October 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] hda-intel via codec on optical sound
by AL13N 04 Oct '09

04 Oct '09

I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

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[alsa-devel] hda-intel via codec on optical sound
by AL13N 04 Oct '09

04 Oct '09

I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

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[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 04 Oct '09

04 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version of this patch with improved issues pointed by Takashi. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..c739374 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -147,12 +147,6 @@ struct soundscape { struct snd_wss *chip; struct snd_mpu401 *mpu; - /* - * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL - */ - spinlock_t fwlock; - unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +158,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +168,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR "sscape: Failed to allocate " + "%lu bytes for DMA\n", + size); return NULL; } } @@ -482,7 +473,8 @@ static int upload_dma_data(struct soundscape *s, */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR "sscape: DMA upload has timed out\n"); + snd_printk(KERN_ERR + "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } @@ -504,10 +496,12 @@ static int upload_dma_data(struct soundscape *s, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); + snd_printk(KERN_ERR "sscape: No response " + "from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); + snd_printk(KERN_ERR + "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -536,7 +530,7 @@ static int sscape_upload_bootblock(struct snd_card *card) ret = request_firmware(&init_fw, "scope.cod", card->dev); if (ret < 0) { - snd_printk(KERN_ERR "Error loading scope.cod"); + snd_printk(KERN_ERR "sscape: Error loading scope.cod"); return ret; } ret = upload_dma_data(sscape, init_fw->data, init_fw->size); @@ -554,7 +548,8 @@ static int sscape_upload_bootblock(struct snd_card *card) data &= 0xf; if (ret == 0 && data > 7) { - snd_printk(KERN_ERR "timeout reading firmware version\n"); + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); ret = -EAGAIN; } @@ -575,12 +570,13 @@ static int sscape_upload_microcode(struct snd_card *card, int version) err = request_firmware(&init_fw, name, card->dev); if (err < 0) { - snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", + version); return err; } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", init_fw->size >> 10); release_firmware(init_fw); @@ -750,7 +746,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); if ((inb(wss_io) & 0x80) != 0) goto _done; @@ -774,7 +769,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); /* * SoundScape successfully detected! @@ -794,38 +788,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ static int mpu401_open(struct snd_mpu401 * mpu) { - int err; - if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); - err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); + snd_printk(KERN_ERR "sscape: MIDI disabled, " + "please load firmware\n"); + return -ENODEV; } - return err; -} - -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); + return 0; } /* @@ -845,8 +814,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -993,13 +960,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1036,7 +1003,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,8 +1040,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + snd_printk(KERN_ERR + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); goto _release_dma; } strcpy(card->driver, "SoundScape"); @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "sscape: device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

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Re: [alsa-devel] Asterisk and ALSA/Jack
by Fabien COMTE 03 Oct '09

03 Oct '09

Hi, I want to use Asterisk with ALSA / Jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } pcm.jack1b { type plug device jack1b server linux slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } ctl.mixer0 { type hw card 1 } in alsa.conf : input_device=default output_device=default That works fine. 3 ) chan_alsa.so with jackd started Jackd launch command : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & my .asoundrc : pcm.!default { type plug slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } pcm.jack1b { type plug device jack1b server linux slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } ctl.mixer0 { type hw card 1 } in alsa.conf : input_device=default output_device=default It doesn't work. Asterisk outputs are : [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open failed: No such file or directory [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open failed: No such file or directory [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:272 soundcard_init: Problem opening ALSA I/O devices But aplay my_test_file.wav works fine and arecord too... Does anyone have an idea ? Thanks, Fabien

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Re: [alsa-devel] My Codec: IDT 92HD75B3X5 issues
by Munzir Taha (منذر طه) 02 Oct '09

02 Oct '09

On Yau al-Thulatha 09 Shawwal 1430 5:37:04 pm you wrote: > At Tue, 29 Sep 2009 16:24:29 +0300, > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > On Yau al-Thulatha 09 Shawwal 1430 9:24:19 am Takashi Iwai wrote: > > > At Tue, 29 Sep 2009 07:40:56 +0300, > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: > > > > > At Mon, 28 Sep 2009 06:58:46 +0300, > > > > > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > > > Hi sirs, > > > > > > These are the issues I have with my Laptop HP Pavilion dv6 1225ee > > > > > > regarding audio support. > > > > > > > > > > > > 1. No sense for headphone jack (jack_present?), when headphone is > > > > > > plugged (present_int_hp?) > > > > > > > > > > > > 2. There headphone mic jack is not working. The one next to the > > > > > > webcam is always recording even when I change the "Mic Jack" > > > > > > control from "Mic In" to "Line In" > > > > > > > > > > Make sure that you use alsa-driver-snapshot tarball instead of > > > > > 1.0.21 released version. There are many fixes since 1.0.21 > > > > > release. > > > > > > > > > > > > > > > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-dr > > > > >iver -sn apshot.tar.gz > > > > > > > > Thanks a lot Takashi, things are better now. However, when I tested > > > > the internal mic it's very noisy. I played a bit with the capture and > > > > digital controls but still compared with Windows, it's lame. > > > > > > OK, could you give alsa-info.sh output at this state? > > > Please run with --no-upload option, and attach the generated file. > > > > Attached. > > Try to run: > > % amixer -c0 set Capture cap 12dB > % amixer -c0 set "Digital Mic" 10dB > % amixer -c0 set Digital 0dB > > Any difference? No, still there is a clear shshshshshsh sound in the background. Please check the attachment for a short arecord -f cd without me speaking on the mic and still you can hear this noise very clear. Also attached is the alsa-info report at the time of recording -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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[alsa-devel] OFFTOPIC atkbd.c: Use 'setkeycodes... (Was: Re: My Codec: IDT 92HD75B3X5 issues)
by Simos Xenitellis 02 Oct '09

02 Oct '09

On Tue, Sep 29, 2009 at 7:40 AM, Munzir Taha (منذر طه) <munzirtaha(a)gmail.com> wrote: > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: >> At Mon, 28 Sep 2009 06:58:46 +0300, >> >> Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > =?utf-8?q?_=D8=B7=D9=87?=) wrote: >> > Hi sirs, >> > These are the issues I have with my Laptop HP Pavilion dv6 1225ee >> > regarding audio support. >> > >> > 1. No sense for headphone jack (jack_present?), when headphone is plugged >> > (present_int_hp?) >> > >> > 2. There headphone mic jack is not working. The one next to the webcam is >> > always recording even when I change the "Mic Jack" control from "Mic In" >> > to "Line In" >> >> Make sure that you use alsa-driver-snapshot tarball instead of 1.0.21 >> released version. There are many fixes since 1.0.21 release. > > >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sn >> apshot.tar.gz > > Thanks a lot Takashi, things are better now. However, when I tested the > internal mic it's very noisy. I played a bit with the capture and digital > controls but still compared with Windows, it's lame. > >> > 3. The mute/unmute LED doesn't change color to RED when its muted like in >> > Windows. >> >> This isn't supported yet as we have no hardware information. >> Likely one of GPIO pins corresponds to the LED on/off. You can try >> hda-verb to change the GPIO data bit. > > Ok, I will try it and report back. > >> > Touching that LED shows this in /var/log/messages: >> > >> > Sep 28 06:53:47 dv6 kernel: [ 9197.986789] atkbd.c: Unknown key released >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9197.986791] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> > Sep 28 06:53:47 dv6 kernel: [ 9198.353568] atkbd.c: Unknown key pressed >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9198.353572] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> > Sep 28 06:53:47 dv6 kernel: [ 9198.359205] atkbd.c: Unknown key released >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9198.359208] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> >> It's not about the sound driver but the input layer, as you can find. >> You can defien the scan code 0xe014 by yourself. > > You are right. What's the best component to report this bug against? kernel? Those keys are a 'HAL' thing. Until recently there has been a website by the HAL developer with detailed instructions. Search for 'HAL Quirk Site' on Google to view the site in Google cache. The website was not updated for some time (http://www.mail-archive.com/pm-utils@lists.freedesktop.org/msg01594.html) and being obsolete, it was probably taken down. Your changes would need to make it eventually to the hal-info repository, http://cgit.freedesktop.org/hal-info/ Use the Freedesktop Bugzilla, product 'hal' and component 'hal-info' when reporting, https://bugs.freedesktop.org/enter_bug.cgi?product=hal Simos

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[alsa-devel] [PATCH] Soc: wm8940: Fix check on error code form snd_soc_codec_set_cache_io
by Jonathan Cameron 02 Oct '09

02 Oct '09

Soc: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Fix for typo in commit 8d50e447d19fec64adebeef55f2b60d695435412 ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs Signed-off-by: Jonathan Cameron <jic23(a)cam.ac.uk> --- sound/soc/codecs/wm8940.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae..1ef2454 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940, codec->reg_cache = &wm8940->reg_cache; ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); - if (ret == 0) { + if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } -- 1.6.3.3

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[alsa-devel] big latency and buffering question
by Achim Herrmann 02 Oct '09

02 Oct '09

Hi all, I'm completely new to sound and ALSA development and I have to develop an application which is capturing periodically a small amount of audio data (e.g. 64 frames), doing some processing and doing playback of the result. I started with ALSA example "pcm.c" to see if ALSA can meet these requirements. During investigation i found out, that up to a choosen buffer size of 140ms, generated sine sound is fine. Below 140ms, sound is stuttering, independent of call method I'm using. For buffer size of 140ms I found out, that target buffer size is 6174 frames (at 44.1kHz sample rate), period size is 3087 frames and loop inside write_and_poll function is calculated every 70 ms and 3087 frames are written. Think this is doublebuffering then, right? But if I now choose a buffersize of 139 ms or lower, target buffer size and period size are always the same. So no double buffering anymore and sound is stuttering. So, currently I'm far away from goal. I'd like to have a buffer size of e.g. 128 frames (2ms). Thus, period size should be 64 frames and loop should be calculated every 1ms. (If necessary I would use Linux-RT) If I choose a buffersize of 2ms, sound is far away from being sine :-) - Can I use ALSA for a application described in first sentence above? - What do I have to change in pcm.c to be able to create a sine signal using a buffersize of e.g. 128 frames and a period size of 64 frames? Any hints are appreciated Thanks in advance Achim

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[alsa-devel] [PATCH] ASoC: Improve the debugfs hierarchy
by Peter Ujfalusi 02 Oct '09

02 Oct '09

Change the way the debugfs entries are created: If the codec->dev is valid, than use: debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/ if the codec->dev is NULL: debugfs/asoc/{codec->name}/ as root for the debugfs entries. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The patch applies on top of sound-2.6 topic/asoc commit 88439ac793934a47f47ad285656b63d09f5937c8 ASoC: add support for multiple cards/codecs in debugfs sound/soc/soc-core.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e4ab36d..1dec9d2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,8 +1256,12 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { char codec_root[128]; - snprintf(codec_root, sizeof(codec_root), - "%s-%s", dev_name(codec->socdev->dev), codec->name); + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); codec->debugfs_codec_root = debugfs_create_dir(codec_root, debugfs_root); -- 1.6.5.rc2

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 02 Oct '09

02 Oct '09

The following changes since commit c877c25170e2655d519b29e91d6c91d5d1a72a6f: Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Mark Brown (1): Merge remote branch 'takashi/fix/asoc' into for-2.6.32 sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++++++---------------------- 1 files changed, 49 insertions(+), 49 deletions(-)

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