June 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 08 Jan '14

08 Jan '14

At Thu, 6 Sep 2012 15:17:58 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 15:08 +0200, Takashi Iwai wrote: > > At Thu, 6 Sep 2012 11:43:48 +0200, > > Markus Trippelsdorf wrote: > > > > > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > > > At Thu, 06 Sep 2012 09:35:26 +0200, > > > > Takashi Iwai wrote: > > > > > > > > In short, a patch like below may fix the issue (note: completely > > > > untested!) > > > > > > No it doesn't, unfortunately... > > > > OK, I start tracking down the problem a bit more deeply now. > > > > The issue happens when the first two URBs are passed to > > retire_playback_urb(). These are URBs filled before start_endpoints() > > are set, so they contain actually zero size. Even though these are > > a sort of dummy packets, the driver still tries to check with the > > queued delay account, and gives bogus errors. > > > > So, essentially the messages are harmless and nothing to worry too > > much, but surely it doesn't look sexy. > > > > The patch below should fix the problem. Please give it a try. > > Yes, your patch finally fixes the problem. > Thank you Takashi-san. Thanks for your quick test! If Daniel has no objection with that patch, I'm going to merge it. Takashi

3 2

[alsa-devel] how to create alsa nodes?
by loody 12 Dec '13

12 Dec '13

Dear all: In destop x86 system, the udev will automatically create the nodes in snd directory, such as pcmC0D0c, pcmC0D1c, etc. I try to let the usb sound device work on mips platform, but I have no idea what is the proper way to create these nodes. Should I 1. create them one by one by hand from copying the attributes in my PC (if I have 2 or more usb sound devices, I have to first plug them in pc and monitor the nodes) 2. smarter way to do so? -- Regards, miloody

7 19

[alsa-devel] alsa/asoundlib.h file not found
by Appalayagari Sreedhar 05 Dec '13

05 Dec '13

Hi, I have integrated the codec driver and the kernel log shows the sound card is detected. now i want to test the codec. i tried the sample test code for capture and playback. i am getting a compilation errors due to the asoundlib.h. please find the error log for my application. main.c:2:23: error: asoundlib.h: No such file or directory main.c: In function 'main': main.c:7: error: 'snd_pcm_t' undeclared (first use in this function) main.c:7: error: (Each undeclared identifier is reported only once main.c:7: error: for each function it appears in.) main.c:7: error: 'pcm_handle' undeclared (first use in this function) main.c:10: error: 'snd_pcm_stream_t' undeclared (first use in this function) main.c:10: error: expected ';' before 'stream' main.c:15: error: 'snd_pcm_hw_params_t' undeclared (first use in this function) main.c:15: error: 'hwparams' undeclared (first use in this function) main.c:22: warning: 'return' with a value, in function returning void main.c:34: error: 'snd_pcm_uframes_t' undeclared (first use in this function) main.c:34: error: expected ';' before 'periodsize' main.c:46: error: 'SND_PCM_ACCESS_RW_INTERLEAVED' undeclared (first use in this function) main.c:48: warning: 'return' with a value, in function returning void main.c:52: error: 'SND_PCM_FORMAT_S16_LE' undeclared (first use in this function) main.c:54: warning: 'return' with a value, in function returning void main.c:62: warning: 'return' with a value, in function returning void main.c:65:19: warning: missing terminating " character main.c:65: error: missing terminating " character main.c:66: error: expected expression before '==' token main.c:66: error: stray '\' in program main.c:66:31: warning: missing terminating " character main.c:66: error: missing terminating " character main.c:67: error: expected ';' before '}' token main.c:72: warning: 'return' with a value, in function returning void main.c:78: warning: 'return' with a value, in function returning void main.c:86: error: 'periodsize' undeclared (first use in this function) main.c:88: warning: 'return' with a value, in function returning void main.c:99: warning: 'return' with a value, in function returning void main.c:112: warning: incompatible implicit declaration of built-in function 'malloc' main.c:115: error: 'num_frames' undeclared (first use in this function) main.c:5: warning: return type of 'main' is not 'int' please suggest what needs to be done to resolve this issue asap. Thank you, Sreedhar.

5 4

[alsa-devel] snd_device_name_hint segfaults
by Forest Bond 03 Oct '13

03 Oct '13

Hi, I'm seeing segfaults with VLC similar to those reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/059168.ht… This is on Ubuntu 12.04 with the latest alsa-lib package (1.0.25-1ubuntu10.2) and the following patches also applied: * 2cfc8b9b44a8e493c41b3d63d5a00b306a18a5ed snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() * 44c1a623dd1fc9e831616b663bebc54ca98df994 Add snd_lib_error_set_local() to install a thread-local error handler. * 25dbb102810b31c02358904d70d53c960fb0a10e snd_device_name_hint(): do not change the global error handler. * f49b2dc522a2564315c76d075203b15a39941e8a snd_device_name_hint(): do not use global snd_config. * 7f2b2c8c1650a1883b48abfcdb455138943854f9 conf: Fix a memory access violation resulting from improper error propogation So I think there are still issues when calling snd_device_name_hint(-1, ...). Here is the backtrace: #0 0x003f0e78 in __strcpy_chk () from /lib/i386-linux-gnu/libc.so.6 #1 0x0625084b in strcpy (__src=0x0, __dest=0xb6025e80 "pcm.(null)") at /usr/include/i386-li nux-gnu/bits/string3.h:105 #2 try_config (config=0xaf1c4df0, list=0xb52fe804, base=0x323e962 "pcm", name=0x0) at nameh int.c:243 #3 0x06251bc5 in add_software_devices (list=0xb52fe804, config=0xaf1c4df0) at namehint.c:51 2 #4 snd_device_name_hint (card=-1, iface=<optimized out>, hints=0xb52fe90c) at namehint.c:59 1 #5 0x0323c9bb in GetDevices (obj=0xb6271700, item=0x0, prefs_dev=<optimized out>) at alsa.c :721 #6 0x0323d4c0 in Probe (dev=0xb6011df8 "default", obj=0xb6271700) at alsa.c:161 #7 Open (obj=0xb6271700) at alsa.c:539 #8 0x0a6c73d0 in generic_start (func=0x323cd10, ap=0xb52fed98 "\273") at modules/modules.c:422 #9 0x0a6c7a3a in vlc_module_load (p_this=0xb6271700, psz_capability=0xa70b959 "audio output", psz_name=<optimized out>, b_strict=false, probe=0xa6c73c0 <generic_start>) at modules/modules.c:347 #10 0x0a6c8002 in module_need (obj=0xb6271700, cap=0xa70b959 "audio output", name=0xa70d2bd "$aout", strict=false) at modules/modules.c:437 #11 0x0a6b1242 in aout_OutputNew (p_aout=0xb6271700, p_format=0xb52feeac) at audio_output/output.c:59 #12 0x0a6ad77f in aout_DecNew (p_aout=0xb6271700, p_format=0xb52feeac, p_replay_gain=0xaef02a14, p_request_vout=0xb52feed0) at audio_output/dec.c:112 #13 0x0a67066e in aout_new_buffer (p_dec=0xaef02870, i_samples=1024) at input/decoder.c:2286 #14 0x0a672690 in decoder_NewAudioBuffer (p_decoder=0xaef02870, i_size=1024) at input/decoder.c:213 #15 0x04032f8c in DecodeBlock (p_dec=0xaef02870, pp_block=0xb52ff05c) at faad.c:464 #16 0x0a670891 in DecoderDecodeAudio (p_dec=0xaef02870, p_block=0xaef03000) at input/decoder.c:1291 #17 0x0a6716f5 in DecoderProcessAudio (b_flush=false, p_block=0xaef03000, p_dec=0xaef02870) at input/decoder.c:1936 #18 DecoderProcess (p_dec=0xaef02870, p_block=<optimized out>) at input/decoder.c:2059 #19 0x0a671959 in DecoderThread (p_data=0xaef02870) at input/decoder.c:938 #20 0x001fbd4c in start_thread () from /lib/i386-linux-gnu/libpthread.so.0 #21 0x003ddace in clone () from /lib/i386-linux-gnu/libc.so.6 As a work around I will probably patch VLC to not call snd_device_name_hint this way. I see this has been done for other applications: http://code.google.com/p/chromium/issues/detail?id=95797 http://code.mythtv.org/trac/ticket/10809 But I wanted to try diagnosing the issue first. Thoughts? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

4 6

[alsa-devel] Fwd: Re: Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 18 Sep '13

18 Sep '13

Here's my initial mail, but with gzipped outputs. Regards, Colomban -------- Message original -------- Sujet: Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again) Date : Thu, 10 Nov 2011 02:11:17 +0100 De : Colomban Wendling <lists.ban(a)herbesfolles.org> Pour : Takashi Iwai <tiwai(a)suse.de> Copie à : alsa-devel(a)alsa-project.org Hi again, and sorry for the delay. Le 08/11/2011 07:32, Takashi Iwai a écrit : > At Mon, 07 Nov 2011 23:45:40 +0100, > Colomban Wendling wrote: >> >> Hi, >> >> Back to 3.0, the sound on my soundcard [1] started to be choppy, I >> reported it [2] and it got fixed (thanks to Takashi Iwai!). >> >> However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): >> I've got similar choppy sound again. >> >> I bisected the few commits that happened on >> sound/pci/hda/patch_realtek.c, and finally found the villain: >> >> "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" >> >> Reverting it from v3.1 fixes the problem. > > Does it? Very weird. This patch has nothing to do with the HD-audio > controller side but purely a codec change. Yep, I just check once again and it really fixes the problem here. >> master (94956ee) also suffers of the problem, but I couldn't test >> without that commit because reverting it fails, too much changes happened. > > Did you try to pass position_fix=1 (or 2) option? > Also, passing enable_msi=0 (or 1) may change anything? I just tried with the 3 kernels: * vanilla v3.1 * vanilla 3.2-rc1 (master at 1ea6b8f) * v3.1 with 8974bd51 reverted and none of the option changed anything: both vanilla always failed, the third always succeeded. Though, I haven't combined the options, just tried each at once. > In anyway, please give alsa-info.sh output again with the fresh > kernel. Run it with --no-upload option and attach the output. Here it is, for the 3 kernels cited above. Regards, Colomban

4 14

[alsa-devel] Support new HDA codec
by Jorge Fernandez Monteagudo 26 Aug '13

26 Aug '13

Hi all, This is my first email to the devel mailing list. I'm asking the same question in the user mailing list but I suspect that I'll have to add some code in order to get this hda codec from Wolfson to work. Attached is the log I get from alsa-info.sh. This codec has three output paths: 1) Node 0x06, DAC1, and Node 0x11, Port A, the stereo headphone playback, which is the default one and it seems it works because I can see the stream set to 5 when a sound is played. 2) Node 0x04, SPDIF Tx1, and Node 0x0f, SPDIF OUT, the SPDIF transmit functionality provides an IEC-60958-3 compatible S/PIDF output which I would like to enable but without luck yet. I can enable the node 0x04, with hda-verb, and the spdif amplifier receives spdif frames but without audio info. 3) Node 0x07, DAC2, and Node 0x12, Port G, the stereo line playback with a differential stereo output wich I would like to enable but without luck again. I just can mute/unmute the node but without effect in the outputs. I would like to know if I have to develop a patch for this device or with the attached log info I can expect to use it with the current code in kernel. Now, with alsamixer and other config tools I can't see any control related to the outputs for 2) and 3)... Any hint is welcome! Thanks Jorge --- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Tue Feb 21 09:44:41 UTC 2012 !!Linux Distribution !!------------------ Ubuntu 11.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 11.10" !!DMI Information !!--------------- Manufacturer: nVidia Product Name: MMP9-ION Product Version: To Be Filled By O.E.M. !!Kernel Information !!------------------ Kernel release: 3.0.0-12-generic Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.24 Library version: 1.0.24.1 Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfae78000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb84 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Wolfson Microelectronics ID 8800 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x1aec8800 Subsystem Id: 0x00000100 Revision Id: 0x100301 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=1 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 Node 0x02 [Audio Input] wcaps 0x1005d1: Stereo Device: name="HDA Generic", type="Audio", device=0 Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x1ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=02, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x0a Processing caps: benign=0, ncoeff=0 Node 0x03 [Audio Input] wcaps 0x1005d1: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=03, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Output] wcaps 0x2d1: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=04, enabled=0 Processing caps: benign=0, ncoeff=0 Node 0x05 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Copyright Digital category: 0x0 PCM: rates [0x1f0]: 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=05, enabled=0 Connection: 1 0x10 Processing caps: benign=0, ncoeff=0 Node 0x06 [Audio Output] wcaps 0x495: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="HDA Generic", type="Audio", device=0 Amp-Out caps: N/A Amp-Out vals: [0x57 0x57] Converter: stream=5, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=06, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x07 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=07, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x08 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=08, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x09 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x0b* 0x0e Node 0x0a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x18 0x18] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x98 0x98] Connection: 1 0x0d Node 0x0c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000064: IN Detect Balanced Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=0c, enabled=0 Node 0x0d [Pin Complex] wcaps 0x400083: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00001767: IN Detect Balanced Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=0d, enabled=0 Node 0x0e [Pin Complex] wcaps 0x40020b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000020: IN Pin Default 0x10a600f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x014190f0: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=0f, enabled=0 Connection: 1 0x04 Node 0x10 [Pin Complex] wcaps 0x400281: Stereo Digital Pincap 0x00000020: IN Pin Default 0x01c520f0: [Jack] SPDIF In at Ext Rear Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=10, enabled=0 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=11, enabled=0 Connection: 1 0x06 Node 0x12 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016121: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=12, enabled=0 Connection: 1 0x07 Node 0x13 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011022: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=13, enabled=0 Connection: 1 0x08 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x04, nsteps=0x04, stepsize=0x17, mute=0 Amp-Out vals: [0x04] Node 0x15 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=2 Digital: Digital category: 0x0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=15, enabled=0 Connection: 1 0x16 Processing caps: benign=0, ncoeff=0 Node 0x16 [Pin Complex] wcaps 0x40030b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000030: IN OUT Pin Default 0x10a601f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Connection: 1 0x17 Node 0x17 [Audio Output] wcaps 0x291: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=17, enabled=0 --endcollapse-- !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfae78000 irq 20' Mixer name : 'Wolfson Microelectronics ID 8800' Components : 'HDA:1aec8800,00000100,00100301' Controls : 3 Simple ctrls : 2 Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 87 [69%] [-20.00dB] [on] Front Right: Playback 87 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { iface MIXER name 'Master Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.2 { iface MIXER name 'Master Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -6350 dbmax 0 dbvalue.0 -2000 dbvalue.1 -2000 } } control.3 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 5 Feb 21 09:38 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Feb 21 09:38 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 3 Feb 21 09:38 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 2 Feb 21 09:39 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Feb 21 09:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Feb 21 09:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Feb 21 09:38 . drwxr-xr-x 3 root root 180 Feb 21 09:38 .. lrwxrwxrwx 1 root root 12 Feb 21 09:38 pci-0000:00:08.0 -> ../controlC0 Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o CONFIDENCIAL. 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[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

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Re: [alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Daniel Mack 23 Aug '13

23 Aug '13

[Please do not drop the mailing list address] On 19.10.2012 15:45, Didier 'Ptitjes' Villevalois wrote: > On ven., 2012-10-19 at 15:20 +0200, Daniel Mack wrote: >> On 19.10.2012 15:09, Didier 'Ptitjes' Villevalois wrote: >>> Hello, >>> >>> I've bought a Reloop Play usb soundcard and I'd like to make it work >>> with ALSA. >>> >>> It is a 4-channels USB soundcard. It is currently correctly recognized >>> by ALSA as a 4-channels soundcard and I can correctly open the channels >>> and feed them with audio. However, no audio is output by the box. >> >> You're most probably affected by a regression in the driver that was >> fixes in 3.6.0 and 3.5.5 (and all later kernels). Could you please try >> one of these and tell us if that works? > > Sure, I'll do that now (I'm currently using 3.5.4 + mbp8,3 patches). > I guess 3.5.7 is OK ? I'm building it now and will reboot on that. Any kernel >= 3.6.0 and 3.5.5 should be ok. >>> It seems to contain a Texas Instrument TUSB3200AC chipset (same chipset >>> as the Maya44 USB and Hercules DJ console). I guess the problem is a >>> firmware problem but I don't know what to do about it. >>> >>> I've got a MacOS X driver that works correctly, if that of any help. (I >>> suppose the Windows driver works too but I don't have any Windows at >>> hand.) >> >> Did you install any vendor driver on OS X or does it just work with the >> one that ships with the OS? > > Yep, I had to install the Vendor-provided driver. What was the effect without that special driver? Could you see the device listed in the AudioMIDISetup? Could you stream audio? IOW, is the effect you see there comparable to how your Linux box behaves? You could try and remove that driver again and see what happens. > But it seems the > built-in USB soundcard support on MacOS is very poor. No it's not. In fact, it's actually very good. Vendors who keep their hardware compliant to the USB audio spec don't have to ship any driver at all. And this is how it should be on Linux, too. > Also the mpkg install file (inside the provided install dmg file) was > called something like "firmware0_and_driver.mpkg". This is why I > suspected a firmware problem. However I tried to look at the package > content and did not find any .bin file or whatever could be a standalone > firmware file. Maybe it is embedded in the driver install executable. Maybe, but they could also write the firmware permanently to the device once with that tool. That would be perfect for Linux users. Daniel

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[alsa-devel] [git pull] [PATCH 0/7] add support for many Roland and Yamaha devices
by Clemens Ladisch 16 Aug '13

16 Aug '13

The following changes since commit f722406faae2d073cc1d01063d1123c35425939e: Linux 3.10-rc1 (2013-05-11 17:14:08 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git full-roland-support for you to fetch changes up to b7f33917bcd993ff81f3f80b9dc1890fb7410c6d: ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE (2013-06-27 21:59:50 +0200) ---------------------------------------------------------------- Clemens Ladisch (7): ALSA: usb-audio: store protocol version in struct audioformat ALSA: usb-audio: detect implicit feedback on Roland devices ALSA: usb-audio: add support for many Roland/Yamaha devices ALSA: usb-audio: add MIDI port names for some Roland devices ALSA: usb-audio: remove superfluous Roland quirks ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE sound/usb/card.h | 1 + sound/usb/clock.c | 4 +- sound/usb/format.c | 34 +--- sound/usb/format.h | 2 +- sound/usb/midi.c | 74 +++++++ sound/usb/pcm.c | 45 ++++- sound/usb/quirks-table.h | 497 +++++++++-------------------------------------- sound/usb/quirks.c | 209 ++++++++++++++++++++ sound/usb/stream.c | 18 +- sound/usb/usbaudio.h | 2 + 10 files changed, 445 insertions(+), 441 deletions(-)

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Re: [alsa-devel] Query on Audio DMA using DMAEngine
by Lars-Peter Clausen 14 Aug '13

14 Aug '13

Added alsa-devel to Cc. On 06/28/2013 05:27 AM, Fernandes, Joel wrote: > Hi Lars, > > Hope you are doing well. > > I am implementing Cyclic DMA support in the EDMA driver that is used by Davinci and now newer TI SoCs. > I am thinking once I am done I can plug it into the snd_dmaengine framework. > > Currently however, the davinci-pcm code directly programs the EDMA. That is what I am working to replace with a single driver and adapt to the snd dmaengine framework. However, once the current code in davinci-pcm uses internal RAM as an intermediate step in the whole DMA process (First data is TX to IRAM from DRAM and then from DRAM to the audio device). > > Do you have any ideas on how we can adapt to the framework, such that we can till use the IRAM? Are there any existing implementations out there that do something similar? Hm, I guess using the snd_dmaengine_pcm helper functions here shouldn't be too hard. Using the generic snd_dmaengine_pcm driver will require some extensions to it though. The mmp platform (pxa/mmp-pcm.c) is also using some kind of on-chip memory, so having support for this in the generic driver certainly makes sense. For the chaining you'd probably have to extend the dmaengine framework, since this kind of interleaved mem-to-mem and mem-to-dev cyclic transfer is currently not possible. I'm wondering though why do you need to copy the data to RAM first, is it not possible to map the IRAM to userspace? - Lars

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