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[alsa-devel] Unable to find an usable access for 'default'
by Timur Tabi 10 Jul '07

10 Jul '07

I'm developing a new ASOC driver, and I get this error message: ALSA lib pcm_plug.c:729:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' Can someone tell me what this means? FYI, I'm in the early stages of testing my driver, and before I started I knew nothing about ALSA. This is on an embedded system, so I don't know if my ALSA installation is correct. Am I missing some kind of configuration file? -- Timur Tabi Linux Kernel Developer @ Freescale

2 7

[alsa-devel] Intel hda ICH7 not worked anymore after using wine
by Xavi de Blas 09 Jul '07

09 Jul '07

Hello, I installed Wine on my Ubuntu Feisty on IBM-lenovo X60s, and after reboot, sound never worked again. I tried to activate and to hide modem on the bios. I tried to stop machine, take off battery, wait 10s, put battery again, and boot without the power cable connected... and i tried lots of things from ubuntuforums. I reinstalled feisty and sound doesn't work. I used alsa-info.sh script, here is the result: http://pastebin.ca/610317 Should the new kernel 2.6.22 help me? http://kernelnewbies.org/Linux_2_6_22#head-0d133cda430e5d18f1c72dd3cfaa6a92… Thanks

3 12

Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Takashi Iwai 09 Jul '07

09 Jul '07

At Mon, 25 Jun 2007 15:16:41 -0500, Matt Mullins wrote: > > On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > Yes, single_cmd mode is used always. > > That's bad. > > I believe we have established that. What can I do about it? *Can* I > do anything about it? What functions should I start debugging? Has > this happened with other codecs before? What documentation will I > need or want other than the Intel HDA and STAC9205 specifications? Similar problems have been seen in some laptops, too. I vaguely remember that they need some ACPI-related boot options to fix it somehow. I think it's likely irrelevant with the codec chip, but rather a problem of the controller side. > I'm definitely willing to help work on this, as not many people have > this hardware. I'm just new to the sound system, so I'm still trying > to figure out what-calls-what. Takashi

3 10

[alsa-devel] stac9200 speaker/hp not detected
by Brian Hinz 09 Jul '07

09 Jul '07

Hi, I'm trying to get the sound working on my Gateway S-7110M (a.k.a. NX100X?). Using the ini files from the windows driver, I created the following patch (applied to both the hg tree, and 1.0.14): --- a/alsa-kernel/pci/hda/patch_sigmatel.c 2007-07-03 21:59:21.000000000 -0400 +++ b/alsa-kernel/pci/hda/patch_sigmatel.c 2007-06-04 03:28:54.000000000 -0400 @@ -39,6 +39,7 @@ enum { STAC_REF, + STAC_M4, STAC_9200_MODELS }; @@ -453,12 +452,19 @@ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; +static unsigned int stac9200_M4_pin_configs[8] = { + 0x400000fe, 0x00451030, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; + static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_REF] = ref9200_pin_configs, + [STAC_M4] = stac9200_M4_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_REF] = "ref", + [STAC_M4] = "m4", }; static struct snd_pci_quirk stac9200_cfg_tbl[] = { @@ -494,6 +486,9 @@ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f5, "Dell Inspiron 1501", STAC_REF), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_M4), + /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF), however it still does not work. I can see in dmesg that the speaker and headphones are not detected, but I can't seem to figure out why: [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:1810: hda_codec: model 'm4' is selected [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x0, fmic=0x10, line=0x0, fline=0x0, cd=0x0, aux=0x0 [95006.432000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [95006.432000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) This laptop has only an internal speaker, a headphone out, and a mic in (unless there are other outs enabled when the dock is connected). The windows INI seems to be setting some gpio bits but I don't understand how to map the ini syntax into the alsa driver (or if it needs to be...) [Models] HDAUDIO\FUNC_01&VEN_8384&DEV_7690=this Default=this [CopyFiles.32] stacgui.cpl = 0x25 stlang.dll = 0x25 stacsv.exe = 0x25 [CopyFiles.64] stacui64.cpl = 0x25 stlang64.dll = 0x25 stacsv64.exe = 0x25 [LaunchProgramsAfter.32] STacSV -r [LaunchProgramsAfter.64] STacSV64 -r [HKLM\Software\Microsoft\Windows\CurrentVersion\Run.32] [HKLM\Software\Microsoft\Windows\CurrentVersion\Run.64] [DelRegKeys] HKR\Settings=Yes HKCU\Software\Sigmatel\Settings=Yes HKLM\Software\Sigmatel\Presets=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel Audio=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel C-Major Audio=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel High Definition Audio=Yes [HKLM\Software\Sigmatel\GlobalState] [HKR] CodecName = "STAC9200" [HKR\Settings] TraceFlags = dword: 0x89B3 CPL_SaveExtraPower = hex: 0 CPL_UserControlsFixedDigitalOutRate = hex: 0 InitVerbs = hex: 0x03, 0xa0, 0x43, 0x01 9200MixerCompensation1PcmMode = hex: 2 9200MixerCompensation2PcmMode = hex: 3 CodecAccessAttempts = dword:00000352 [HKR\Settings\GPIO] 03 = hex: 0x85 AntiPopGpo = hex: 3,2 [HKR\Settings\pin] [HKR\Settings\Pin\08] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\09] CfgDflt = dword: 0x00451030 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400100F0 [HKR\Settings\Pin\0E] CfgDflt = dword: 0x90110010 Action = hex: 81,09 [HKR\Settings\Pin\0F] CfgDflt = dword: 0x400100F1 [HKR\Settings\Pin\10] CfgDflt = dword: 0x02A1902E [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000F2 [HKR\Settings\Pin\12] CfgDflt = dword: 0x500000F3 [HKR\Settings\Connselector] [HKR\settings\filter\Speaker] UniqueID = hex: 0 LegacyIoctlHere = hex: 1 Pins = hex: 0E RenderConverter = hex: 2 Associations = hex: 1 DigitalVolumeControl = hex: 0 EnableSetFormatSupport = hex: 1 EnableHwSpeakerConfigurationSupport = hex: 0 EnabledPcmRenderBitsAndRates = dword: 0x000FFFFE MasterControlsIndependentMixerInputs = hex: 0 EnableVendorPcBeep = hex: 1 PcBeepVolume = dword: 0 PcBeepMute = hex: 1 EnableInputMonitor = hex: 1 RecMonitorMute = hex: 1 EnabledNonPcmBitsAndRates = dword: 0x20040 BiosInterfaceAddress = dword: 0x000DF7E0 [HKR\settings\filter\SpdifOut] UniqueID = hex: 1 LegacyIoctlHere = hex: 0 Pins = hex: 09 SpdifOutConverter = hex: 5 SpdifMaster = hex: Associations = hex: 3 EnabledPcmRenderSpdifBitsAndRates = dword: 0x000FFFFF DisableSPDIFControl = hex: 0 [HKR\Settings\filter\MuxedIn] UniqueID = hex: 2 LegacyIoctlHere = hex: 0 Pins = hex: 10 CaptureConverter = hex: 3 EnableCompositeMuxAndAdcCtrl = hex: 0 Associations = hex: 2 MicBoost = hex: 1 DisableMicBoost = hex: 0 EnabledPcmCaptureBitsAndRates = dword: 0x000FFFFF MicInRecVolume = dword: 0x000a8000 [HKR\Settings\filter\MuxedIn\MaxLvlLimiters] MicBoostLevel = dword: 0x000c0000 [HKR\Settings\filter\MuxedIn\TopoNames] [HKLM\software\SigmaTel\Apo\GFX\Speaker\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{637c490d-eee3-4c0a-973f-371958802da2}" [HKLM\software\SigmaTel\Apo\LFX\Speaker\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{62dc1a93-ae24-464c-a43e-452f824c4250}" [HKLM\software\SigmaTel\Apo\GFX\SpdifOut\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{637c490d-eee3-4c0a-973f-371958802da2}" [HKLM\software\SigmaTel\Apo\LFX\SpdifOut\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{62dc1a93-ae24-464c-a43e-452f824c4250}" [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\SysEq] Load = dword: 1 IdEffect = dword: 101 IdEffectType = dword: 1 Order = dword: 1100 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq] NumBands = dword:12 RegVersion = dword: 1 Bypass = dword: 0 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band0] idxBand = dword: 0 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 1 GroupId = dword: 0 Freq =dword: 150 GainDbx1000 = dword: -3000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band1] idxBand = dword: 1 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 300 GainDbx1000 = dword: -3000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band2] idxBand = dword: 2 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 500 GainDbx1000 = dword: -2000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band3] idxBand = dword: 3 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 750 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band4] idxBand = dword: 4 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 1000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band5] idxBand = dword: 5 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 2000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band6] idxBand = dword: 6 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 8000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band7] idxBand = dword: 7 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 12000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band8] idxBand = dword: 8 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 15000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band9] idxBand = dword: 9 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band10] idxBand = dword: 10 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band11] idxBand = dword: 11 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 2 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\Compressor] Load = dword: 1 IdEffect = dword: 5 Order = dword: 1150 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\Compressor] CompRatio= dword: 200 CompAttack= dword: 9910 CompRelease= dword: 9910 CompThreshDb= dword: -600 MakeupGainDb= dword: 200 GateRatio= dword: 600 GateAttack= dword: 9990 GateRelease= dword: 9990 GateThreshDb= dword: -9000 Flags= dword: 8 [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\Limiter] Load = dword: 1 IdEffect = dword: 6 Order = dword: 1155 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\Limiter] CompRatio= dword: 3700 CompAttack= dword: 9900 CompRelease= dword: 9900 CompThreshDb= dword: 0 MakeupGainDb= dword: 0 GateRatio= dword: 100 GateAttack= dword: 9000 GateRelease= dword: 9000 GateThreshDb= dword: -9000 Flags= dword: 10 [HKLM\SOFTWARE\SigmaTel\STSysTray\1.0\EventExpressions] Version=dword:1 [HKLM\SOFTWARE\SigmaTel\STSysTray\1.0\EventExpressions\JackPresenseD] Action1="if(JackState(13) == 1,(BypassSystemEq(1)),(BypassSystemEq(0)))" Action2="if(JackState(13) == 1,(BypassSystemComp(1)),(BypassSystemComp(0)))" [HKLM\SOFTWARE\Sigmatel\STacGUI] Config_Tab = dword: 0x8 Config_Spk = dword: 0x0 Config_Adv = dword: 0x01 [HKLM\SOFTWARE\Sigmatel\STacGUI\XML] Nconfigs = dword: 0x0 here's the output of /proc/asound/card0/codec#0: Codec: SigmaTel STAC9200 Address: 0 Vendor Id: 0x83847690 Subsystem Id: 0x100 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 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Power: 0x0 Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x0a Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x08 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Node 0x07 [Audio Selector] wcaps 0x300901: Stereo Connection: 3 0x02* 0x08 0x0a Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x400000fe: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x00451030: [Jack] SPDIF Out at Ext N/A Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x0a Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0d 0x0d] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x80 0x80] Connection: 1 0x07 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Connection: 5 0x10* 0x0f 0x0e 0x0d 0x12 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x400100f0: [N/A] Line Out at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x0b Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x90110010: [Fixed] Speaker at Int N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x400100f1: [N/A] Line Out at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x0b Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a1902e: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x0b Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x500000f2: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x13 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x500000f3: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown 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] Thanks for any assistance! -brian

2 7

[alsa-devel] Question about write asoc driver
by Roy Huang 09 Jul '07

09 Jul '07

Hi guys, I am write a codec driver under asoc frame work for ad1980 by study the sample driver wm9712.c. I find there are some control item likes "out3 switch" to control 16h:15. There are also some widget item likes "out 3 PGA" to control 24:5. How can I determine a register bit should be a control item or widget item? Thanks ahead. -Roy

2 1

Re: [alsa-devel] [Pkg-alsa-devel] [PATCH] amixer and special 'front...' channels
by Elimar Riesebieter 09 Jul '07

09 Jul '07

This isn't Ubuntu/Debian specific and should be applied upstream? On Sat, 07 Jul 2007 the mental interface of David Decotigny told: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > Hi, > > Imagine you have this kind of control: > > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Mic' > > ... and you want to set it to 'Front Mic': > > shell> amixer sset 'Input Source' 'Front Mic' > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Mic' > > This does not work ! Because amixer will think that 'Front' is a > modifier for 'Mic'. > > Attached is a patch against alsa-utils-1.0.13-1ubuntu5 that solves this. > With this patch, one can bypass the modifier parsing: one simply escapes > the parameter with one or more backslash(es), and that's all: > > shell> amixer sset 'Input Source' '\Front Mic' > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Front Mic' > > What this patch also does, is that it adds correct identification of > items. With the original version, given the following enumeration: > foo foobar > Then the parameter 'foobar' will be identified as 'foo', not 'foobar'. > The patch should solve this potential problem (not tested though). > > Bye, > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.6 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFGj/X0ld7vhusVrCERAjIZAJ9w9GLD+QBwDCaTs32hpW6i9nVMfgCfdS5w > jhipZ2jm87fq4zx4f5Q9Vsc= > =M2KP > -----END PGP SIGNATURE----- > diff -ru alsa-utils-1.0.13/amixer/amixer.1 alsa-utils-1.0.13-mine/amixer/amixer.1 > --- alsa-utils-1.0.13/amixer/amixer.1 2006-09-29 04:53:26.000000000 -0700 > +++ alsa-utils-1.0.13-mine/amixer/amixer.1 2007-07-07 13:09:04.000000000 -0700 > @@ -44,11 +44,14 @@ > The parameters \fIcap, nocap, mute, unmute, toggle\fP are used to > change capture (recording) and muting for the group specified. > > -The optional modifiers can be put as extra parameters to specify > -the stream direction or channels to apply. > -The modifiers \fIplayback\fP and \fIcapture\fP specify the stream, > -and the modifiers \fIfront, rear, center, woofer\fP are used to specify > -channels to be changed. > +The optional modifiers can be put as extra parameters to specify the > +stream direction or channels to apply. The modifiers \fIplayback\fP > +and \fIcapture\fP specify the stream, and the modifiers \fIfront, > +rear, center, woofer\fP are used to specify channels to be changed. If > +there is a confusion between the parameter and these modifiers, then > +any character in the parameter may be escaped with a backslash to > +disambiguate: '\\Front Mic' will be understood as "parameter 'Front > +Mic'", and not as "modifier 'Front' applied to 'Mic' parameter". > > A simple mixer control must be specified. Only one device can be controlled > at a time. > diff -ru alsa-utils-1.0.13/amixer/amixer.c alsa-utils-1.0.13-mine/amixer/amixer.c > --- alsa-utils-1.0.13/amixer/amixer.c 2007-07-07 13:03:39.000000000 -0700 > +++ alsa-utils-1.0.13-mine/amixer/amixer.c 2007-07-07 12:59:19.000000000 -0700 > @@ -1333,6 +1333,16 @@ > return def; > } > > +/** Transform '\a\b c \de' into 'ab c de' */ > +static void unescape(char * arg) > +{ > + char * c; > + for (c = arg ; arg && *arg ; arg ++) > + if (*arg != '\\') > + *(c++) = *arg; > + *c = '\0'; > +} > + > static int get_enum_item_index(snd_mixer_elem_t *elem, char **ptrp) > { > char *ptr = *ptrp; > @@ -1347,7 +1357,7 @@ > if (snd_mixer_selem_get_enum_item_name(elem, i, sizeof(name)-1, name) < 0) > continue; > len = strlen(name); > - if (! strncmp(name, ptr, len)) { > + if (!strncmp(name, ptr, len) && (ptr[len] == '\0')) { > if (! ptr[len] || ptr[len] == ',' || ptr[len] == '\n') { > ptr += len; > *ptrp = ptr; > @@ -1428,6 +1438,7 @@ > dir = dir_mask(&ptr, dir); > if (*ptr == '\0') > continue; > + unescape(ptr); > multi = (strchr(ptr, ',') != NULL); > optr = ptr; > for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { > _______________________________________________ > Pkg-alsa-devel mailing list > Pkg-alsa-devel(a)lists.alioth.debian.org > http://lists.alioth.debian.org/mailman/listinfo/pkg-alsa-devel -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

2 1

[alsa-devel] [PATCH] usb-audio: another Logitech camera/microphone ID match
by Daniel Drake 09 Jul '07

09 Jul '07

From: George Shapovalov <george(a)gentoo.org> Another case of ALSA ticket #3040: we need an explicit USB ID match for another Logitech camera/microphone device. Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -57,6 +57,15 @@ USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, + .idProduct = 0x08ae, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL

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[alsa-devel] M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 09 Jul '07

09 Jul '07

Hi all ! I don't know if this is already known (?), but the M-Audio Audiophile USB seem to be possible to run in native S16_LE mode if initialized with device_setup=0x01 (possibly with 0x11 also, have not tested that). This is at least the case with my rather new box (bought a couple of weeks ago). Patch for usb/usbaudio.c (alsa-driver-1.0.14): static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp) { switch (chip->usb_id) { case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */ if (fp->endpoint & USB_DIR_IN) return 1; break; case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + // Is small endian if setup with 0x01 ! + if (device_setup[chip->index] != 0x01) return 1; break; } return 0; } /Håkan

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[alsa-devel] [PATCH] ALSA AICA sound on SEGA Dreamcast - fix behaviour in poor resource conditions
by Adrian McMenamin 09 Jul '07

09 Jul '07

Patch against code in alsa-kernel - tested with patched up 2.6.21-rc7 and working well Submitted by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff -ruN aicaold/aica.c aicanew/aica.c --- aicaold/aica.c 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.c 2007-07-05 23:02:02.000000000 +0100 @@ -65,10 +65,6 @@ /* Use workqueue */ -static struct spu_work_holder { - struct work_struct spu_dma_work; - void *sspointer; -} spu_working; static struct workqueue_struct *aica_queue; @@ -252,18 +248,15 @@ static void run_spu_dma(struct work_struct *work) { int buffer_size; - struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_card_aica *dreamcastcard; - struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); - substream = holder-> sspointer; - dreamcastcard = substream->pcm->private_data; - runtime = substream->runtime; + dreamcastcard = container_of(work, struct snd_card_aica, spu_dma_work); + runtime = dreamcastcard->substream->runtime; if (unlikely(dreamcastcard->dma_check == 0)) { buffer_size = frames_to_bytes(runtime, runtime->buffer_size); if (runtime->channels > 1) dreamcastcard->channel->flags |= 0x01; - aica_dma_transfer(runtime->channels, buffer_size, substream); + aica_dma_transfer(runtime->channels, buffer_size, dreamcastcard->substream); startup_aica(dreamcastcard); dreamcastcard->clicks = buffer_size / (AICA_PERIOD_SIZE * runtime->channels); @@ -271,7 +264,7 @@ } else { aica_dma_transfer(runtime->channels, AICA_PERIOD_SIZE * runtime->channels, - substream); + dreamcastcard->substream); snd_pcm_period_elapsed(dreamcastcard->substream); dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) @@ -307,7 +300,8 @@ dreamcastcard->current_period = play_period; if (unlikely(dreamcastcard->dma_check == 0)) dreamcastcard->dma_check = 1; - queue_work(aica_queue, &(spu_working.spu_dma_work)); + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); + } static void spu_begin_dma(struct snd_pcm_substream *substream) @@ -317,10 +311,8 @@ struct snd_pcm_runtime *runtime; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; - /* Use queue to do the heavy lifting */ - spu_working.sspointer = substream; - INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); - queue_work(aica_queue, &(spu_working.spu_dma_work)); + //get the queue to do the work + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); /* Timer may already be running */ if (unlikely(dreamcastcard->timer.data)) { mod_timer(&dreamcastcard->timer, jiffies + 4); @@ -366,7 +358,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,17 +396,10 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; - case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); - break; default: return -EINVAL; } @@ -610,6 +597,8 @@ strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); strcpy(dreamcastcard->card->longname, "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + //start the worker thread + INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) diff -ruN aicaold/aica.h aicanew/aica.h --- aicaold/aica.h 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.h 2007-07-05 23:02:02.000000000 +0100 @@ -69,6 +69,7 @@ }; struct snd_card_aica { + struct work_struct spu_dma_work; struct snd_card *card; struct aica_channel *channel; struct snd_pcm_substream *substream;

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[alsa-devel] [HDA Intel/STAC92xx] mic not working (retasking?)...
by Daniel J Blueman 08 Jul '07

08 Jul '07

With both 2.6.20 and 2.6.22 kernels on Ubuntu 7.04 on my Sony Vaio SZ240, I'm unable to get my mic connector working at any cost. Well, almost...reading from /dev/dsp, I do see a minute change from 127 for 126 or 128 if I tap my external mic. This would imply the jack-retasking feature of this darn STAC chipset, right? How can I change the jack retasking with the Intel HDA drivers or get further insight into this problem? Thanks in advance, Daniel --- [1] $ lspci -s0:1b.0 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ amixer info Card default 'Intel'/'HDA Intel at 0xde400000 irq 22' Mixer name : 'SigmaTel CXD9872RD/K' Components : 'HDA:83847661 HDA:14f12bfa' Controls : 6 Simple ctrls : 5 $ amixer controls numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' numid=6,iface=MIXER,name='PCM Playback Volume' numid=5,iface=MIXER,name='Capture Source' numid=4,iface=MIXER,name='Capture Switch' numid=3,iface=MIXER,name='Capture Volume' $ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'PCM',0 Simple mixer control 'Line',0 Simple mixer control 'Mic',0 Simple mixer control 'Capture',0 $ amixer contents numid=2,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=127,step=0 : values=66,66 numid=6,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=230,230 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=5,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Line' ; Item #1 'Mic' : values=1 numid=4,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=15,step=0 : values=15,15 | dBscale-min=0.00dB,step=1.50dB,mute=0 -- Daniel J Blueman

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