August 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] multistream singlechanel DACs
by Rafal Powierski 25 Aug '12

25 Aug '12

Developers, how to enable multiple playback streaming?: I am manipulating bios verb pin table, for special analog hardware, using ALC886 and kubuntu / 64bit. I need a number of separate stereo (only stereo) i/o's I have correctly configured inputs: 4. single stereo adc port-C 1Ah 5. single spdif input 1Fh I have configured 3 independent stereo outputs, but ALSA is joining them into one pcmC1D0p device. 1.port-A,15h 2.port-E,1Bh 3.port-D,14h I have tried this on win7 and I got all associations as a separate devices. in my kubuntu 12.4LTS I have only one playback device. Mixer is showing even more then needed number of volume controls, But there is only one playback device. it seams, all 3 outputs are connected to single DAC. I have tried to buildup more input devices - this is working good. I can have up to 3 (pcmC1D0c, pcmC1D1c pcmC1D2c) I can also have 2 outputs if I enable spdif-out. But there is no separate DAC assignment to separate devices resulting from my config. Is it possible to get independent DACs devices under ALSA? or else is no such configuration possible and I have to use single multiple channel device?

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 24 Aug '12

24 Aug '12

Hi, On 24.08.2012 21:08, Josh Boyer wrote: > We've had a report[1] that the Logitech USB headset 0003:046D:0A0C isn't > working with 3.6-rc3. It seems the last working kernel was based on > commit 10c63c9, and it first stopped working with a kernel based on > commit 23dcfa6. There are only a few ALSA commits between those > revisions, so hopefully this is something that is fairly easy to > identify. The only commit to USB audio in that set is: [...] > !!ALSA/HDA dmesg > !!------------------ > > [ 38.190306] SELinux: initialized (dev configfs, type configfs), uses genfs_contexts > [ 38.229616] snd_hda_intel 0000:00:1b.0: irq 66 for MSI/MSI-X > [ 38.270699] ALSA sound/usb/mixer.c:866 6:0: cannot get min/max values for control 2 (id 6) > [ 38.274097] ALSA sound/usb/mixer.c:866 1:0: cannot get min/max values for control 2 (id 1) > [ 38.276753] ALSA sound/usb/mixer.c:866 2:0: cannot get min/max values for control 2 (id 2) > [ 38.279322] ALSA sound/pci/hda/hda_auto_parser.c:322 autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) type:line > [ 38.279326] ALSA sound/pci/hda/hda_auto_parser.c:326 speaker_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 38.279329] ALSA sound/pci/hda/hda_auto_parser.c:330 hp_outs=1 (0xd/0x0/0x0/0x0/0x0) > [ 38.279331] ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 > [ 38.279333] ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: > [ 38.279336] ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x10 > [ 38.279342] ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0xf > [ 38.285983] usbcore: registered new interface driver snd-usb-audio Also, according to this dmesg, the device is probed just fine (which the commit you mentioned wouldn't change anything about though). So what does "isn't working anymore" refer to precisely? Are there any more dmesg entries generated once the stream is started? Daniel

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 24 Aug '12

24 Aug '12

On Fri, Aug 24, 2012 at 9:08 PM, Josh Boyer <jwboyer(a)redhat.com> wrote: > Hi All, > > We've had a report[1] that the Logitech USB headset 0003:046D:0A0C isn't > working with 3.6-rc3. It seems the last working kernel was based on > commit 10c63c9, and it first stopped working with a kernel based on > commit 23dcfa6. There are only a few ALSA commits between those > revisions, so hopefully this is something that is fairly easy to > identify. The only commit to USB audio in that set is: > > commit e9ba389c5ffc4dd29dfe17e00e48877302111135 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Wed Aug 15 12:32:00 2012 +0200 > > ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream > > > I've CC'd the reporter and attached the alsa-info is below. Does it work again once you revert that commit? Daniel > [1] https://bugzilla.redhat.com/show_bug.cgi?id=851619 > > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.60 > !!################################ > > !!Script ran on: Fri Aug 24 18:35:42 UTC 2012 > > > !!Linux Distribution > !!------------------ > > Fedora release 18 (Rawhide) Fedora release 18 (Rawhide) NAME=Fedora ID=fedora PRETTY_NAME="Fedora 18 (Rawhide)" CPE_NAME="cpe:/o:fedoraproject:fedora:18" Fedora release 18 (Rawhide) Fedora release 18 (Rawhide) > > > !!DMI Information > !!--------------- > > Manufacturer: Dell Inc. > Product Name: Precision WorkStation 690 > Product Version: > > > !!Kernel Information > !!------------------ > > Kernel release: 3.6.0-0.rc3.git0.1.fc18.x86_64 > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.25 > Library version: 1.0.25 > Utilities version: 1.0.25 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > snd_usb_audio > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > aRts: > Installed - Yes (/usr/bin/artsd) > Running - No > > Jack: > Installed - Yes (/usr/bin/jackd) > Running - No > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xfcffc000 irq 66 > 1 [Headset ]: USB-Audio - Logitech USB Headset > Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 631xESB/632xESB High Definition Audio Controller (rev 09) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:269a (rev 09) > Subsystem: 1028:01c0 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > align_buffer_size : -1 > bdl_pos_adj : 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 > beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N > 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 > snoop : Y > > !!Module: snd_usb_audio > async_unlink : Y > device_setup : 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 > 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) > ignore_ctl_error : N > 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 > nrpacks : 8 > pid : -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 > vid : -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 > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: SigmaTel STAC9200 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x83847690 > Subsystem Id: 0x102801c0 > Revision Id: 0x102201 > No Modem Function Group found > Default PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 > GPIO: io=4, 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=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0xd0401: Stereo > Device: name="STAC92xx Analog", type="Audio", device=0 > Converter: stream=8, channel=0 > Power states: > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo > Device: name="STAC92xx Analog", type="Audio", device=0 > Converter: stream=4, channel=0 > SDI-Select: 0 > Power states: > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x0a > Processing caps: benign=0, ncoeff=0 > Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x160]: 44100 48000 96000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Connection: 1 > 0x08 > Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x1e0]: 44100 48000 88200 96000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital > Delay: 3 samples > Node 0x07 [Audio Selector] wcaps 0x300901: Stereo R/L > Connection: 3 > 0x02* 0x08 0x0a > Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital > Pincap 0x00010024: IN EAPD Detect > EAPD 0x0: > Pin Default 0x400001f0: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: > Power: setting=D0, actual=D0 > Delay: 3 samples > Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x00000010: OUT > Pin Default 0x400001f1: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 2 > 0x05* 0x0a > Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Control: name="Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Capture Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x07 0x07] > Connection: 1 > 0x0c > Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out > Control: name="PCM Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="PCM Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: N/A > Amp-Out vals: [0x9f 0x9f] > Connection: 1 > 0x07 > Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Control: name="Mux Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 5 > 0x10* 0x0f 0x0e 0x0d 0x12 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Control: name="Front Headphone Jack", index=0, device=0 > Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense > Pin Default 0x0221401f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Connection: 1 > 0x0b > Node 0x0e [Pin Complex] wcaps 0x400181: Stereo > Control: name="Line Out Jack", index=0, device=0 > Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense > Pin Default 0x01014010: [Jack] Line Out at Ext Rear > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=02, enabled=1 > Connection: 1 > 0x0b > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Control: name="Line Jack", index=0, device=0 > Pincap 0x00000037: IN OUT Detect Trigger ImpSense > Pin Default 0x01813020: [Jack] Line In at Ext Rear > Conn = 1/8, Color = Blue > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x20: IN > Unsolicited: tag=04, enabled=1 > Connection: 1 > 0x0b > Node 0x10 [Pin Complex] wcaps 0x400181: Stereo > Control: name="Mic Jack", index=0, device=0 > Pincap 0x00001737: IN OUT Detect Trigger ImpSense > Vref caps: HIZ 50 GRD 80 > Pin Default 0x02a19021: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=03, enabled=1 > Connection: 1 > 0x0b > Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out > Control: name="LFE Playback Volume", index=0, device=0 > ControlAmp: chs=1, dir=Out, idx=0, ofs=0 > Control: name="LFE Playback Switch", index=0, device=0 > ControlAmp: chs=1, dir=Out, idx=0, ofs=0 > Control: name="Speaker Phantom Jack", index=0, device=0 > Amp-Out caps: N/A > Amp-Out vals: [0x00] > Pincap 0x00000010: OUT > Pin Default 0x90100140: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x13 > Node 0x12 [Pin Complex] wcaps 0x400001: Stereo > Pincap 0x00000020: IN > Pin Default 0x400001f2: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x07 > Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > --endcollapse-- > > > !!USB Mixer information > !!--------------------------- > --startcollapse-- > > USB Mixer: usb_id=0x046d0a0c, ctrlif=0, ctlerr=0 > Card: Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed > Unit: 1 > Control: name="Speaker Playback Volume", index=0 > Info: id=1, control=2, cmask=0x3, channels=2, type="S16" > Volume: min=-10496, max=768, dBmin=-4100, dBmax=300 > Unit: 1 > Control: name="Speaker Playback Switch", index=0 > Info: id=1, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > Unit: 2 > Control: name="Mic Capture Volume", index=0 > Info: id=2, control=2, cmask=0x0, channels=1, type="S16" > Volume: min=4096, max=7424, dBmin=1600, dBmax=2900 > Unit: 2 > Control: name="Mic Capture Switch", index=0 > Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > Unit: 6 > Control: name="Mic Playback Volume", index=0 > Info: id=6, control=2, cmask=0x0, channels=1, type="S16" > Volume: min=-10496, max=-3072, dBmin=-4100, dBmax=-1200 > Unit: 6 > Control: name="Mic Playback Switch", index=0 > Info: id=6, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw----+ 1 root audio 116, 8 Aug 24 12:18 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 4 Aug 24 12:18 /dev/snd/controlC1 > crw-rw----+ 1 root audio 116, 7 Aug 24 12:18 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 6 Aug 24 12:19 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 5 Aug 24 12:19 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 3 Aug 24 12:19 /dev/snd/pcmC1D0c > crw-rw----+ 1 root audio 116, 2 Aug 24 12:30 /dev/snd/pcmC1D0p > crw-rw----+ 1 root audio 116, 1 Aug 24 12:18 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 Aug 24 12:18 /dev/snd/timer > > /dev/snd/by-id: > total 0 > drwxr-xr-x. 2 root root 60 Aug 24 12:18 . > drwxr-xr-x. 4 root root 260 Aug 24 12:18 .. > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 usb-Logitech_Logitech_USB_Headset-00 -> ../controlC1 > > /dev/snd/by-path: > total 0 > drwxr-xr-x. 2 root root 80 Aug 24 12:18 . > drwxr-xr-x. 4 root root 260 Aug 24 12:18 .. > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 pci-0000:00:1d.7-usb-0:6.4:1.0 -> ../controlC1 > > > !!ALSA configuration files > !!------------------------ > > !!System wide config file (/etc/asound.conf) > > # > # Place your global alsa-lib configuration here... > # > > > !!Aplay/Arecord output > !!------------ > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: Headset [Logitech USB Headset], device 0: USB Audio [USB Audio] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: Headset [Logitech USB Headset], device 0: USB Audio [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xfcffc000 irq 66' > Mixer name : 'SigmaTel STAC9200' > Components : 'HDA:83847690,102801c0,00102201' > Controls : 15 > Simple ctrls : 6 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 31 > Mono: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'PCM',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 31 [100%] [0.00dB] [off] > Front Right: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'LFE',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 31 > Mono: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 7 [47%] [10.50dB] [on] > Front Right: Capture 7 [47%] [10.50dB] [on] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Line' > Item0: 'Mic' > Simple mixer control 'Mux',0 > Capabilities: cvolume penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 4 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > > !!-------Mixer controls for card 1 [Headset] > > Card hw:1 'Headset'/'Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed' > Mixer name : 'USB Mixer' > Components : 'USB046d:0a0c' > Controls : 6 > Simple ctrls : 2 > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch pswitch-joined penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 44 > Mono: > Front Left: Playback 35 [80%] [-6.00dB] [on] > Front Right: Playback 35 [80%] [-6.00dB] [on] > Simple mixer control 'Mic',0 > Capabilities: pvolume pvolume-joined cvolume cvolume-joined pswitch pswitch-joined cswitch cswitch-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: Playback 0 - 29 Capture 0 - 13 > Mono: Playback 0 [0%] [-41.00dB] [off] Capture 13 [100%] [29.00dB] [on] > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > iface MIXER > name 'PCM Playback Volume' > value.0 31 > value.1 31 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -9999999 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.2 { > iface MIXER > name 'PCM Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.3 { > iface MIXER > name 'Capture Volume' > value.0 7 > value.1 7 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 15' > dbmin 0 > dbmax 2250 > dbvalue.0 1050 > dbvalue.1 1050 > } > } > control.4 { > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.5 { > iface MIXER > name 'LFE Playback Volume' > value 31 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 31' > dbmin -4650 > dbmax 0 > dbvalue.0 0 > } > } > control.6 { > iface MIXER > name 'LFE Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.7 { > iface MIXER > name 'Mux Capture Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 4' > dbmin 0 > dbmax 4000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.8 { > iface MIXER > name 'Input Source' > value Mic > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Mic > item.1 Line > } > } > control.9 { > iface MIXER > name 'Master Playback Volume' > value 31 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 31' > dbmin -9999999 > dbmax 0 > dbvalue.0 0 > } > } > control.10 { > iface MIXER > name 'Master Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.11 { > iface CARD > name 'Line Out Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.12 { > iface CARD > name 'Front Headphone Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.13 { > iface CARD > name 'Speaker Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.14 { > iface CARD > name 'Mic Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.15 { > iface CARD > name 'Line Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > } > state.Headset { > control.1 { > iface MIXER > name 'Mic Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.2 { > iface MIXER > name 'Mic Playback Volume' > value 0 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 29' > dbmin -4100 > dbmax -1200 > dbvalue.0 -4100 > } > } > control.3 { > iface MIXER > name 'Speaker Playback Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.4 { > iface MIXER > name 'Speaker Playback Volume' > value.0 35 > value.1 35 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 44' > dbmin -4100 > dbmax 300 > dbvalue.0 -600 > dbvalue.1 -600 > } > } > control.5 { > iface MIXER > name 'Mic Capture Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.6 { > iface MIXER > name 'Mic Capture Volume' > value 13 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 13' > dbmin 1600 > dbmax 2900 > dbvalue.0 2900 > } > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > fuse > nf_conntrack_ipv4 > nf_defrag_ipv4 > xt_state > nf_conntrack > snd_hda_codec_idt > snd_usb_audio > snd_hda_intel > snd_usbmidi_lib > snd_hda_codec > snd_rawmidi > snd_hwdep > iTCO_wdt > iTCO_vendor_support > lpc_ich > i5000_edac > snd_seq > snd_seq_device > edac_core > snd_pcm > mfd_core > serio_raw > ppdev > snd_page_alloc > snd_timer > coretemp > snd > i5k_amb > soundcore > parport_pc > dcdbas > microcode > shpchp > parport > i2c_i801 > vhost_net > tun > macvtap > macvlan > kvm_intel > kvm > uinput > binfmt_misc > xts > gf128mul > dm_crypt > raid1 > radeon > i2c_algo_bit > drm_kms_helper > ttm > drm > tg3 > mptsas > mptscsih > mptbase > i2c_core > scsi_transport_sas > sunrpc > be2iscsi > bnx2i > cnic > uio > cxgb4i > cxgb4 > cxgb3 > mdio > libcxgbi > libiscsi_tcp > qla4xxx > iscsi_boot_sysfs > libiscsi > scsi_transport_iscsi > > > !!Sysfs Files > !!----------- > > /sys/class/sound/hwC0D0/init_pin_configs: > 0x08 0x40f000f0 > 0x09 0x40f000f1 > 0x0d 0x0221101f > 0x0e 0x01014010 > 0x0f 0x0181302e > 0x10 0x02a11020 > 0x11 0x40f000f2 > 0x12 0x40f000f3 > > /sys/class/sound/hwC0D0/driver_pin_configs: > 0x08 0x400001f0 > 0x09 0x400001f1 > 0x0d 0x0221401f > 0x0e 0x01014010 > 0x0f 0x01813020 > 0x10 0x02a19021 > 0x11 0x90100140 > 0x12 0x400001f2 > > /sys/class/sound/hwC0D0/user_pin_configs: > > /sys/class/sound/hwC0D0/init_verbs: > > > !!ALSA/HDA dmesg > !!------------------ > > [ 38.190306] SELinux: initialized (dev configfs, type configfs), uses genfs_contexts > [ 38.229616] snd_hda_intel 0000:00:1b.0: irq 66 for MSI/MSI-X > [ 38.270699] ALSA sound/usb/mixer.c:866 6:0: cannot get min/max values for control 2 (id 6) > [ 38.274097] ALSA sound/usb/mixer.c:866 1:0: cannot get min/max values for control 2 (id 1) > [ 38.276753] ALSA sound/usb/mixer.c:866 2:0: cannot get min/max values for control 2 (id 2) > [ 38.279322] ALSA sound/pci/hda/hda_auto_parser.c:322 autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) type:line > [ 38.279326] ALSA sound/pci/hda/hda_auto_parser.c:326 speaker_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 38.279329] ALSA sound/pci/hda/hda_auto_parser.c:330 hp_outs=1 (0xd/0x0/0x0/0x0/0x0) > [ 38.279331] ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 > [ 38.279333] ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: > [ 38.279336] ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x10 > [ 38.279342] ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0xf > [ 38.285983] usbcore: registered new interface driver snd-usb-audio > [ 38.290348] input: HDA Intel Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 > [ 38.294975] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 > [ 38.297406] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 > [ 38.301369] input: HDA Intel Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 > [ 39.377352] EXT4-fs: Warning: mounting with data=journal disables delayed allocation and O_DIRECT support! > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

1 0

[alsa-devel] [PATCH] ASoC: samsung dma - prepare dma in start/resume trigger path
by Dylan Reid 24 Aug '12

24 Aug '12

On suspend all dma transactions are terminated (by the stop/suspend trigger calling ops->stop), they need to be set up again during resume. Move the prepare op to the start/resume/unpause trigger path so that dma and audio will restart on resume/unpause. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..5772762 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -250,6 +250,7 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dma_prepare(substream); prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; @@ -352,7 +353,6 @@ static struct snd_pcm_ops dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = dma_hw_params, .hw_free = dma_hw_free, - .prepare = dma_prepare, .trigger = dma_trigger, .pointer = dma_pointer, .mmap = dma_mmap, -- 1.7.7.3

3 6

Re: [alsa-devel] linux-next: Tree for Aug 23 (sound/isa/cmi)
by Randy Dunlap 24 Aug '12

24 Aug '12

On 08/22/2012 10:22 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120822: > on i386: CC sound/isa/cmi8328.o sound/isa/cmi8328.c: In function 'snd_cmi8328_remove': sound/isa/cmi8328.c:416:24: error: 'cmi' undeclared (first use in this function) sound/isa/cmi8328.c:416:24: note: each undeclared identifier is reported only once for each function it appears in make[3]: *** [sound/isa/cmi8328.o] Error 1 Full randconfig file is attached. -- ~Randy

2 1

[alsa-devel] [PATCH v2 1/1] [ALSA] hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
by mengdong.lin＠intel.com 24 Aug '12

24 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 6 ++++++ sound/pci/hda/hda_intel.c | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d6b3a6d..e618394 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1320,11 +1320,13 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_CLKSTOP); if (!codec->d3_stop_clk) bus->power_keep_link_on = 1; +#endif /* power-up all before initialization */ hda_set_power_state(codec, @@ -3523,7 +3525,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk_ok = 0; +#endif if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); @@ -3548,9 +3552,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, break; } +#ifdef CONFIG_SND_HDA_POWER_SAVE if ((power_state == AC_PWRST_D3) && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) codec->d3_stop_clk_ok = 1; +#endif } #ifdef CONFIG_SND_HDA_HWDEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2baa3ee..97307df 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2515,8 +2515,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- 1.7.9.5

2 1

Re: [alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Ricardo Neri 24 Aug '12

24 Aug '12

On 08/21/2012 07:39 AM, Clark, Rob wrote: > On Tue, Aug 21, 2012 at 1:01 AM, Tomi Valkeinen <tomi.valkeinen(a)ti.com> wrote: >> On Mon, 2012-08-20 at 20:47 -0500, Ricardo Neri wrote: >>> Hello! >>> >>> I have been working on prototypes for the ASoC OMAP HDMI audio driver to >>> propagate events from the HDMI output (e.g., display getting >>> enabled/disabled/suspended). This for the users of the driver to react >>> to such events. For instance, if the display is disabled or disconected, >>> audio could be stopped, rerouted or whatever other decision the user >>> makes. This is needed because, if, for instance, the HDMI IP goes off, >>> audio will stall and the audio users will only see a "playback write >>> error (DMA or IRQ trouble?)" >>> >>> In my prototypes I have used snd_soc_jack for this purpose and I have >>> some questions: >>> >>> *I see snd_soc_jack is used mostly for headsets and microphones with >>> actual external mechanical connections. Strictly, in my case I propagate >>> events originated by the OMAP display driver (changes in the power >>> state), and not from external events. Some of these events are generated >>> from an actual HDMI cable connection/disconnection, though. >>> >>> *Maybe the event should be propagated by omapdss/omapdrm/drm and the >>> entity in charge of the audio policy should listen those events instead. >>> >>> *I do see SND_JACK_VIDEOOUT and SND_JACK_AVOUT types so maybe it is >>> feasible for an audio driver to report events from an AV output. >>> >>> I was wondering about how much sense does it make to you guys use a >>> snd_soc_jack in this case? >> >> How does DRM handle audio? I made a quick grep, but I see the drm >> drivers only enabling the audio in the HW, nothing else. > > I confess to not knowing too much about audio/alsa, but what does > audio driver need from hdmi? Just hotplug events? At least for the case of the ASoC HDMI audio driver (but hopefully for other audio drivers as well), it needs to detect whether an HDMI output is available, if the display's current configuration supports audio (e.g., a 1080p display configured as VGA should be reported as not supporting audio). It also needs interfaces to configure/prepare/start/stop audio. Also, of course, needs to know if the display is off/on/suspended and when transitions occur. For OMAP, omapdss provide an interface for this functionality for ALSA or any other interested user. > > From a quick look, it seems most of what the existing drm drivers are > doing is detecting if the display supports audio, and then turning > on/off the hw.. (and some infoframe stuff in some cases). Yes, it seems to me that every driver makes its own audio implementation, mainly focused on configuration. I could not find any audio common interface so that users like ALSA can take advantage of. Also, I could not see any ALSA driver using functionality provided by a drm driver. Maybe the lack of audio support in drm is because the audio users should not talk to drm directly but to a lower level component (omapdrm, omapdss?). However, today there exists video technology supports audio as well, such as DisplayPort or HDMI. Could it make more sense now to provide audio support? > > Does ASoC support 'hotplug' of audio devices? If so, maybe it makes > some sense to have some support in drm core. At least all the edid > parsing stuff to determine if the display supports audio should be > generic and not driver specific. drm already supports generic edid parsing to check for audio support. This is a good example of how an audio driver could just use that functionality to probe audio support. Ricardo > > BR, > -R > >> If there's a common generic way to handle this, we should obviously use >> that. But if we need to choose between doing something custom or doing >> it in omapdrm driver, I think we should go for drm the only solution and >> forget about audio with omapfb. >> >> Tomi >>

3 4

[alsa-devel] [PATCH] [ALSA] hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
by mengdong.lin＠intel.com 24 Aug '12

24 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2baa3ee..97307df 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2515,8 +2515,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Don't force card bias level to be updated
by Mark Brown 23 Aug '12

23 Aug '12

Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1fb6a6..53f69b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: dapm: Don't force card bias level to be updated
by Mark Brown 23 Aug '12

23 Aug '12

Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1fb6a6..53f69b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- 1.7.10.4

1 0