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[alsa-devel] ALC882 on Abit AW9D-MAX
by Dan Aloni 10 Jul '07

10 Jul '07

Hello, Yesterday I posted this bug in the bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3013 I have received no reponse yet. I'd like to cooperate on solving this. Thanks, -- Dan Aloni XIV LTD, http://www.xivstorage.com da-x (at) monatomic.org, dan (at) xiv.co.il

2 14

[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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