ratelimit.c: 99 >>>> events >>>> >> > suppressed >>>> >> > >>>> >> > Later if I try to play audio I get terrible skipping or stuttering >>>> and >>>> >> my >>>> >> > syslog (in less than a minute) writes: >>>> >> > Nov 2 11:55:16 lars-desktop pulseaudio[1612]: ratelimit.c: 16 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:21 lars-desktop pulseaudio[1612]: ratelimit.c: 34 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:26 lars-desktop pulseaudio[1612]: ratelimit.c: 70 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:31 lars-desktop pulseaudio[1612]: ratelimit.c: 29 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:36 lars-desktop pulseaudio[1612]: ratelimit.c: 43 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:41 lars-desktop pulseaudio[1612]: ratelimit.c: 161 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:46 lars-desktop pulseaudio[1612]: ratelimit.c: 28 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:51 lars-desktop pulseaudio[1612]: ratelimit.c: 18 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:56 lars-desktop pulseaudio[1612]: ratelimit.c: 16 >>>> events >>>> >> > suppressed >>>> >> > >>>> >> > Here's the relevant part (concerning my sound card) of lspci -vvv >>>> >> > >>>> >> > 02:09.0 Multimedia audio controller: Xilinx Corporation RME >>>> Hammerfall >>>> >> DSP >>>> >> > (rev 9a) >>>> >> > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- >>>> ParErr- >>>> >> > Stepping- SERR- FastB2B- DisINTx- >>>> >> > Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- >>>> >> > <TAbort- <MAbort- >SERR- <PERR- INTx- >>>> >> > Latency: 255 >>>> >> > Interrupt: pin A routed to IRQ 19 >>>> >> > Region 0: Memory at fdff0000 (32-bit, non-prefetchable) >>>> [size=64K] >>>> >> > Kernel driver in use: RME Hammerfall DSP >>>> >> > Kernel modules: snd-hdsp >>>> >> > >>>> >> > And: >>>> >> > lars@lars-desktop:~$ aplay -l >>>> >> > **** List of PLAYBACK Hardware Devices **** >>>> >> > card 0: DSP [Hammerfall DSP], device 0: RME Hammerfall HDSP 9632 >>>> [RME >>>> >> > Hammerfall HDSP 9632] >>>> >> > Subdevices: 1/1 >>>> >> > Subdevice #0: subdevice #0 >>>> >> > >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/version >>>> >> > Advanced Linux Sound Architecture Driver Version 1.0.21. >>>> >> > Compiled on Oct 15 2009 for kernel 2.6.31-14-generic (SMP). >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/cards >>>> >> > 0 [DSP ]: H-DSP - Hammerfall DSP >>>> >> > RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 >>>> >> > >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/modules >>>> >> > 0 snd_hdsp >>>> >> > >>>> >> > I don't have .asound.conf files. >>>> >> > >>>> >> > lars@lars-desktop:~$ hdsploader >>>> >> > hdsploader - firmware loader for RME Hammerfall DSP cards >>>> >> > Looking for HDSP + Multiface or Digiface cards : >>>> >> > Card 0 : RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 >>>> >> > >>>> >> > And of course. My computer makes no sound if I don't load the >>>> hdspmixer >>>> >> at >>>> >> > startup. I guess that's just normal for this card though. >>>> >> > >>>> >> > Please let me know, if you need more info. I really hope you can >>>> help >>>> >> out >>>> >> > here. And that this might help you to. >>>> >> > >>>> >> > Thanks for the great work you do. Greetings. >>>> >> > Lars Zobbe. >>>> >> > _______________________________________________ >>>> >> > Alsa-devel mailing list >>>> >> > Alsa-devel(a)alsa-project.org >>>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >> > >>>> >> _______________________________________________ >>>> >> Alsa-devel mailing list >>>> >> Alsa-devel(a)alsa-project.org >>>> >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >> >>>> > >>>> > >>>> _______________________________________________ >>>> Alsa-devel mailing list >>>> Alsa-devel(a)alsa-project.org >>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>> >>> >> >

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix typo in comment in header file
by Thomas Weber 14 Jan '10

14 Jan '10

Signed-off-by: Thomas Weber <weber(a)corscience.de> --- sound/soc/codecs/twl4030.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index dd6396e..f206d24 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -25,7 +25,7 @@ /* Register descriptions are here */ #include <linux/mfd/twl4030-codec.h> -/* Sgadow register used by the audio driver */ +/* Shadow register used by the audio driver */ #define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -- 1.6.4.4

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Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 14 Jan '10

14 Jan '10

>mmap is supported by the Samsung drivers. Thanks for this information which i overlooked. >This looks like the dmix data rates are getting out of sync with what >the hardware is doing. Enabling CONFIG_SND_PCM_XRUN_DEBUG will produce >more diagnostic information from the driver side I got the problem and after doing the changes DMIX plugin is working very fine. As you know XRUN was happening and this happening in snd_pcm_update_hw_ptr_post function. This happened because stop_threshold is increased suddenly(Is it done by dmix? If yes then why ?) ?????? In normal condtions this would have been fine but in my code i was multiplying it by some amount causing it to wrap around(stop_threshold=0 );Causing XRUN.So i just removed this multiplying factor. One more Rate conversion query: ./alsa_aplay -v -D"plug:'dmix:RATE=44100'" bond.wav Playing WAVE 'bond.wav' : Unsigned 8 bit, Rate 11025 Hz, Mono Plug PCM: Rate conversion PCM (44100, sformat=U8) As you can see bond.wav is sampled at 11025k but when i am playing with Plug PCM there is some noise in the song why????Is rate conversion algoritm not able to resample it properly? If i play with ./alsa_aplay -v -D"plug:'dmix:RATE=11025'" bond.wav then no noise.Probably because no resampling happening? Mark Brown Heartfelt Thanks for your time.

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Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 14 Jan '10

14 Jan '10

2 1

[alsa-devel] Linux kernel 2.6.32.3 bug report
by d binderman 14 Jan '10

14 Jan '10

Hello there, I just ran the sourceforge tool cppcheck over the source code of the Linux kernel 2.6.32.3 It said [./sound/soc/s3c24xx/s3c24xx-pcm.c:128]: (error) Possible null pointer dereference: substream - otherwise it is redundant to check if substream is null at line 130 I checked the source code and I agree with the compiler. Suggest code rework. Regards David Binderman _________________________________________________________________ Do you have a story that started on Hotmail? Tell us now http://clk.atdmt.com/UKM/go/195013117/direct/01/

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[alsa-devel] capturing data from the microphone
by Riccardo Magliocchetti 14 Jan '10

14 Jan '10

[the same message is waiting in moderator queue, sorry if you receive it two times] Hello, i'm trying to capture audio data from the microphone, but i'm not able to read any frames. The frames are always 0 and delay is fixed to 2730. I've tried calling snd_pcm_avail_delay, snd_pcm_avail_update but does not make a difference. The soundcard is driven by snd_hda_intel. The code is more or less the same as Paul Davis' tutorial on using the alsa api. This is the code for init and the loop that tries to read the data: #define MIC_BUFSIZE 4096 static gpointer snd_pcm_read(gpointer data) { int error; snd_pcm_sframes_t frames = MIC_BUFSIZE; while (TRUE) { if ((error = snd_pcm_wait (pcm_handle, 1000)) < 0) { g_printerr("Failed to poll: %s\n", snd_strerror(error)); continue; } if ((error = snd_pcm_avail(pcm_handle)) < 0) { if (error == -EPIPE) { g_printerr("xrun! %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } else { g_printerr("alsa_pcm_avail_update error %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } } frames = error; if (frames == 0) continue; frames = frames > MIC_BUFSIZE ? MIC_BUFSIZE : frames; g_printerr ("frames: %ld\n", frames); g_static_mutex_lock(&mutex); error = snd_pcm_readi(pcm_handle, Mic_Buffer[Mic_WriteBuf], frames); g_static_mutex_unlock(&mutex); if (error < 0) error = snd_pcm_recover(pcm_handle, error, 0); if (error < 0) { LOG("snd_pcm_readi FAIL!: %s\n", snd_strerror(error)); } } return GINT_TO_POINTER(TRUE); } BOOL Mic_Init() { snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int err; if (Mic_Inited) return TRUE; // Open the default sound card in capture if ((err = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_CAPTURE, /*SND_PCM_NONBLOCK*/ 0)) < 0) { g_printerr("Failed to open device: %s\n", snd_strerror(err)); return FALSE; } // Allocate the snd_pcm_hw_params_t structure and fill it. snd_pcm_hw_params_alloca(&hwparams); if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { g_printerr("Failed to setup hw parameters: %s\n", snd_strerror(err)); return FALSE; } //Set the access if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { g_printerr("Failed to set access: %s\n", snd_strerror(err)); return FALSE; } //dir 0 == exacte (Rate = 16K exacte) if ((err = snd_pcm_hw_params_set_rate(pcm_handle, hwparams, 16000, 0)) < 0) { g_printerr("Failed to set rate: %s\n", snd_strerror(err)); return FALSE; } /* Set sample format */ if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0) { g_printerr("Failed to set format: %s\n", snd_strerror(err)); return FALSE; } // Set one channel (mono) if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1)) < 0) { g_printerr("Failed to set channels: %s\n", snd_strerror(err)); return FALSE; } //Set the params if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { g_printerr("Failed to set hw parameters: %s\n", snd_strerror(err)); return FALSE; } snd_pcm_sw_params_alloca(&swparams); if ((err = snd_pcm_sw_params_current (pcm_handle, swparams)) < 0) { g_printerr("Failed to set current sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_avail_min (pcm_handle, swparams, MIC_BUFSIZE)) < 0) { g_printerr("Failed to set minimum available count: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_start_threshold (pcm_handle, swparams, 0U)) < 0) { g_printerr("Failed to set start mode: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) { g_printerr("Failed to set sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_prepare (pcm_handle)) < 0) { g_printerr("Failed to prepare audio interface to use: %s\n", snd_strerror(err)); return FALSE; } Mic_Inited = TRUE; Mic_Reset(); mic_reader = g_thread_create(snd_pcm_read, NULL, TRUE, NULL); return TRUE; } thanks, Riccardo

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[alsa-devel] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write!
by Konrad Schlichtherle 14 Jan '10

14 Jan '10

Hello, sometimes it occure that my speakers do not play sound although the speakers should play music with VLC. Then i took a look at /var/log/messages and found this: Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. The output of utils_alsa-info.sh is there: http://www.alsa-project.org/db/?f=1767e9bc745724a67c908549d20c137375d0605e I didn't know wether this is a bug or a misconfiguration of my openSUSE 11.2 Konrad

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Re: [alsa-devel] [PATCH] OMAP: Exporting functions doing common register access
by Aggarwal, Anuj 13 Jan '10

13 Jan '10

> -----Original Message----- > From: Paul Walmsley [mailto:paul@pwsan.com] > Sent: Tuesday, November 17, 2009 2:46 AM > To: Aggarwal, Anuj > Cc: linux-omap(a)vger.kernel.org > Subject: Re: [PATCH] OMAP: Exporting functions doing common register > access > > Hello Anuj, > > On Mon, 16 Nov 2009, Anuj Aggarwal wrote: > > > These functions need to be exported so that drivers (e.g. McBSP) can > > be configured as modules. > > > > McBSP driver gets built as a module when ASoC driver for OMAP3 EVM > > is configured as module. McBSP driver uses functions like > > omap_ctrl_readl/omap_ctrl_writel, which are defined in control.c > > file but not exported. Without that, McBSP driver fails to build as > > a module. > > > > Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> > > This will prevent the McBSP driver from being usable across chips. For > example, if a future OMAP comes out with McBSP clock source switching > implemented in a different way, or if a DaVinci chip comes out with a > McBSP block but no SCM block, this either won't build or won't run. > > A better approach would be to move the OMAP2/3-specific functions from the > McBSP driver into the OMAP McBSP integration code, > arch/arm/*omap*/mcbsp.c, and to define function pointers in > struct platform_data for these functions, and to pass them from the > integration code into the device driver code. That will make these > drivers independent of chip integration details. Sorry to confuse you, I was talking about the platform specific ASoC-McBSP code for OMAP (file - sound/soc/omap/omap-mcbsp.c) platform. The McBSP driver - both platform and machine specific - doesn't use the functions exported by the patch. Including alsa-devel list too... > > > - Paul

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[alsa-devel] Newbie question: Sending untimed MIDI data to ALSA seq
by Josh Lehan 13 Jan '10

13 Jan '10

Hello. Newbie question: I'm wondering about sending untimed MIDI data to the ALSA sequencer subsystem. The application would be for a "beep"-like program, from the command line, that instead of making beeps, would play MIDI instruments to make better-sounding beeps. I have TiMidity hooked up into ALSA, as a softsynth, and it works just fine for playing MIDI data sent to it by ALSA. The "aplaymidi" command works for .MID files, and the "vkeybd" program works for simulating a keyboard and sending in untimed MIDI data. The "aseqview" program also works great for watching the notes being played, as they go by. So, my MIDI software stack appears to be in order. I don't have a hardware synthesizer hooked up at the moment (my Kawai K1 is showing its age, and a few keys are broken). I also looked at "amidi", which features a -S option to send in untimed MIDI bytes, but unfortunately, that addresses an entirely different namespace. Evidently, it only speaks to "RawMIDI" devices, of which TiMidity is not. So, I can't generate untimed MIDI data on my own. I was thinking of writing a small "midibeep" program that would do this, similar to what vkeybd does, but without requiring any user interaction. Does such a program already exist? Thanks! Josh

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[alsa-devel] Mic Boost control problem with snd-hda-intel codec AL662 rev 1
by Anisse Astier 13 Jan '10

13 Jan '10

Hi, The Mic Boost control on a laptop seems to behave incorrectly. The control has the flag "volume" instead of "cvolume", causing it to appear in both the "Playback" and "Capture" tabs of alsamixer. This wouldn't be a problem if it would just work in the Capture tab: it doesn't. When I modify it in Capture, it doesn't retain its value: it's gone after i close alsamixer. But if i modify it in "Playback" it will retain its value and the value will be mirrored by the Mic Boost in Capture (both Mic Boost seem to reflect the same control, except one is RO, the other is RW). It's a minor a annoyance, but I'd like to fix it and have "Mic Boost" appear only as a Capture control and retaining its value. Maybe I should create a new mixer configuration for this hardware in patch_realtek.c, but I don't know where to start. Bios autoprobing works for everything but just what I mentioned. My alsa-info.sh : http://www.alsa-project.org/db/?f=fc17b509e532fc366eb2d5fe5ec9e6730c96c837 Any ideas? Where should I start? Regards, Anisse

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