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[alsa-devel] HDA regression on Fujitsu S7020 laptop (ALC260 codec)
by Jonathan Woithe 07 Mar '13

07 Mar '13

Having been running kernel 3.3.5 for a while, I decided to upgrade to 3.7.1 tonight. Unfortunately there have been a number of HDA-related regressions between these two kernels when running on a Fujitsu S7020 laptop which are clearly related to the HDA rewrite which has been going on recently. This laptop utilises the Realtek ALC260 codec. Testing was initially done with alsa-lib 1.0.24 and alsa-utils 1.0.24. I upgraded to version 1.0.26 of both but the behaviour remained the same under 3.3.5 and 3.7.1. Under 3.3.5, sound works perfectly. Under 3.7.1 there are a number of regressions: - there is no alsamixer control for the internal speaker. I have a "PCM" volume control which appears to do nothing. Previously there was a speaker control, the PCM control affected every output (from memory), as did the master volume. - no audio comes out of the internal speaker regardless of the "auto mute" setting. - there are no pin mode (re-tasking) controls for the mic or headphone jack in alsamixer. On this laptop the mic jack should have various bias options as well as line-in. The headphone jack should include headphone (aka output amplifier on), line-out (output amplifier off), line-in and mic options. The second ADC does show up in alsamixer as a second capture device, but probably due to the lack of retasking support it is not possible to select the headphone jack as an input option. Audio does come out of the headphone jack. However, as noted above it is not possible to switch this from headphone to line-out. The volume of the headphone output is affected by the "master" and "headphone" volume controls. The inability to set the pin modes is a showstopper for me - I need to be able to do this and thus will be stuck on older kernels until this is fixed. At the end of this post I have included the /proc/asound/card0/codec#0 output under 3.3.5 (which works well) and 3.7.1 (which is broken as described above). What needs to be done to restore the functionality for the S7020 that has been working for years? It seems that some form of model quirk needs to be ported into the new HDA framework from earlier kernels, but since this sort of thing appears to have been removed with the rewrite (at least for realtek codecs) it's difficult to see where to begin. Regards jonathan Output from /proc/asound/card0/codec#0 under kernel 3.3.5: Codec: Realtek ALC260 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0260 Subsystem Id: 0x10cf0000 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC260 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x04 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC260 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 7 0x12* 0x13 0x14 0x15 0x16 0x0f 0x10 Node 0x05 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=1, device=0 Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC260 Analog", type="Audio", device=2 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 8 0x12* 0x13 0x14 0x15 0x16 0x07 0x0f 0x10 Node 0x06 [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x19 Node 0x07 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Mic/Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic/Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x03 0x03] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x12 0x13 0x14 0x15 0x16 0x17 0x0f 0x10 Node 0x08 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x27 0x27] Connection: 2 0x02 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x02 0x07 Node 0x0a [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=0 Amp-Out vals: [0x00] Connection: 2 0x02 0x07 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x10 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0xe4011110: [Both] Line Out at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x09 Node 0x11 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x0a Node 0x12 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Mic/Line Jack Mode", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x03a11820: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x13 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x24811121: [Jack] Line In at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Jack Mode", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x16 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x88331122: [Fixed] CD at Ext Drive Bar Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x17 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0xb7931123: [Fixed] Aux at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x18 [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x19 [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x1a [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=13 Node 0x1b [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0 Output from /proc/asound/card0/codec#0 under kernel 3.7.1: Codec: Realtek ALC260 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0260 Subsystem Id: 0x10cf0000 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC260 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x04 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC260 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 7 0x12* 0x13 0x14 0x15 0x16 0x0f 0x10 Node 0x05 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=1, device=0 Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC260 Analog", type="Audio", device=2 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 8 0x12* 0x13 0x14 0x15 0x16 0x07 0x0f 0x10 Node 0x06 [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x19 Node 0x07 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x03 0x03] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x12 0x13 0x14 0x15 0x16 0x17 0x0f 0x10 Node 0x08 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x27 0x27] Connection: 2 0x02 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x33 0x33] Connection: 2 0x02 0x07 Node 0x0a [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x80] Amp-Out caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=0 Amp-Out vals: [0x23] Connection: 2 0x02 0x07 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x10 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0xe4011110: [Both] Line Out at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x09 Node 0x11 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0a Node 0x12 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Mic Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x03a11820: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Connection: 1 0x0b Node 0x13 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Line Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x24811121: [Jack] Line In at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 1 0x0d Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x16 [Pin Complex] wcaps 0x400001: Stereo Control: name="CD Phantom Jack", index=0, device=0 Pincap 0x00000020: IN Pin Default 0x88331122: [Fixed] CD at Ext Drive Bar Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x17 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0xb7931123: [Fixed] Aux at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x18 [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x19 [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x1a [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=13 Node 0x1b [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0

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[alsa-devel] [PATCH] ALSA: vmaster: Fix slave change notification
by Takashi Iwai 07 Mar '13

07 Mar '13

When a value of a vmaster slave control is changed, the ctl change notification is sometimes ignored. This happens when the master control overrides, e.g. when the corresponding master control is muted. The reason is that slave_put() returns the value of the actual slave put callback, and it doesn't reflect the virtual slave value change. This patch fixes the function just to return 1 whenever a slave value is changed. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/vmaster.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 8575861..0097f36 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol, } if (!changed) return 0; - return slave_put_val(slave, ucontrol); + err = slave_put_val(slave, ucontrol); + if (err < 0) + return err; + return 1; } static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, -- 1.8.1.4

1 0

[alsa-devel] ASoC updates for v3.9
by Mark Brown 07 Mar '13

07 Mar '13

The following changes since commit 6dbe51c251a327e012439c4772097a13df43c5b8: Linux 3.9-rc1 (2013-03-03 15:11:05 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.9-rc1 for you to fetch changes up to f73c06dc2cbf1ddf1c92f05f9843f923c8c69529: Merge remote-tracking branch 'asoc/fix/wm8960' into tmp (2013-03-07 14:29:43 +0800) ---------------------------------------------------------------- ASoC: Updates for v3.9 A few driver fixes, none of them terribly dramatic. ---------------------------------------------------------------- Axel Lin (1): ASoC: wm8350: Use jiffies rather than msecs in schedule_delayed_work() Charles Keepax (2): ASoC: wm5102: Correct OUT2 volume and switch names ASoC: wm5110: Correct OUT2/3 volume and switch names Mark Brown (7): ASoC: wm8960: Correct register 0 and 1 defaults ASoC: wm8960: Fix ADC power bits ASoC: wm5102: Apply a SYSCLK patch for later revs Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/tegra' into tmp Merge remote-tracking branch 'asoc/fix/wm8350' into tmp Merge remote-tracking branch 'asoc/fix/wm8960' into tmp Stephen Warren (1): ASoC: tegra: fix I2S bit count mask sound/soc/codecs/wm5102.c | 15 +++++++++++++-- sound/soc/codecs/wm5110.c | 16 ++++++++-------- sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8960.c | 8 ++++---- sound/soc/tegra/tegra20_i2s.h | 2 +- sound/soc/tegra/tegra30_i2s.h | 2 +- 6 files changed, 29 insertions(+), 18 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Avoid division by zero in dspxfr_one_seg()
by Xi Wang 07 Mar '13

07 Mar '13

Move the zero check `hda_frame_size_words == 0' before the modulus `buffer_size_words % hda_frame_size_words'. Also remove the redundant null check `buffer_addx == NULL'. Signed-off-by: Xi Wang <xi.wang(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index db02c1e..eefc456 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec, hda_frame_size_words = ((sample_rate_div == 0) ? 0 : (num_chans * sample_rate_mul / sample_rate_div)); + if (hda_frame_size_words == 0) { + snd_printdd(KERN_ERR "frmsz zero\n"); + return -EINVAL; + } + buffer_size_words = min(buffer_size_words, (unsigned int)(UC_RANGE(chip_addx, 1) ? 65536 : 32768)); @@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, chip_addx, hda_frame_size_words, num_chans, sample_rate_mul, sample_rate_div, buffer_size_words); - if ((buffer_addx == NULL) || (hda_frame_size_words == 0) || - (buffer_size_words < hda_frame_size_words)) { + if (buffer_size_words < hda_frame_size_words) { snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n"); return -EINVAL; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - check NULL pointer when creating SPDIF PCM switch
by mengdong.lin＠intel.com 07 Mar '13

07 Mar '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If the new control cannot be created, this function will return to avoid snd_hda_ctl_add dereferencing a NULL control pointer. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3dc6566..97c68dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3433,11 +3433,16 @@ static struct snd_kcontrol_new spdif_share_sw = { int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { + struct snd_kcontrol *kctl; + if (!mout->dig_out_nid) return 0; + + kctl = snd_ctl_new1(&spdif_share_sw, mout); + if (!kctl) + return -ENOMEM; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, mout->dig_out_nid, - snd_ctl_new1(&spdif_share_sw, mout)); + return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl); } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: let device core setup the default pin configuration
by Thomas Abraham 07 Mar '13

07 Mar '13

With device core now able to setup the default pin configuration, the pin configuration code based on the deprecated Samsung specific gpio bindings in i2s driver is removed. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- .../devicetree/bindings/sound/samsung-i2s.txt | 30 +-------- sound/soc/samsung/i2s.c | 67 +------------------ 2 files changed, 8 insertions(+), 89 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 3070046..1db65e5 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -20,23 +20,11 @@ Optional SoC Specific Properties: then this flag is enabled. - samsung,idma-addr: Internal DMA register base address of the audio sub system(used in secondary sound source). - -Required Board Specific Properties: - -- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK - interface lines. The format of the gpio specifier depends on the gpio - controller. - The syntax of samsung gpio specifier is - <[phandle of the gpio controller node] - [pin number within the gpio controller] - [mux function] - [flags and pull up/down] - [drive strength]> +- pinctrl-0: Should specify pin control groups used for this controller. +- pinctrl-names: Should contain only one value - "default". Example: -- SoC Specific Portion: - i2s@03830000 { compatible = "samsung,i2s-v5"; reg = <0x03830000 0x100>; @@ -48,16 +36,6 @@ i2s@03830000 { samsung,supports-rstclr; samsung,supports-secdai; samsung,idma-addr = <0x03000000>; -}; - -- Board Specific Portion: - -i2s@03830000 { - gpios = <&gpz 0 2 0 0>, /* I2S_0_SCLK */ - <&gpz 1 2 0 0>, /* I2S_0_CDCLK */ - <&gpz 2 2 0 0>, /* I2S_0_LRCK */ - <&gpz 3 2 0 0>, /* I2S_0_SDI */ - <&gpz 4 2 0 0>, /* I2S_0_SDO[1] */ - <&gpz 5 2 0 0>, /* I2S_0_SDO[2] */ - <&gpz 6 2 0 0>; /* I2S_0_SDO[3] */ + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e3..f982a83 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -16,7 +16,6 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -79,7 +78,6 @@ struct i2s_dai { u32 suspend_i2smod; u32 suspend_i2scon; u32 suspend_i2spsr; - unsigned long gpios[7]; /* i2s gpio line numbers */ }; /* Lock for cross i/f checks */ @@ -1009,52 +1007,6 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } -#ifdef CONFIG_OF -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) -{ - struct device *dev = &i2s->pdev->dev; - int index, gpio, ret; - - for (index = 0; index < 7; index++) { - gpio = of_get_gpio(dev->of_node, index); - if (!gpio_is_valid(gpio)) { - dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); - goto free_gpio; - } - - ret = gpio_request(gpio, dev_name(dev)); - if (ret) { - dev_err(dev, "gpio [%d] request failed\n", gpio); - goto free_gpio; - } - i2s->gpios[index] = gpio; - } - return 0; - -free_gpio: - while (--index >= 0) - gpio_free(i2s->gpios[index]); - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) -{ - unsigned int index; - for (index = 0; index < 7; index++) - gpio_free(i2s->gpios[index]); -} -#else -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) -{ - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) -{ -} - -#endif - static const struct of_device_id exynos_i2s_match[]; static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) @@ -1223,18 +1175,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->sec_dai = sec_dai; } - if (np) { - if (samsung_i2s_parse_dt_gpio(pri_dai)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } - } else { - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } + if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; } snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); @@ -1259,9 +1203,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; - if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) - samsung_i2s_dt_gpio_free(i2s->pri_dai); - if (other) { other->pri_dai = NULL; other->sec_dai = NULL; -- 1.6.6.rc2

3 2

[alsa-devel] [PATCH] ALSA: hda - check NULL pointer when creating SPDIF controls
by mengdong.lin＠intel.com 07 Mar '13

07 Mar '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If the SPDIF control array cannot be reallocated, the function will return to avoid dereferencing a NULL pointer. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..3dc6566 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3334,6 +3334,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, return -EBUSY; } spdif = snd_array_new(&codec->spdif_out); + if (!spdif) + return -ENOMEM; for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); if (!kctl) -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2] ALSA: ASoC: cs4271: convert to direct regmap API usage
by Daniel Mack 07 Mar '13

07 Mar '13

By using the regmap API directly, we can make use of the .write_flag_mask for SPI, which allows us to drop the strange register hacks that were necessary so far. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v2: add a read_flag_mask to SPI regmap config sound/soc/codecs/cs4271.c | 157 +++++++++++++++++++++++++++------------------- 1 file changed, 91 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a41..6d77a1e 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -42,14 +42,14 @@ * High byte represents SPI chip address (0x10) + write command (0) * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +144,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +214,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +231,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +256,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +345,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +382,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -389,6 +393,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; @@ -398,10 +403,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -463,25 +471,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +558,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +595,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +622,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +654,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +676,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); -- 1.8.1.4

1 0

[alsa-devel] [PATCH 1/2] ALSA: ASoC: ak4104: convert to direct regmap API usage
by Daniel Mack 07 Mar '13

07 Mar '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak4104.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 6f6c335..58f390d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; int ret; @@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, - val); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, + val); if (ret < 0) return ret; @@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; - snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val); + regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val); val = 0; @@ -132,7 +134,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val); + return regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); } static const struct snd_soc_dai_ops ak4101_dai_ops = { @@ -160,20 +162,17 @@ static int ak4104_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4104->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) - return ret; /* set power-up and non-reset bits */ - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); if (ret < 0) return ret; /* enable transmitter */ - ret = snd_soc_update_bits(codec, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); if (ret < 0) return ret; @@ -182,8 +181,10 @@ static int ak4104_probe(struct snd_soc_codec *codec) static int ak4104_remove(struct snd_soc_codec *codec) { - snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); return 0; } -- 1.8.1.4

2 2

[alsa-devel] Creative CA0132 1102:0012 (rev 01)
by Stefan de Konink 07 Mar '13

07 Mar '13

I've been the proud owner of a Creative PCIe soundcard, sadly getting anything else but static noise out of it while playing a file seems impossible. After the device is closed the noise stops again. The mixer does function in all aspects, muting, volume etc. HDAAnalyzer shows that I should get output from CA0132 Analog, I have validated this by manipulating the PIN Widget Controls. Is there any documentation available on this specific device? It seemed that some DSP work was done recently by Creative. I wouldn't mind to do some hacking myself, but it is hard for me to believe that a driver was posted in the 1.0.25 release that didn't have any normal functioning analogue output. Stefan 03:00.0 0403: 1102:0012 (rev 01) Subsystem: 1102:0013 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at f7104000 (64-bit, non-prefetchable) [size=16K] Region 2: Memory at f7100000 (64-bit, non-prefetchable) [size=16K] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Enable- Count=1/1 Maskable+ 64bit+ Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [70] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 512 bytes, PhantFunc 0, Latency L0s <1us, L1 <8us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <512ns, L1 <16us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM L0s Enabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis+, LTR-, OBFF Not Supported DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis-, LTR-, OBFF Disabled LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap+ CGenEn- ChkCap+ ChkEn- Capabilities: [140 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable- ID=1 ArbSelect=Fixed TC/VC=00 Status: NegoPending- InProgress- Capabilities: [170 v1] Device Serial Number 00-00-00-00-00-00-00-00 Capabilities: [180 v1] Power Budgeting <?> Kernel driver in use: snd_hda_intel snd 49214 11 snd_hwdep,snd_timer,snd_pcm,snd_hda_codec,snd_hda_intel,snd_hda_codec_ca0132

1 0

[alsa-devel] [PATCH] ALSA: ASoC: cs4271: convert to direct regmap API usage
by Daniel Mack 06 Mar '13

06 Mar '13

By using the regmap API directly, we can make use of the .write_flag_mask for SPI, which allows us to drop the strange register hacks that were necessary so far. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- I could only test this on I2C, so if anyone with appropriate hardware would give it a try on SPI, that would be good. Thanks, Daniel sound/soc/codecs/cs4271.c | 156 ++++++++++++++++++++++++++-------------------- 1 file changed, 90 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a41..b6d670f 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -42,14 +42,14 @@ * High byte represents SPI chip address (0x10) + write command (0) * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +144,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +214,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +231,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +256,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +345,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +382,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -389,6 +393,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; @@ -398,10 +403,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -463,25 +471,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +558,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +595,23 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +621,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +653,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +675,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); -- 1.8.1.4

1 1

[alsa-devel] [PATCH] Make snd_BUG_ON() always evaluate and return the conditional expression.
by Christine Spang 06 Mar '13

06 Mar '13

Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG is set leads to frequent bugs, since other similar macros in the kernel have different behavior. Let's make snd_BUG_ON() act like those macros so it will stop being accidentally misused. Signed-off-by: Christine Spang <christine.spang(a)oracle.com> --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 12 +++++------- include/sound/core.h | 24 ++++++++--------------- 2 files changed, 13 insertions(+), 23 deletions(-) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index bd6fee2..06741e9 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -6164,14 +6164,12 @@ struct _snd_pcm_runtime { <para> The macro takes an conditional expression to evaluate. - When <constant>CONFIG_SND_DEBUG</constant>, is set, the - expression is actually evaluated. If it's non-zero, it shows - the warning message such as + When <constant>CONFIG_SND_DEBUG</constant>, is set, if the + expression is non-zero, it shows the warning message such as <computeroutput>BUG? (xxx)</computeroutput> - normally followed by stack trace. It returns the evaluated - value. - When no <constant>CONFIG_SND_DEBUG</constant> is set, this - macro always returns zero. + normally followed by stack trace. + + In both cases it returns the evaluated value. </para> </section> diff --git a/include/sound/core.h b/include/sound/core.h index 7cede2d..a63680b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -379,18 +379,10 @@ void __snd_printk(unsigned int level, const char *file, int line, * snd_BUG_ON - debugging check macro * @cond: condition to evaluate * - * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition, - * and call WARN() and returns the value if it's non-zero. - * - * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given - * condition is ignored. - * - * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n. - * Thus, don't put any statement that influences on the code behavior, - * such as pre/post increment, to the argument of this macro. - * If you want to evaluate and give a warning, use standard WARN_ON(). + * Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set, + * otherwise just evaluates the conditional and returns the value. */ -#define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond)) +#define snd_BUG_ON(cond) WARN_ON((cond)) #else /* !CONFIG_SND_DEBUG */ @@ -400,11 +392,11 @@ __printf(2, 3) static inline void _snd_printd(int level, const char *format, ...) {} #define snd_BUG() do { } while (0) -static inline int __snd_bug_on(int cond) -{ - return 0; -} -#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */ + +#define snd_BUG_ON(condition) ({ \ + int __ret_warn_on = !!(condition); \ + unlikely(__ret_warn_on); \ +}) #endif /* CONFIG_SND_DEBUG */ -- 1.8.2.rc0

3 5

Re: [alsa-devel] [PATCH] ASoC: samsung: let device core setup the default pin configuration
by Mark Brown 06 Mar '13

06 Mar '13

On Wed, Mar 06, 2013 at 12:42:09PM +0100, Heiko Stübner wrote: > What is going away is the deprecated non-pinctrl way to setup the pins on dt > platforms. OK, can we have a changelog improvement please? I didn't read the patch properly because the changelog said it would break my primary development system.

2 1

Re: [alsa-devel] ALSA: usb: Work around CM6631 sample
by Demian Martin 06 Mar '13

06 Mar '13

Date: Tue, 5 Mar 2013 23:03:03 +0100 From: Torstein Hegge <hegge(a)resisty.net> To: alsa-devel(a)alsa-project.org Subject: [alsa-devel] [PATCH v3] ALSA: usb: Work around CM6631 sample rate change bug Message-ID: <20130305220303.GD20417(a)pvv.ntnu.no> Content-Type: text/plain; charset=utf-8 The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all current C-Media UAC V2.0 devices, including CM6610, CM6620 and CM6631, but has only been tested with CM6631. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> I met with the CMedia team late last year. They told me that the chips are different and they were showing me a CM6631a that has some key differences from the CM6631. The two are pin compatible. I got the impression that the driver requirements are different. I asked about Linux support and what the problem was. First, on UAC2 Their software wizard dug deep into the problem with their CM6631 and ALSA. He concluded that there is a sequence problem between setting sample rate and sending the "play' command. Something about UAC1 method not being the same as UAC2. He indicated that the ALSA driver handles this differently from the OSX and their Windows drivers. I may well have this description confused. In any case he has a revised firmware for the new CM6631A chip that addresses this in ALSA. I asked them to contribute the change but they seemed to get distracted with other projects. Needless to say for them the volume on these chips has been small. I just requested that they contribute to this effort again. I don't know if it will happen. Demian

2 1

[alsa-devel] [PATCH 4/4] ASoC: arizona: Optimise FLL loop gains
by Mark Brown 06 Mar '13

06 Mar '13

For optimal performance the FLL loop gain should be adjusted depending on the frequency of the input clock for the loop. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 43 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 39 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 41c116f..29dd641 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1175,6 +1175,16 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static struct { + unsigned int min; + unsigned int max; + u16 gain; +} fll_gains[] = { + { 0, 256000, 0 }, + { 256000, 1000000, 2 }, + { 1000000, 13500000, 4 }, +}; + struct arizona_fll_cfg { int n; int theta; @@ -1182,6 +1192,7 @@ struct arizona_fll_cfg { int refdiv; int outdiv; int fratio; + int gain; }; static int arizona_calc_fll(struct arizona_fll *fll, @@ -1241,6 +1252,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } + for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { + if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { + cfg->gain = fll_gains[i].gain; + break; + } + } + if (i == ARRAY_SIZE(fll_gains)) { + arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", + Fref); + return -EINVAL; + } + cfg->n = target / (ratio * Fref); if (target % (ratio * Fref)) { @@ -1268,13 +1291,15 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain); return 0; } static void arizona_apply_fll(struct arizona *arizona, unsigned int base, - struct arizona_fll_cfg *cfg, int source) + struct arizona_fll_cfg *cfg, int source, + bool sync) { regmap_update_bits(arizona->regmap, base + 3, ARIZONA_FLL1_THETA_MASK, cfg->theta); @@ -1289,6 +1314,15 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + if (sync) + regmap_update_bits(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + else + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits(arizona->regmap, base + 2, ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, ARIZONA_FLL1_CTRL_UPD | cfg->n); @@ -1326,17 +1360,18 @@ static void arizona_enable_fll(struct arizona_fll *fll, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); + arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + false); if (fll->sync_src >= 0) arizona_apply_fll(arizona, fll->base + 0x10, sync, - fll->sync_src); + fll->sync_src, true); } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, sync, - fll->sync_src); + fll->sync_src, false); } else { arizona_fll_err(fll, "No clocks provided\n"); return; -- 1.7.10.4

1 0

[alsa-devel] [PATCH 3/4] ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies
by Mark Brown 06 Mar '13

06 Mar '13

If we are using a high freqency SYNCCLK then increasing the bandwidth of the synchroniser improves performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 81da6ef..41c116f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1342,6 +1342,17 @@ static void arizona_enable_fll(struct arizona_fll *fll, return; } + /* + * Increase the bandwidth if we're not using a low frequency + * sync source. + */ + if (fll->sync_src >= 0 && fll->sync_freq > 100000) + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); + else + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); -- 1.7.10.4

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[alsa-devel] [PATCH 2/4] mfd: wm5102: Map in additional FLL control registers
by Mark Brown 06 Mar '13

06 Mar '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5102-tables.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index f3c2e9a..768fd42 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -291,12 +291,14 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000176, 0x0000 }, /* R374 - FLL1 Control 6 */ { 0x00000177, 0x0181 }, /* R375 - FLL1 Loop Filter Test 1 */ { 0x00000178, 0x0000 }, /* R376 - FLL1 NCO Test 0 */ + { 0x00000179, 0x0000 }, /* R377 - FLL1 Control 7 */ { 0x00000181, 0x0000 }, /* R385 - FLL1 Synchroniser 1 */ { 0x00000182, 0x0000 }, /* R386 - FLL1 Synchroniser 2 */ { 0x00000183, 0x0000 }, /* R387 - FLL1 Synchroniser 3 */ { 0x00000184, 0x0000 }, /* R388 - FLL1 Synchroniser 4 */ { 0x00000185, 0x0000 }, /* R389 - FLL1 Synchroniser 5 */ { 0x00000186, 0x0000 }, /* R390 - FLL1 Synchroniser 6 */ + { 0x00000187, 0x0001 }, /* R391 - FLL1 Synchroniser 7 */ { 0x00000189, 0x0000 }, /* R393 - FLL1 Spread Spectrum */ { 0x0000018A, 0x0004 }, /* R394 - FLL1 GPIO Clock */ { 0x00000191, 0x0000 }, /* R401 - FLL2 Control 1 */ @@ -307,12 +309,14 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000196, 0x0000 }, /* R406 - FLL2 Control 6 */ { 0x00000197, 0x0000 }, /* R407 - FLL2 Loop Filter Test 1 */ { 0x00000198, 0x0000 }, /* R408 - FLL2 NCO Test 0 */ + { 0x00000199, 0x0000 }, /* R409 - FLL2 Control 7 */ { 0x000001A1, 0x0000 }, /* R417 - FLL2 Synchroniser 1 */ { 0x000001A2, 0x0000 }, /* R418 - FLL2 Synchroniser 2 */ { 0x000001A3, 0x0000 }, /* R419 - FLL2 Synchroniser 3 */ { 0x000001A4, 0x0000 }, /* R420 - FLL2 Synchroniser 4 */ { 0x000001A5, 0x0000 }, /* R421 - FLL2 Synchroniser 5 */ { 0x000001A6, 0x0000 }, /* R422 - FLL2 Synchroniser 6 */ + { 0x000001A7, 0x0001 }, /* R423 - FLL2 Synchroniser 7 */ { 0x000001A9, 0x0000 }, /* R425 - FLL2 Spread Spectrum */ { 0x000001AA, 0x0004 }, /* R426 - FLL2 GPIO Clock */ { 0x00000200, 0x0006 }, /* R512 - Mic Charge Pump 1 */ @@ -1056,12 +1060,14 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_CONTROL_6: case ARIZONA_FLL1_LOOP_FILTER_TEST_1: case ARIZONA_FLL1_NCO_TEST_0: + case ARIZONA_FLL1_CONTROL_7: case ARIZONA_FLL1_SYNCHRONISER_1: case ARIZONA_FLL1_SYNCHRONISER_2: case ARIZONA_FLL1_SYNCHRONISER_3: case ARIZONA_FLL1_SYNCHRONISER_4: case ARIZONA_FLL1_SYNCHRONISER_5: case ARIZONA_FLL1_SYNCHRONISER_6: + case ARIZONA_FLL1_SYNCHRONISER_7: case ARIZONA_FLL1_SPREAD_SPECTRUM: case ARIZONA_FLL1_GPIO_CLOCK: case ARIZONA_FLL2_CONTROL_1: @@ -1072,12 +1078,14 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_CONTROL_6: case ARIZONA_FLL2_LOOP_FILTER_TEST_1: case ARIZONA_FLL2_NCO_TEST_0: + case ARIZONA_FLL2_CONTROL_7: case ARIZONA_FLL2_SYNCHRONISER_1: case ARIZONA_FLL2_SYNCHRONISER_2: case ARIZONA_FLL2_SYNCHRONISER_3: case ARIZONA_FLL2_SYNCHRONISER_4: case ARIZONA_FLL2_SYNCHRONISER_5: case ARIZONA_FLL2_SYNCHRONISER_6: + case ARIZONA_FLL2_SYNCHRONISER_7: case ARIZONA_FLL2_SPREAD_SPECTRUM: case ARIZONA_FLL2_GPIO_CLOCK: case ARIZONA_MIC_CHARGE_PUMP_1: -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Apply a SYSCLK patch for later revs
by Mark Brown 06 Mar '13

06 Mar '13

Evaluation has identified some performance improvements to the device. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index deb1097..cef288c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -573,6 +573,13 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = { { 0x025e, 0x0112 }, }; +static const struct reg_default wm5102_sysclk_revb_patch[] = { + { 0x3081, 0x08FE }, + { 0x3083, 0x00ED }, + { 0x30C1, 0x08FE }, + { 0x30C3, 0x00ED }, +}; + static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, patch = wm5102_sysclk_reva_patch; patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); break; + default: + patch = wm5102_sysclk_revb_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch); + break; } switch (event) { -- 1.7.10.4

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Re: [alsa-devel] [PATCH v3 09/12] ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open()
by Shawn Guo 06 Mar '13

06 Mar '13

On Wed, Mar 06, 2013 at 12:28:18AM +0100, Marek Vasut wrote: > Hi Shawn, > > > With generic DMA device tree binding and helper function > > dma_request_slave_channel() in place, dmaengine_pcm should support > > that in requesting DMA channel for users that support generic DMA > > device tree binding. > > > > Add a new API snd_dmaengine_generic_pcm_open() as the variant of > > snd_dmaengine_pcm_open(). It takes DMA client struct device pointer > > and slave channel name as arguments and calls generic DMA helper > > dma_request_slave_channel() to request DMA channel from dmaengine. > > > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > Cc: alsa-devel(a)alsa-project.org > > --- > > Changes since v2: > > - Drop 'const' from argument 'name' to fix the warning below. > > > [...] > > v3 changelog missing ;-) What did you adjust here please? The "Changes since v2" means v3 changelog. Shawn

1 0

[alsa-devel] [PATCH] ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs.
by Paul Handrigan 06 Mar '13

06 Mar '13

Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6361dab..3b20c86 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1180,7 +1180,11 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_MCLK; + /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ + if (priv->mclk >= 6400000) + priv->config[id].spc |= MCK_SCLK_64FS; + else + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: omap3pandora: Fix compilation error
by Peter Ujfalusi 06 Mar '13

06 Mar '13

Fixes: sound/soc/omap/omap3pandora.c: In function ‘omap3pandora_dac_event’: sound/soc/omap/omap3pandora.c:92:19: error: request for member ‘dev’ in something not a structure or union make[3]: *** [sound/soc/omap/omap3pandora.o] Error 1 Which is introduced by: dd194b4 ASoC: omap: Check regulator enable for DAC on Pandora Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap3pandora.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 10ced9d..9e46e1d 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -89,7 +89,7 @@ static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { ret = regulator_enable(omap3pandora_dac_reg); if (ret) { - dev_err(w->dapm.dev, "Failed to power DAC: %d\n", ret); + dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret); return ret; } mdelay(1); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 05 Mar '13

05 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sampling frequency while the interface is active. Reset the interface after setting the sampling frequency, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous file, causing distorted sound on sample rate changes. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- sound/usb/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d82e378..01978a6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -710,6 +710,16 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->need_setup_ep = false; } + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0d8c, 0x0304): /* C-Media - Schiit USB Interface */ + case USB_ID(0x0d8c, 0x0309): /* C-Media CM6631 */ + usb_set_interface(subs->dev, subs->cur_audiofmt->iface, + subs->cur_audiofmt->altset_idx); + break; + } + /* some unit conversions in runtime */ subs->data_endpoint->maxframesize = bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); -- 1.7.10.4

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Re: [alsa-devel] common way for hardware metering and monitoring between kernel/user land
by Clemens Ladisch 05 Mar '13

05 Mar '13

Takashi Sakamoto wrote: > Now i'm working in kernel land to support Echo's Fireworks (TM) devices > with better integration. > [...] > Well, now I need your advice about common ways for hardware metering and > monitoring between kernel land and user land. For almost all FireWire audio devices, it would not be possible to create ALSA mixer controls that would allow to control the device in a sensible way with a generic mixer application like alsamixer. (Things like matrix mixers, or meters integrated with corresponding volume sliders.) Therefore, custom mixer applications are required. Because devices typically have most of their customizations in the way in which they handle (mixer) controls, the FFADO guys and me have agreed that the kernel driver handles only stuff required for audio streaming; all other controls are accessed *only* from user space by going directly through /dev/fw* (or through a hwdep device if the kernel driver and userspace tools need to share a communication channel). So for now, there's no integration needed in the kernel driver. (I won't have time to test your driver before the next weekend.) Regards, Clemens

1 0

[alsa-devel] ASoC: no-pcm (backend) error propagation
by Frkuska, Joshua 05 Mar '13

05 Mar '13

Hello, I am working off of the 3.5.7 branch of the kernel using a frontend-backend approach for my driver. No additional patches have been added for ASoC/ALSA from the 3.5.7 kernel branch. My question relates to the way the no-pcm (backend) propagates errors to the front end. My FE(frontend) consists of a cpu-dai, a pcm platform driver, and a dummy-codec. My BE(backend) consists of another cpu-dai, a non-pcm platform driver, and a real codec. The non-pcm operates with the dmaengine but just in an autonomous fashion external to the PCM's knowledge. I have the pcm-platform driver passing data to the FE cpu-dai and the backend platform then takes the data from the frontend's cpu-dai and passes it on to the BE cpu-dai, which then configures the codec-dai to get output. In the frontend if/whenever there is an underrun error in the cpu-dai, I can directly call snd_pcm_stop to stop the stream and report the error state. I can do this because the substream->ops have properly been defined in the frontend pcm. However, in the backend, the substream->ops never get initialized and rightfully so since there isn't a valid pcm for the backend. My question is, what is the correct/any way to let the frontend PCM stream know that an underrun has occurred in the backend? Is there a mechanism to handle this in ASoC? Looking through the code, I cannot find such a mechanism. I have a feeling that there should be something like snd_soc_pcm_stop to check to see if we are in a FE/BE configuration and act accordingly to stop the FE PCM stream and report errors. Again, perhaps such a mechanism exists and if so any help would be appreciated. Thanks for your time and any comments/suggestions, Joshua

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[alsa-devel] am335x: mcasp in DIT mode
by Yegor Yefremov 05 Mar '13

05 Mar '13

I've solved the problem with DIT/SPDIF mode (see the issue description here: http://e2e.ti.com/support/arm/sitara_arm/f/791/p/247447/870030.aspx). In davinci_mcasp_hw_params() the DIT or I2S params will be set in the beginning. DIT mode configures the DAVINCI_MCASP_TXMASK_REG and DAVINCI_MCASP_TXFMT_REG. And here comes the problem: at the end of davinci_mcasp_hw_params() the davinci_config_channel_size() will touch the same registers again and thus overwrite the settings necessary for DIT. After I commented this routine I got the sound over S/PDIF and sii9022a HDMI transmitter and I could see the proper bits appearing on my oscilloscope. What were the best way to solve this problem? 1. execute davinci_config_channel_size() only if not in DIT mode? 2. for DIT only change the DAVINCI_MCASP_TXMASK_REG according to channel width? 3. execute if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); else davinci_hw_param(dev, substream->stream); after davinci_config_channel_size() Yegor

3 11

[alsa-devel] [PATCH v2] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 05 Mar '13

05 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- sound/usb/clock.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2..d12a9c4 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -255,6 +255,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, unsigned char data[4]; int err, crate; int clock = snd_usb_clock_find_source(chip, fmt->clock); + unsigned int previous_rate; if (clock < 0) return clock; @@ -266,6 +267,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data))) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + previous_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -294,6 +307,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, if (crate != rate) snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (crate != previous_rate) { + switch (chip->usb_id) { + case USB_ID(0x0d8c, 0x0304): /* C-Media - Schiit USB */ + case USB_ID(0x0d8c, 0x0309): /* C-Media CM6631 */ + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + break; + } + } + return 0; } -- 1.7.10.4

2 2

[alsa-devel] HDA Beep Device for Asus EeeBox EB1501P
by Harald Becker 05 Mar '13

05 Mar '13

Hi, I own an Asus EeeBox EB1501P with an ALC662 codec and had no HDA mixer channel for the speaker beep device. After digging somewhat in the kernel source I found that I only need to add a single line in the beep_white_list in patch_realtek.c. May be this entry can be included in the mainstream drivers: SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), After recompiling the driver with this entry, the mixer device for the beep channel appears and the digital beep sound is emitted on the connected speakers. ... with one problem: If I use echo -e "\a" in a shell or run the beep command utility the sound appears only on second and following invocations. Whenever the frequency changes with the beep command the first invocation stays silent or just produce some click due to ALSA driver power saving. On all further invocations with same frequency the requested tone is generated on external speakers. There is no difference in behavior if the commands are called in a terminal emulator or directly on a virtual text console. Does this happen only on my machine or is this a known problem? Is there anything I can do to fix this behavior? The kernel I used for testing is 3.8.0 from Gentoo Sources. Feel free to ask for any information you may need for debugging. Thx for any help ahead Harald

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Re: [alsa-devel] [PATCH 1/5 v7][RFC] ASoC: add snd_soc_register_cpu()
by Mark Brown 05 Mar '13

05 Mar '13

On Mon, Mar 04, 2013 at 07:48:36PM +0000, Liam Girdwood wrote: > I think we probably have more in common between platforms and codecs now > than we have that is different (certainly, things were very different in > the past). But you are right, is it really worth the big churn..... Definitely, I'm just not sure that the existing platform object really captures the new bits or if it maps more onto DAIs.

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[alsa-devel] [PATCH V2] ASoC: tegra: fix I2S bit count mask
by Stephen Warren 05 Mar '13

05 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> This register field is 11 bits wide, not 15 bits wide. Given the way this value is currently, used, this patch has no practical effect. However, it's still best if the value is correct. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Apply fix to Tegra20 as well as Tegra30. --- sound/soc/tegra/tegra20_i2s.h | 2 +- sound/soc/tegra/tegra30_i2s.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index c27069d..7299587 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -121,7 +121,7 @@ #define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA20_I2S_FIFO_SCR */ diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 34dc47b..a294d94 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -110,7 +110,7 @@ #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA30_I2S_OFFSET */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: If we only have a clock to synchronise with make it REFCLK
by Mark Brown 05 Mar '13

05 Mar '13

If there is only one clock active the FLL should use REFCLK rather than SYNCCLK as the clock to synchronise with since REFCLK is always required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 76 ++++++++++++++++++++++---------------------- 1 file changed, 38 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e456cb4..0599ff8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1132,14 +1132,30 @@ static void arizona_enable_fll(struct arizona_fll *fll, struct arizona *arizona = fll->arizona; int ret; - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); - if (fll->sync_src >= 0) - arizona_apply_fll(arizona, fll->base + 0x10, sync, + /* + * If we have both REFCLK and SYNCCLK then enable both, + * otherwise apply the SYNCCLK settings to REFCLK. + */ + if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); + if (fll->sync_src >= 0) + arizona_apply_fll(arizona, fll->base + 0x10, sync, + fll->sync_src); + } else if (fll->sync_src >= 0) { + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + arizona_apply_fll(arizona, fll->base, sync, fll->sync_src); + } else { + arizona_fll_err(fll, "No clocks provided\n"); + return; + } if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); @@ -1149,7 +1165,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->sync_src >= 0) + if (fll->ref_src >= 0 && fll->sync_src >= 0 && + fll->ref_src != fll->sync_src) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1180,9 +1197,6 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (source < 0) - return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref) return 0; @@ -1216,39 +1230,25 @@ int arizona_set_fll(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (fll->ref_src < 0 || fll->ref_src == source) { - if (fll->sync_src == ARIZONA_FLL_SRC_NONE && - fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) - return 0; - - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); - if (ret != 0) - return ret; - } - - fll->sync_src = ARIZONA_FLL_SRC_NONE; - fll->ref_src = source; - fll->ref_freq = Fref; - } else { - if (fll->sync_src == source && - fll->sync_freq == Fref && fll->fout == Fout) - return 0; - - if (Fout) { - ret = arizona_calc_fll(fll, &ref, fll->ref_freq, Fout); - if (ret != 0) - return ret; + if (fll->sync_src == source && + fll->sync_freq == Fref && fll->fout == Fout) + return 0; - ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (Fout) { + if (fll->ref_src >= 0) { + ret = arizona_calc_fll(fll, &ref, fll->ref_freq, + Fout); if (ret != 0) return ret; } - fll->sync_src = source; - fll->sync_freq = Fref; + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; } + + fll->sync_src = source; + fll->sync_freq = Fref; fll->fout = Fout; if (Fout) { -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: tegra: fix I2S bit count mask
by Stephen Warren 04 Mar '13

04 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> This register field is 11 bits wide, not 15 bits wide. Given the way this value is currently, used, this patch has no practical effect. However, it's still best if the value is correct. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 34dc47b..a294d94 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -110,7 +110,7 @@ #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA30_I2S_OFFSET */ -- 1.7.10.4

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[alsa-devel] [PATCH] hdajackretask: Some more UI fixes
by David Henningsson 04 Mar '13

04 Mar '13

- don't overwrite a set advanced override unless necessary - fix "not present" never selected - remove some trailing whitespaces Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/main-gtk.c | 116 ++++++++++++++++++++++--------------- hdajackretask/sysfs-pin-configs.c | 4 +- 2 files changed, 71 insertions(+), 49 deletions(-) diff --git a/hdajackretask/main-gtk.c b/hdajackretask/main-gtk.c index 4227669..9101af4 100644 --- a/hdajackretask/main-gtk.c +++ b/hdajackretask/main-gtk.c @@ -5,19 +5,22 @@ #include <string.h> #include <stdlib.h> #include <gtk/gtk.h> +#include <stdbool.h> #include "sysfs-pin-configs.h" #include "apply-changes.h" +typedef struct ui_data_t ui_data_t; + typedef struct pin_ui_data_t { pin_configs_t* pin_config; typical_pins_t pins_info[32]; GtkWidget *frame, *override, *jacktype; - GtkWidget* free_override_cb[FREE_OVERRIDES_COUNT]; + ui_data_t* owner; } pin_ui_data_t; -typedef struct ui_data_t { +struct ui_data_t { GList* pin_ui_data; GtkWidget *main_window; GtkWidget *content_scroll_widget; @@ -25,22 +28,35 @@ typedef struct ui_data_t { GtkWidget *codec_selection_combo; codec_name_t* current_codec; - int sysfs_pincount; + int sysfs_pincount; codec_name_t sysfs_codec_names[128]; pin_configs_t sysfs_pins[32]; gboolean free_overrides; gboolean trust_codec; gboolean trust_defcfg; gboolean model_auto; -} ui_data_t; +}; -static void override_toggled(GtkWidget* sender, pin_ui_data_t* data) +static void update_user_pin_config(ui_data_t* ui, pin_configs_t* cfg); + +static void update_override_sensitive(GtkWidget* sender, pin_ui_data_t* data) { int i; gboolean checked = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(sender)); gtk_widget_set_sensitive(data->jacktype, checked); for (i = 0; i < FREE_OVERRIDES_COUNT; i++) - gtk_widget_set_sensitive(data->free_override_cb[i], checked); + gtk_widget_set_sensitive(data->free_override_cb[i], checked); +} + +static void override_toggled(GtkWidget* sender, pin_ui_data_t* data) +{ + update_override_sensitive(sender, data); + update_user_pin_config(data->owner, data->pin_config); +} + +static void jacktype_changed(GtkWidget* sender, pin_ui_data_t* data) +{ + update_user_pin_config(data->owner, data->pin_config); } static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) @@ -56,7 +72,8 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) data = calloc(1, sizeof(pin_ui_data_t)); data->pin_config = pin_cfg; - + data->owner = ui; + { /* Frame */ gchar* d = get_config_description(pin_cfg->init_pin_config); gchar* c = g_strdup_printf("Pin ID: 0x%02x", pin_cfg->nid); @@ -93,6 +110,7 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) gtk_combo_box_set_active(GTK_COMBO_BOX(jacktype), index); gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(override), pin_cfg->user_override); g_signal_connect(override, "toggled", G_CALLBACK(override_toggled), data); + g_signal_connect(jacktype, "changed", G_CALLBACK(jacktype_changed), data); gtk_container_add(box, override); gtk_container_add(GTK_CONTAINER(jacktype_box), jacktype); @@ -125,6 +143,7 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) } if (index >= 0) gtk_combo_box_set_active(GTK_COMBO_BOX(data->free_override_cb[i]), index); + g_signal_connect(data->free_override_cb[i], "changed", G_CALLBACK(jacktype_changed), data); } gtk_grid_attach(grid, gtk_label_new("Connectivity"), 0, 0, 1, 1); @@ -148,7 +167,7 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) if (ui->free_overrides) gtk_container_add(box, GTK_WIDGET(grid)); } - override_toggled(data->override, data); + update_override_sensitive(data->override, data); gtk_container_add(GTK_CONTAINER(result), GTK_WIDGET(box)); ui->pin_ui_data = g_list_prepend(ui->pin_ui_data, data); @@ -166,45 +185,48 @@ static void free_pin_ui_data(pin_ui_data_t* data) static gint pin_config_find(pin_ui_data_t* pin_ui, pin_configs_t* cfg) { - return pin_ui->pin_config == cfg ? 0 : 1; + return pin_ui->pin_config == cfg ? 0 : 1; } -static void update_user_pin_config(ui_data_t* ui) +static void update_user_pin_config(ui_data_t* ui, pin_configs_t* cfg) { - int i; - for (i = 0; i < ui->sysfs_pincount; i++) { - pin_ui_data_t* pin_ui; - GList *pos = g_list_find_custom(ui->pin_ui_data, &ui->sysfs_pins[i], (GCompareFunc) pin_config_find); - ui->sysfs_pins[i].user_override = FALSE; - if (!pos) - continue; - pin_ui = pos->data; - if (!gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(pin_ui->override))) - continue; + pin_ui_data_t* pin_ui; + GList *pos = g_list_find_custom(ui->pin_ui_data, cfg, (GCompareFunc) pin_config_find); + cfg->user_override = FALSE; + if (!pos) + return; + pin_ui = pos->data; + if (!gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(pin_ui->override))) + return; - if (ui->free_overrides) { - int j; - int index; - unsigned long val = 0; - for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); - if (index < 0) - break; - val += get_free_override_list(j)[index].value; - } + if (ui->free_overrides) { + int j; + int index; + unsigned long val = 0; + for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); if (index < 0) - continue; - ui->sysfs_pins[i].user_pin_config = val; - } else { - int index; - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); - if (index < 0) - continue; - ui->sysfs_pins[i].user_pin_config = pin_ui->pins_info[index].pin_set; + break; + val += get_free_override_list(j)[index].value; } - ui->sysfs_pins[i].user_override = TRUE; + if (index < 0) + return; + cfg->user_pin_config = val; + } else { + int index; + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); + if (index < 0) + return; + cfg->user_pin_config = pin_ui->pins_info[index].pin_set; } + cfg->user_override = TRUE; +} +static void update_all_user_pin_config(ui_data_t* ui) +{ + int i; + for (i = 0; i < ui->sysfs_pincount; i++) + update_user_pin_config(ui, &ui->sysfs_pins[i]); } static gboolean validate_user_pin_config(ui_data_t* ui, GError** err) @@ -215,7 +237,7 @@ static gboolean validate_user_pin_config(ui_data_t* ui, GError** err) g_set_error(err, 0, 0, "You must first select a codec!"); return FALSE; } - update_user_pin_config(ui); + update_all_user_pin_config(ui); if (ui->free_overrides) return TRUE; @@ -325,7 +347,7 @@ static void resize_main_window(ui_data_t* ui) } } -static void update_codec_ui(ui_data_t* ui) +static void update_codec_ui(ui_data_t* ui, bool codec_change) { int codec_index = gtk_combo_box_get_active(GTK_COMBO_BOX(ui->codec_selection_combo)); int i; @@ -337,7 +359,8 @@ static void update_codec_ui(ui_data_t* ui) if (codec_index < 0) return; ui->current_codec = &ui->sysfs_codec_names[codec_index]; - ui->sysfs_pincount = get_pin_configs_list(ui->sysfs_pins, 32, ui->current_codec->card, ui->current_codec->device); + if (codec_change) + ui->sysfs_pincount = get_pin_configs_list(ui->sysfs_pins, 32, ui->current_codec->card, ui->current_codec->device); for (i = 0; i < ui->sysfs_pincount; i++) { GtkWidget *w = create_pin_ui(ui, &ui->sysfs_pins[i]); if (w) @@ -350,14 +373,14 @@ static void update_codec_ui(ui_data_t* ui) static void codec_selected(GtkComboBox* combo, gpointer user_data) { - update_codec_ui(user_data); + update_codec_ui(user_data, true); } static void showallpins_toggled(GtkWidget* sender, ui_data_t* ui_data) { gboolean checked = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(sender)); ui_data->trust_defcfg = !checked; - update_codec_ui(ui_data); + update_codec_ui(ui_data, false); } static void automodel_toggled(GtkWidget* sender, ui_data_t* ui_data) @@ -367,9 +390,8 @@ static void automodel_toggled(GtkWidget* sender, ui_data_t* ui_data) static void free_override_toggled(GtkWidget* sender, ui_data_t* ui_data) { - update_user_pin_config(ui_data); ui_data->free_overrides = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(sender)); - update_codec_ui(ui_data); + update_codec_ui(ui_data, false); } static const char* readme_text = @@ -517,7 +539,7 @@ int main(int argc, char *argv[]) ui = create_ui(); gtk_widget_show_all(ui->main_window); if (ui->codec_selection_combo) - update_codec_ui(ui); + update_codec_ui(ui, true); gtk_main(); return 0; } diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index cd2897e..4044989 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -460,7 +460,7 @@ static free_override_t channel_arr[] = { static free_override_t* type_order[FREE_OVERRIDES_COUNT] = { - pc_arr, location_arr, device_arr, jack_arr, + pc_arr, location_arr, device_arr, jack_arr, color_arr, no_presence_arr, group_nr_arr, channel_arr }; @@ -472,7 +472,7 @@ unsigned long get_free_override_mask(int type) AC_DEFCFG_DEVICE, AC_DEFCFG_CONN_TYPE, AC_DEFCFG_COLOR, - AC_DEFCFG_MISC & 1, + AC_DEFCFG_MISC & (AC_DEFCFG_MISC_NO_PRESENCE << AC_DEFCFG_MISC_SHIFT), AC_DEFCFG_DEF_ASSOC, AC_DEFCFG_SEQUENCE, }; -- 1.7.9.5

2 1

[alsa-devel] [patch] [ALSA] seq_oss_event: missing range checks
by Dan Carpenter 04 Mar '13

04 Mar '13

The "dev" variable could be out of bounds. Calling snd_seq_oss_synth_is_valid() checks that it is is a valid device which has been opened. We check this inside set_note_event() so this function can't succeed without a valid "dev". But we need to do the check earlier to prevent invalid dereferences and memory corruption. One call tree where "dev" could be out of bounds is: -> snd_seq_oss_oob_user() -> snd_seq_oss_process_event() -> extended_event() -> note_on_event() Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 066f5f3..c390886 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev static int note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { @@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st static int note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) {

2 1

[alsa-devel] [PATCH v4 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

3 5

[alsa-devel] [PATCH v3] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- v3: don't set mask in davinci_hw_dit_param() v2: fix checkpatch.pl warnings sound/soc/davinci/davinci-mcasp.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..dcd56f2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,20 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } @@ -772,9 +776,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* TXMASK for 24 bits */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, -- 1.7.7

3 3

[alsa-devel] Issue with Dmix
by mateen 04 Mar '13

04 Mar '13

HI We have enabled DMIX plugin for sample rate conversion and mixing of 2 pcm streams in our alsa package (Alsa lib 1.0.22 and alsa driver 1.0.22.1) to 48K output Sample rate.However we are seeing a noise issue with 44100 Sample rate input streams or Unsigned Stereo 48K stream. This is how our current design is: We call snd_pcm_lib_malloc_page() in which alsa driver internally allocates a dma buffer and map it when Dmix is invoked. We currently reads from that mmaped buffer to our hardware buffer inside a transfer_ack_begin() function which is called from snd_pcm_period_elapsed(). However we are getting a problem in understanding how do we know how much data has been written by dmix lib in mapped buffer? As we see no ioctl or corresponding hardware function call where dmix may pass information on Size of data being written to our driver. We do see appl_ptr and hw_ptr in runtime->mmap->control and status struct. Out architecture doesn’t support defining and mapping of these status and control structure at hardware level.Thus Dmix lib allocates them and copy them to runtime substream via ioctl. As far as I could understand both appl_ptr and hw_ptr are initialised to 0 and then appl_ptr is the one adjusted by lib after every write. Should we tap this value to update hw_ptr value in status information? Will difference of hw_ptr and appl_ptr will let us know amount of frames been written? Can anyone please help us to understand criteria and concept of data copy from intermediate to hardware buffer? Any information on this would be of immense help. Regards, Shaikh Mateen

1 1

[alsa-devel] [PATCH] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- sound/soc/davinci/davinci-mcasp.c | 27 +++++++++++++++++---------- 1 files changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..de59075 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,23 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + } + else + { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + } return 0; } -- 1.7.7

3 2

[alsa-devel] [PATCH v2] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- v2: fix checkpatch.pl warnings sound/soc/davinci/davinci-mcasp.c | 22 +++++++++++++--------- 1 files changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..ce32b75 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,20 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } -- 1.7.7

1 0

[alsa-devel] [PATCH 09/12] ASoC: dmaengine_pcm: support use of generic DMA helper
by Shawn Guo 04 Mar '13

04 Mar '13

With generic DMA device tree binding and helper function dma_request_slave_channel() in place, dmaengine_pcm should support that in requesting DMA channel for users that support generic DMA device tree binding. Instead of inventing a new API, it defines the parameters needed by dma_request_slave_channel() into struct snd_dma_channel_params, interprets filter_data into snd_dma_channel_params, and calls the helper in case that dmaengine_pcm users pass in a NULL filter_fn. Then, dmaengine_pcm users can call snd_dmaengine_pcm_open() with NULL filter_fn and snd_dma_channel_params being filter_data to direct the API to request DMA channel using generic DMA helper. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- include/sound/dmaengine_pcm.h | 5 +++++ sound/soc/soc-dmaengine-pcm.c | 19 +++++++++++++------ 2 files changed, 18 insertions(+), 6 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b877334..6c0f795 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -18,6 +18,11 @@ #include <sound/pcm.h> #include <linux/dmaengine.h> +struct snd_dma_channel_params { + struct device *dev; + char *name; +}; + /** * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM * substream diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d92..7ef8034 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -247,12 +247,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, dma_filter_fn filter_fn, void *filter_data) { - dma_cap_mask_t mask; - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_CYCLIC, mask); - prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data); + if (filter_fn) { + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + prtd->dma_chan = dma_request_channel(mask, filter_fn, + filter_data); + } else { + struct snd_dma_channel_params *params = filter_data; + prtd->dma_chan = dma_request_slave_channel(params->dev, + params->name); + } if (!prtd->dma_chan) return -ENXIO; -- 1.7.9.5

3 4

[alsa-devel] [PATCH] davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

2 1

[alsa-devel] [PATCH 2/2] ALSA: snd-usb-caiaq: switch to dev_*() logging
by Daniel Mack 03 Mar '13

03 Mar '13

Get rid of the proprietary functions log() and debug() and use the generic dev_*() approach. A macro is needed to cast a cdev to a struct device *. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/audio.c | 61 ++++++++++++++++++++++++++++++----------------- sound/usb/caiaq/control.c | 1 + sound/usb/caiaq/device.c | 42 +++++++++++++++++--------------- sound/usb/caiaq/device.h | 10 +------- sound/usb/caiaq/input.c | 8 +++++-- sound/usb/caiaq/midi.c | 9 +++---- 6 files changed, 75 insertions(+), 56 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 75d8ba9..67330af 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/spinlock.h> #include <linux/slab.h> #include <linux/init.h> @@ -101,8 +102,9 @@ all_substreams_zero(struct snd_pcm_substream **subs) static int stream_start(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, cdev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); if (cdev->streaming) return -EINVAL; @@ -118,7 +120,8 @@ static int stream_start(struct snd_usb_caiaqdev *cdev) for (i = 0; i < N_URBS; i++) { ret = usb_submit_urb(cdev->data_urbs_in[i], GFP_ATOMIC); if (ret) { - log("unable to trigger read #%d! (ret %d)\n", i, ret); + dev_err(dev, "unable to trigger read #%d! (ret %d)\n", + i, ret); cdev->streaming = 0; return -EPIPE; } @@ -130,8 +133,9 @@ static int stream_start(struct snd_usb_caiaqdev *cdev) static void stream_stop(struct snd_usb_caiaqdev *cdev) { int i; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, cdev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); if (!cdev->streaming) return; @@ -150,17 +154,21 @@ static void stream_stop(struct snd_usb_caiaqdev *cdev) static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); - debug("%s(%p)\n", __func__, substream); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, substream); substream->runtime->hw = cdev->pcm_info; snd_pcm_limit_hw_rates(substream->runtime); + return 0; } static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (all_substreams_zero(cdev->sub_playback) && all_substreams_zero(cdev->sub_capture)) { /* when the last client has stopped streaming, @@ -175,14 +183,12 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); - debug("%s(%p)\n", __func__, sub); deactivate_substream(cdev, sub); return snd_pcm_lib_free_pages(sub); } @@ -201,8 +207,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) int index = substream->number; struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { int out_pos; @@ -283,8 +290,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p) cmd %d\n", __func__, sub, cmd); + dev_dbg(dev, "%s(%p) cmd %d\n", __func__, sub, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -443,6 +451,7 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + struct device *dev = caiaqdev_to_dev(cdev); int stream, i; /* paranoia check */ @@ -479,8 +488,8 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, if (usb_buf[i] != ((stream << 1) | c) && !cdev->first_packet) { if (!cdev->input_panic) - printk(" EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", - ((stream << 1) | c), usb_buf[i], c, stream, i); + dev_warn(dev, " EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", + ((stream << 1) | c), usb_buf[i], c, stream, i); cdev->input_panic = 1; } @@ -497,6 +506,8 @@ static void read_in_urb(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { + struct device *dev = caiaqdev_to_dev(cdev); + if (!cdev->streaming) return; @@ -516,7 +527,7 @@ static void read_in_urb(struct snd_usb_caiaqdev *cdev, } if ((cdev->input_panic || cdev->output_panic) && !cdev->warned) { - debug("streaming error detected %s %s\n", + dev_warn(dev, "streaming error detected %s %s\n", cdev->input_panic ? "(input)" : "", cdev->output_panic ? "(output)" : ""); cdev->warned = 1; @@ -619,6 +630,7 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *cdev; + struct device *dev; struct urb *out = NULL; int i, frame, len, send_it = 0, outframe = 0; size_t offset = 0; @@ -627,6 +639,7 @@ static void read_completed(struct urb *urb) return; cdev = info->cdev; + dev = caiaqdev_to_dev(cdev); if (!cdev->streaming) return; @@ -639,7 +652,7 @@ static void read_completed(struct urb *urb) } if (!out) { - log("Unable to find an output urb to use\n"); + dev_err(dev, "Unable to find an output urb to use\n"); goto requeue; } @@ -708,6 +721,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) int i, frame; struct urb **urbs; struct usb_device *usb_dev = cdev->chip.dev; + struct device *dev = caiaqdev_to_dev(cdev); unsigned int pipe; pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? @@ -716,7 +730,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL); if (!urbs) { - log("unable to kmalloc() urbs, OOM!?\n"); + dev_err(dev, "unable to kmalloc() urbs, OOM!?\n"); *ret = -ENOMEM; return NULL; } @@ -724,7 +738,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) for (i = 0; i < N_URBS; i++) { urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); if (!urbs[i]) { - log("unable to usb_alloc_urb(), OOM!?\n"); + dev_err(dev, "unable to usb_alloc_urb(), OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -732,7 +746,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { - log("unable to kmalloc() transfer buffer, OOM!?\n"); + dev_err(dev, "unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -783,6 +797,7 @@ static void free_urbs(struct urb **urbs) int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); cdev->n_audio_in = max(cdev->spec.num_analog_audio_in, cdev->spec.num_digital_audio_in) / @@ -792,12 +807,12 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) CHANNELS_PER_STREAM; cdev->n_streams = max(cdev->n_audio_in, cdev->n_audio_out); - debug("cdev->n_audio_in = %d\n", cdev->n_audio_in); - debug("cdev->n_audio_out = %d\n", cdev->n_audio_out); - debug("cdev->n_streams = %d\n", cdev->n_streams); + dev_dbg(dev, "cdev->n_audio_in = %d\n", cdev->n_audio_in); + dev_dbg(dev, "cdev->n_audio_out = %d\n", cdev->n_audio_out); + dev_dbg(dev, "cdev->n_streams = %d\n", cdev->n_streams); if (cdev->n_streams > MAX_STREAMS) { - log("unable to initialize device, too many streams.\n"); + dev_err(dev, "unable to initialize device, too many streams.\n"); return -EINVAL; } @@ -805,7 +820,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) cdev->n_audio_out, cdev->n_audio_in, &cdev->pcm); if (ret < 0) { - log("snd_pcm_new() returned %d\n", ret); + dev_err(dev, "snd_pcm_new() returned %d\n", ret); return ret; } @@ -880,7 +895,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *cdev) { - debug("%s(%p)\n", __func__, cdev); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, cdev); stream_stop(cdev); free_urbs(cdev->data_urbs_in); free_urbs(cdev->data_urbs_out); diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 2c51959..ae6b50f 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -17,6 +17,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/init.h> #include <linux/usb.h> #include <sound/control.h> diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 45c3853..d898f73 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -20,6 +20,7 @@ */ #include <linux/moduleparam.h> +#include <linux/device.h> #include <linux/interrupt.h> #include <linux/module.h> #include <linux/init.h> @@ -159,10 +160,11 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) { int ret; struct snd_usb_caiaqdev *cdev = urb->context; + struct device *dev = caiaqdev_to_dev(cdev); unsigned char *buf = urb->transfer_buffer; if (urb->status || !cdev) { - log("received EP1 urb->status = %i\n", urb->status); + dev_warn(dev, "received EP1 urb->status = %i\n", urb->status); return; } @@ -170,7 +172,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) case EP1_CMD_GET_DEVICE_INFO: memcpy(&cdev->spec, buf+1, sizeof(struct caiaq_device_spec)); cdev->spec.fw_version = le16_to_cpu(cdev->spec.fw_version); - debug("device spec (firmware %d): audio: %d in, %d out, " + dev_dbg(dev, "device spec (firmware %d): audio: %d in, %d out, " "MIDI: %d in, %d out, data alignment %d\n", cdev->spec.fw_version, cdev->spec.num_analog_audio_in, @@ -209,7 +211,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) cdev->ep1_in_urb.actual_length = 0; ret = usb_submit_urb(&cdev->ep1_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, @@ -239,6 +241,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, { int ret; char tmp[5]; + struct device *dev = caiaqdev_to_dev(cdev); switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; @@ -259,7 +262,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, tmp[3] = bpp >> 8; tmp[4] = 1; /* packets per microframe */ - debug("setting audio params: %d Hz, %d bits, %d bpp\n", + dev_dbg(dev, "setting audio params: %d Hz, %d bits, %d bpp\n", rate, depth, bpp); cdev->audio_parm_answer = -1; @@ -274,7 +277,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, return -EPIPE; if (cdev->audio_parm_answer != 1) - debug("unable to set the device's audio params\n"); + dev_dbg(dev, "unable to set the device's audio params\n"); else cdev->bpp = bpp; @@ -293,6 +296,7 @@ static void setup_card(struct snd_usb_caiaqdev *cdev) { int ret; char val[4]; + struct device *dev = caiaqdev_to_dev(cdev); /* device-specific startup specials */ switch (cdev->chip.usb_id) { @@ -346,32 +350,32 @@ static void setup_card(struct snd_usb_caiaqdev *cdev) cdev->spec.num_digital_audio_in > 0) { ret = snd_usb_caiaq_audio_init(cdev); if (ret < 0) - log("Unable to set up audio system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up audio system (ret=%d)\n", ret); } if (cdev->spec.num_midi_in + cdev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(cdev); if (ret < 0) - log("Unable to set up MIDI system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up MIDI system (ret=%d)\n", ret); } #ifdef CONFIG_SND_USB_CAIAQ_INPUT ret = snd_usb_caiaq_input_init(cdev); if (ret < 0) - log("Unable to set up input system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up input system (ret=%d)\n", ret); #endif /* finally, register the card and all its sub-instances */ ret = snd_card_register(cdev->chip.card); if (ret < 0) { - log("snd_card_register() returned %d\n", ret); + dev_err(dev, "snd_card_register() returned %d\n", ret); snd_card_free(cdev->chip.card); } ret = snd_usb_caiaq_control_init(cdev); if (ret < 0) - log("Unable to set up control system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up control system (ret=%d)\n", ret); } static int create_card(struct usb_device *usb_dev, @@ -412,10 +416,11 @@ static int init_card(struct snd_usb_caiaqdev *cdev) char *c, usbpath[32]; struct usb_device *usb_dev = cdev->chip.dev; struct snd_card *card = cdev->chip.card; + struct device *dev = caiaqdev_to_dev(cdev); int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { - log("can't set alt interface.\n"); + dev_err(dev, "can't set alt interface.\n"); return -EIO; } @@ -473,8 +478,7 @@ static int init_card(struct snd_usb_caiaqdev *cdev) } usb_make_path(usb_dev, usbpath, sizeof(usbpath)); - snprintf(card->longname, sizeof(card->longname), - "%s %s (%s)", + snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", cdev->vendor_name, cdev->product_name, usbpath); setup_card(cdev); @@ -496,7 +500,7 @@ static int snd_probe(struct usb_interface *intf, usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { - log("unable to init card! (ret=%d)\n", ret); + dev_err(&usb_dev->dev, "unable to init card! (ret=%d)\n", ret); snd_card_free(card); return ret; } @@ -506,15 +510,16 @@ static int snd_probe(struct usb_interface *intf, static void snd_disconnect(struct usb_interface *intf) { - struct snd_usb_caiaqdev *cdev; struct snd_card *card = usb_get_intfdata(intf); - - debug("%s(%p)\n", __func__, intf); + struct snd_usb_caiaqdev *cdev = caiaqdev(card); + struct device *dev; if (!card) return; - cdev = caiaqdev(card); + dev = caiaqdev_to_dev(cdev); + dev_dbg(dev, "%s(%p)\n", __func__, intf); + snd_card_disconnect(card); #ifdef CONFIG_SND_USB_CAIAQ_INPUT @@ -539,4 +544,3 @@ static struct usb_driver snd_usb_driver = { }; module_usb_driver(snd_usb_driver); - diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 7176a0e..ad102fa 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -25,16 +25,7 @@ #define CAIAQ_USB_STR_LEN 0xff #define MAX_STREAMS 32 -//#define SND_USB_CAIAQ_DEBUG - #define MODNAME "snd-usb-caiaq" -#define log(x...) snd_printk(KERN_WARNING MODNAME" log: " x) - -#ifdef SND_USB_CAIAQ_DEBUG -#define debug(x...) snd_printk(KERN_WARNING MODNAME " debug: " x) -#else -#define debug(x...) do { } while(0) -#endif #define EP1_CMD_GET_DEVICE_INFO 0x1 #define EP1_CMD_READ_ERP 0x2 @@ -129,6 +120,7 @@ struct snd_usb_caiaq_cb_info { }; #define caiaqdev(c) ((struct snd_usb_caiaqdev*)(c)->private_data) +#define caiaqdev_to_dev(d) (d->chip.card->dev) int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bbp); int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *cdev, int digital, int analog, int erp); diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a32ad7c..fe8f4b4 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/gfp.h> #include <linux/init.h> #include <linux/usb.h> @@ -340,6 +341,8 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, const unsigned char *buf, unsigned int len) { + struct device *dev = caiaqdev_to_dev(cdev); + while (len) { unsigned int i, block_id = (buf[0] << 8) | buf[1]; @@ -466,7 +469,7 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, break; default: - debug("%s(): bogus block (id %d)\n", + dev_dbg(dev, "%s(): bogus block (id %d)\n", __func__, block_id); return; } @@ -500,6 +503,7 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *cdev = urb->context; + struct device *dev = caiaqdev_to_dev(cdev); unsigned char *buf = urb->transfer_buffer; int ret; @@ -535,7 +539,7 @@ requeue: cdev->ep4_in_urb->actual_length = 0; ret = usb_submit_urb(cdev->ep4_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } static int snd_usb_caiaq_input_open(struct input_dev *idev) diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 63c5a2c..2d75884 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/usb.h> #include <linux/gfp.h> #include <sound/rawmidi.h> @@ -65,6 +66,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, struct snd_rawmidi_substream *substream) { int len, ret; + struct device *dev = caiaqdev_to_dev(cdev); cdev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; cdev->midi_out_buf[1] = 0; /* port */ @@ -79,9 +81,9 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, ret = usb_submit_urb(&cdev->midi_out_urb, GFP_ATOMIC); if (ret < 0) - log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," - "ret=%d, len=%d\n", - substream, ret, len); + dev_err(dev, + "snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d\n", substream, ret, len); else cdev->midi_out_active = 1; } @@ -171,4 +173,3 @@ void snd_usb_caiaq_midi_output_done(struct urb* urb) snd_usb_caiaq_midi_send(cdev, cdev->midi_out_substream); } - -- 1.8.1.4

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[alsa-devel] usb: Endpoints sharing the same interface
by Mark Hills 03 Mar '13

03 Mar '13

Novation Twitch [1] is a USB device with audio capture and playback endpoints on the same bInterfaceNumber. This has thrown up some tricky cases in the USB audio driver, and I'd appreciate some advice. I'll follow up with my patches so far as an RFC. This device has endpoints for 2 channels capture (0x01), 4 channels playback (0x82), both on interface #0. Interface #1 is raw MIDI. I seem to be finding code that assumes pcm to always use the first endpoint on an interface. eg. problems with maxpacksize (see patch in thread) Now I'm in a position where playback _or_ capture works reliably. But to start capture stops playback, and vice-versa (or sometimes causes hangs) It seems like snd_pcm_prepare() concludes that need_setup_ep is required, but I think this may be based on something going on in set_sample_rate_v1() also looking to the first endpoint -- usb_sndctrlpipe(dev,0). Before I do any more work on this, are there any hints where to go with this? If we can overcome thse limitations then the same would likely apply to Saffire 6 USB and other Focusrite devices [2], and maybe some of the other devices with quirks that only support playback. Many thanks [1] http://uk.novationmusic.com/digital-dj/twitch/ [2] http://focusritedevelopmentteam.wordpress.com/category/drivers/ -- Mark --- Bus 005 Device 010: ID 1235:0018 Novation EMS Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1235 Novation EMS idProduct 0x0018 bcdDevice 1.00 iManufacturer 1 Focusrite A.E. Ltd iProduct 2 Twitch iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 64 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 498mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x024c 1x 588 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 1 Device Status: 0x0000 (Bus Powered)

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[alsa-devel] amdtp: syt handling
by Robert von Chamier 03 Mar '13

03 Mar '13

Hello, i am working on the firewire amdtp module. My aim is to add the capture functionality. Already in my prototype the module captures up to 192k sampling rate. But ... How i have to handle the syt in the CIP packages? Do i compare the syt with the corresponding cycle? or ... Do i take the syt as timestamp for alsa? Is it possible to commit a timestamp to alsa for the corresponding samples? Can you help me? Robert

2 1

[alsa-devel] [PATCH] ASoC: max98090: Convert to devm_regmap_init_i2c
by Axel Lin 03 Mar '13

03 Mar '13

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc17604..6dd36da 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2324,28 +2324,20 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; - max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { ret = PTR_ERR(max98090->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); - goto err_enable; + return ret; } - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_max98090, max98090_dai, - ARRAY_SIZE(max98090_dai)); - if (ret < 0) - regmap_exit(max98090->regmap); - -err_enable: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, + max98090_dai, ARRAY_SIZE(max98090_dai)); } static int max98090_i2c_remove(struct i2c_client *client) { - struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); snd_soc_unregister_codec(&client->dev); - regmap_exit(max98090->regmap); return 0; } -- 1.7.9.5

2 2

[alsa-devel] [PATCH RFT] ASoC: wm8350: Use jiffies rather than msecs in schedule_delayed_work()
by Axel Lin 03 Mar '13

03 Mar '13

The delay parameter of schedule_delayed_work() is number of jiffies to wait rather than miliseconds. Before commit 6d3c26bcb "ASoC: Use delayed work to debounce WM8350 jack IRQs", the debounce time is 200 miliseconds in wm8350_hp_jack_handler(). So I think this is a bug when convert to use delayed work. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index ec0efc1..0e8b3aa 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, 200); + schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1318,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, 200); + schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: omap: Check regulator enable for DAC on Pandora
by Mark Brown 03 Mar '13

03 Mar '13

This will probably never fail but it's better style. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/omap3pandora.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 805512f..953483e 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -85,7 +85,11 @@ static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, * VCC power on/off and /PD pin high/low */ if (SND_SOC_DAPM_EVENT_ON(event)) { - regulator_enable(omap3pandora_dac_reg); + ret = regulator_enable(omap3pandora_dac_reg); + if (ret != 0) { + dev_err(w->dapm.dev, "Failed to power DAC: %d\n", ret); + return ret; + } mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); } else { -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH v2 3/4] spi: s3c64xx: add dma maxburst size initialization
by Grant Likely 02 Mar '13

02 Mar '13

On Fri, 22 Feb 2013 17:09:18 +0900, Boojin Kim <boojin.kim(a)samsung.com> wrote: > This patch adds dma maxburst size initialization. > The maxburst should be set by MODE_CFGn.DMA_TYPE, > because the pl330 dma driver supports burst mode. > > Signed-off-by: Hyeonkook Kim <hk619.kim(a)samsung.com> Acked-by: Grant Likely <grant.likely(a)secretlab.ca> This series probably needs to be merged together through the arm-soc tree. g. > --- > drivers/spi/spi-s3c64xx.c | 18 ++++++++++++++++++ > 1 files changed, 18 insertions(+), 0 deletions(-) > > diff --git a/drivers/spi/spi-s3c64xx.c b/drivers/spi/spi-s3c64xx.c > index e862ab8..5e6cafe 100644 > --- a/drivers/spi/spi-s3c64xx.c > +++ b/drivers/spi/spi-s3c64xx.c > @@ -287,10 +287,20 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > struct s3c64xx_spi_driver_data *sdd; > struct samsung_dma_prep info; > struct samsung_dma_config config; > + void __iomem *regs; > + unsigned int mode_cfg; > > if (dma->direction == DMA_DEV_TO_MEM) { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, rx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->rx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > config.width = sdd->cur_bpw / 8; > @@ -298,6 +308,14 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > } else { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, tx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->tx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > config.width = sdd->cur_bpw / 8; > -- > 1.7.5.4 > > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- Grant Likely, B.Sc, P.Eng. Secret Lab Technologies, Ltd.

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Re: [alsa-devel] [PATCH v2 3/4] spi: s3c64xx: add dma maxburst size initialization
by Grant Likely 02 Mar '13

02 Mar '13

On Fri, 22 Feb 2013 17:09:18 +0900, Boojin Kim <boojin.kim(a)samsung.com> wrote: > This patch adds dma maxburst size initialization. > The maxburst should be set by MODE_CFGn.DMA_TYPE, > because the pl330 dma driver supports burst mode. > > Signed-off-by: Hyeonkook Kim <hk619.kim(a)samsung.com> Hello Hyeonkook, I applied this patch but then discovered that it relys on the previous patches that you've posted. When patched depend on each other like this please post them as a series with [PATCH x/4] in the subject line. 'git send-email' will format this correctly for you. g. > --- > drivers/spi/spi-s3c64xx.c | 18 ++++++++++++++++++ > 1 files changed, 18 insertions(+), 0 deletions(-) > > diff --git a/drivers/spi/spi-s3c64xx.c b/drivers/spi/spi-s3c64xx.c > index e862ab8..5e6cafe 100644 > --- a/drivers/spi/spi-s3c64xx.c > +++ b/drivers/spi/spi-s3c64xx.c > @@ -287,10 +287,20 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > struct s3c64xx_spi_driver_data *sdd; > struct samsung_dma_prep info; > struct samsung_dma_config config; > + void __iomem *regs; > + unsigned int mode_cfg; > > if (dma->direction == DMA_DEV_TO_MEM) { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, rx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->rx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > config.width = sdd->cur_bpw / 8; > @@ -298,6 +308,14 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > } else { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, tx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->tx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > config.width = sdd->cur_bpw / 8; > -- > 1.7.5.4 > > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- Grant Likely, B.Sc, P.Eng. Secret Lab Technologies, Ltd.

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Re: [alsa-devel] [PATCH v2 3/4] spi: s3c64xx: add dma maxburst size initialization
by Grant Likely 02 Mar '13

02 Mar '13

On Fri, 22 Feb 2013 17:09:18 +0900, Boojin Kim <boojin.kim(a)samsung.com> wrote: > This patch adds dma maxburst size initialization. > The maxburst should be set by MODE_CFGn.DMA_TYPE, > because the pl330 dma driver supports burst mode. > > Signed-off-by: Hyeonkook Kim <hk619.kim(a)samsung.com> Applied, thanks. g.

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Re: [alsa-devel] [Alsa-user] Status of CM6631 USB
by Daniel Mack 02 Mar '13

02 Mar '13

Hi Torstein, (+ alsa-devel) On 02.03.2013 12:09, Torstein Hegge wrote: > Daniel Mack <zonque <at> gmail.com> writes: >>>> This is a known bug, also most probably flaw in the CMEDIA chip, >>>> and not yet properly worked around in the snd-usb driver. If you >>>> want to investigate, have a look at the feedback format >>>> autodetection code in sound/usb/endpoint.c. >>> >>> Thanks, I'll have a look. >> >> I believe what's going on here is that the device is confused after >> sample rate changes and keeps sending back feedback data for the old >> rate on its feedback endpoint. That causes the autodetection algorithm, >> which kicks in on every first packet of a stream, to go nuts, which >> causes the part of the driver that sends the data assume a very wrong >> rate. Some printk() there should give you a hint. > > I had a look at this using another C-Media CM6631 based device, the Corda > Daccord (usbid 0x0d8c, 0x0309). As far as I can see, what happens in the > feedback format auto-detection is: > > When starting playback of a file with a different sampling frequency than the > previous file, the reported feedback frequency is in the same range as the > previously played file, varying within +-1%. > > The feedback format detection gives a shift of -1 and reports the feedback > frequency as ~48kHz when playing a 44.1kHz file after a 96kHz file. Same thing > happens when playing a 96kHz file after a 44.1kHz file, the device reports > feedback frequency in the 44.1kHz range, the feedback format detection gives a > shift of +1 and reports the feedback frequency as ~88kHz. That's what I suspected. > Bypassing the feedback format detection by hard-coding the frequency shift to > zero and setting the feedback frequency ep->freqm equal to the nominal > frequency ep->freqn if the feedback frequency is far off from the expected > value (like 48kHz when expecting 44.1kHz) does not change the distortion. > > Similarly, forcing the feedback detection to run multiple times until it > reaches an acceptable value just keeps the feedback detection to repeatedly > until playback is stopped. This probably means any amount of skipping packages > early in the stream won't work. Ok. > It seems like the device stays in the previous frequency until the interface > is reset. One possible workaround is to call usb_set_interface() again after > snd_usb_init_sample_rate(), like this: I see - that also explains why explicitly stopping and restarting the stream fixes the problem in most cases, right? > > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index d82e378..01978a6 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -710,6 +710,16 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream > *substream) > subs->need_setup_ep = false; > } > > + /* Some devices doesn't respond to sample rate changes while the > + * interface is active. */ > + switch (subs->stream->chip->usb_id) { > + case USB_ID(0x0d8c, 0x0304): /* C-Media - Schiit USB Interface */ > + case USB_ID(0x0d8c, 0x0309): /* C-Media CM6631 */ > + usb_set_interface(subs->dev, subs->cur_audiofmt->iface, > + subs->cur_audiofmt->altset_idx); > + break; > + } > + > /* some unit conversions in runtime */ > subs->data_endpoint->maxframesize = > bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); > > This fixes all problems related to sample rate changes for me. I have only > tested with the Daccord, but it should work for the Schiit and other CM6631 > based devices as well. Many thanks for looking into this! Much appreciated. Could you cook a real patch please and send it to alsa-devel? Have a look at Documentation/SubmittingPatches if that's your first submission. Best, Daniel

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[alsa-devel] [PATCH 1/2] ASoC: max98090: Fix checkpatch errors related to spacing
by Sachin Kamat 02 Mar '13

02 Mar '13

Fixes the following type of checkpatch errors: ERROR: "foo * bar" should be "foo *bar" Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/max98090.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1cf017f..89f83f8 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -507,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, return 0; } -static const char * max98090_perf_pwr_text[] = +static const char *max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; -static const char * max98090_pwr_perf_text[] = +static const char *max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; static const struct soc_enum max98090_vcmbandgap_enum = SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); -static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; +static const char *max98090_osr128_text[] = { "64*fs", "128*fs" }; static const struct soc_enum max98090_osr128_enum = SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, @@ -533,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum = M98090_FLT_DMIC34MODE_SHIFT, ARRAY_SIZE(max98090_mode_text), max98090_mode_text); -static const char * max98090_drcatk_text[] = +static const char *max98090_drcatk_text[] = { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; static const struct soc_enum max98090_drcatk_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); -static const char * max98090_drcrls_text[] = +static const char *max98090_drcrls_text[] = { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; static const struct soc_enum max98090_drcrls_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); -static const char * max98090_alccmp_text[] = +static const char *max98090_alccmp_text[] = { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; static const struct soc_enum max98090_alccmp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); -static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; +static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; static const struct soc_enum max98090_drcexp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, @@ -857,7 +857,7 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); -static const char * max98090_micpre_text[] = { "Off", "On" }; +static const char *max98090_micpre_text[] = { "Off", "On" }; static const struct soc_enum max98090_pa1en_enum = SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: soc-core: use snd_soc_register_dais() on codec
by Kuninori Morimoto 02 Mar '13

02 Mar '13

snd_soc_register_dais() considers dai counts inside. snd_soc_register_codec() does not need to care for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ceac893..5f44d22 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4047,11 +4047,10 @@ int snd_soc_register_codec(struct device *dev, mutex_unlock(&client_mutex); /* register any DAIs */ - if (num_dai) { - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) - dev_err(codec->dev, "ASoC: Failed to regster" - " DAIs: %d\n", ret); + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto fail_codec_name; } dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); @@ -4073,7 +4072,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; - int i; list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) @@ -4082,9 +4080,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - if (codec->num_dai) - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(dev, codec->num_dai); mutex_lock(&client_mutex); list_del(&codec->list); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/5 v7][RFC] ASoC: add snd_soc_register_cpu()
by Mark Brown 02 Mar '13

02 Mar '13

On Fri, Mar 01, 2013 at 09:42:10PM +0000, Liam Girdwood wrote: > On Fri, 2013-03-01 at 21:50 +0100, Lars-Peter Clausen wrote: > > object. I mean the distinction is mostly for historic reasons from a time where > > it was only possible to bind a the cpu_dai of a link to a CPU DAI and the > > codec_dai to a CODEC DAI. But these dais both can for example be bound CODEC > > DAIs. I think to have a common base here would allow us to do some cleanups to > > the ASoC core and also avoid the duplication with snd_soc_of_get_cpu_dai_name > > and snd_soc_of_get_codec_dai_name which was introduced in this series. > I've been thinking about this distinction for some time too (since > multi-component support). You are right, the division with CODEC and CPU > DAIs is entirely historical. The same logic also applies to platform and > codec drivers too. The DAIs are now the same of course, it's just the containing object that differs. Platform drivers are a bit different to CODEC drivers for me though in that they encapsulate the DMA equivalent of a DAI - I can see a SoC having multiple platform drivers if it's got a couple of different DMA controllers for example (I've seen some which have a fancier one on faster buses for example). > This does need some mechanical cleanup where we have a generic audio > component driver object that contains generic DAIs. i.e. codec and > platform drivers would become generic component drivers with generic > DAIs. There's also the perennial questions about code churn and if it's worth it of course. Should we be adding a chip driver type rather than a CPU one here and then going through and doing the work to make sure that you can use a chip driver wherever you use a CODEC driver so we can start converting things over?

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Re: [alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by wendelin klimann 01 Mar '13

01 Mar '13

Hello and thanks for the support I would advice to update. Eventually you want to upstream the codec driver, > so > it is going to be easier for you. > Well i followed your advice and updated to kernel version 3.7.4+ and to ALSA version k3.7.4+, and i will be glad to upstram the finalized driver. As you recommended i am using your omap_twl4030.c file as a starting-point for my driver development, and i am trying to get the adapted machine driver to run but without success. The machine driver is not even going into * static __devinit int omap_pcm3168_probe(struct platform_device *pdev)* and i am really stuck at this point. It would be really nice if you could help me. Regards Wendelin File: /* * omap-pcm3168.c -- PCM3168 ASoC driver for the BeagleBoard. * * based on: * > omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec * > * > Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com * > All rights reserved. * > * > Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA * */ #include <linux/platform_device.h> //#include <linux/platform_data/omap-twl4030.h> #include <linux/module.h> #include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include "omap-mcbsp.h" #include "omap-pcm.h" static int omap_pcm3168_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; unsigned int fmt; int ret; fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF; /* Set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { dev_err(card->dev, "can't set codec DAI configuration\n"); return ret; } /* Set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { dev_err(card->dev, "can't set cpu DAI configuration\n"); return ret; } return 0; } static struct snd_soc_ops omap_pcm3168_ops = { .hw_params = omap_pcm3168_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_pcm3168_dai_links[] = { { .name = "PCM3168", .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "pcm3168-hifi", .platform_name = "omap-pcm-audio", .codec_name = "pcm3168-codec.0", .ops = &omap_pcm3168_ops, }, }; /* Audio machine driver */ static struct snd_soc_card omap_pcm3168_card = { .owner = THIS_MODULE, .dai_link = omap_pcm3168_dai_links, .num_links = ARRAY_SIZE(omap_pcm3168_dai_links), }; static __devinit int omap_pcm3168_probe(struct platform_device *pdev) { printk("in __devinit omap_pcm3168_probe -> by kli"); struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_pcm3168_card; int ret = 0; card->dev = &pdev->dev; if (node) { struct device_node *dai_node; if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } dai_node = of_parse_phandle(node, "ti,mcbsp", 0); if (!dai_node) { dev_err(&pdev->dev, "McBSP node is not provided\n"); return -EINVAL; } omap_pcm3168_dai_links[0].cpu_dai_name = NULL; omap_pcm3168_dai_links[0].cpu_of_node = dai_node; } /*else if (pdata) { if (pdata->card_name) { card->name = pdata->card_name; } else { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } }*/ else { dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } return 0; } static int __devexit omap_pcm3168_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static const struct of_device_id omap_pcm3168_of_match[] = { {.compatible = "ti,omap-pcm3168", }, { }, }; MODULE_DEVICE_TABLE(of, omap_pcm3168_of_match); static struct platform_driver omap_pcm3168_driver = { .driver = { .name = "omap-pcm3168", .owner = THIS_MODULE, // .pm = &snd_soc_pm_ops, // .of_match_table = omap_pcm3168_of_match, }, .probe = omap_pcm3168_probe, .remove = __devexit_p(omap_pcm3168_remove), }; module_platform_driver(omap_pcm3168_driver); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("ALSA SoC PCM3168 add on Soundcard"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:omap-pcm3168");

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[alsa-devel] Thinkpad T400 - internal mic doesn't work, can't set CAPTURE - schematics included
by Arkadiusz Miśkiewicz 01 Mar '13

01 Mar '13

Hi, My Thinkpad T400 (2764CTO model) doesn't have ability to set CAPTURE on microphone inputs with alsa. Most likely some quirk is missing. The laptop has internal microphone, microphone in input, headphones out output and 2 internal speakers. That's all. Is uses Conexant CX20561_15Z internally and there is detailed T400 schematic available: http://kythuatphancung.com/download/lenovo-thinkpad-t400-schematic- diagram.html?dl (for audio look around page 43). Kernels 3.7.x, fresh 3.8 - no luck. Tried even linux-next from today but it dies early on alsa unrelated issues. I've tried model options laptop, hp, hp-dv6736, hp-700, toshiba, ideapad, auto. None of these work fine. With model=hp-700 I can set CAPTURE but it doesn't work (silent audio is captured using arecord or in google hangout). External mic connected also doesn't work with these setups. http://www.alsa-project.org/db/?f=394ff4cdca33e75c8a47203d42b89d8de36deca7 [3.8 kernel] http://www.alsa-project.org/db/?f=302877e50f99835075d038207f6db0dfd89873b0 [3.7 kernel] Thinkpad is 2764CTO model exactly. I can test various hacks, patches etc easily. Could someone look at schematic and provide quirks to test? Thanks! -- Arkadiusz Miśkiewicz, arekm / maven.pl

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[alsa-devel] [PATCH v3] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 01 Mar '13

01 Mar '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. For TDM mode, McASP is capable to play or receive 32 channels for one serializer. McAsp has support for max 16 serializer, therefore max channels is 32 * 8. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 66 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 +++++---- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 68 insertions(+), 15 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..0a1af1a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, @@ -1015,8 +1057,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->op_mode = val; ret = of_property_read_u32(np, "tdm-slots", &val); - if (ret >= 0) + if (ret >= 0) { + if (val < 2 || val > 32) { + dev_err(&pdev->dev, + "tdm-slots must be in rage [2-32]\n"); + ret = -EINVAL; + goto nodata; + } + pdata->tdm_slots = val; + } ret = of_property_read_u32(np, "num-serializer", &val); if (ret >= 0) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

3 3

[alsa-devel] [PATCH 1/2] ASoC: wm5102: Correct OUT2 volume and switch names
by Charles Keepax 01 Mar '13

01 Mar '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ab69c83..deb1097 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -755,7 +755,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), @@ -767,7 +767,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: arizona: Don't enable FLL on REFCLK configuration
by Charles Keepax 01 Mar '13

01 Mar '13

Enabling the FLL when REFCLK is being configured is not what the user would expect and can cause issues if SYNCCLK has no specified frequency. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 15 ++++++--------- 1 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6837863..eb25ab9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1148,17 +1148,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (source < 0) return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) + if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (fll->fout) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); if (ret != 0) return ret; if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, Fout); + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, + fll->fout); if (ret != 0) return ret; } @@ -1166,12 +1166,9 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - fll->fout = Fout; - if (Fout) { + if (fll->fout) { arizona_enable_fll(fll, &ref, &sync); - } else { - arizona_disable_fll(fll); } return 0; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] hdajackretask: Fix "advanced override" bug
by David Henningsson 01 Mar '13

01 Mar '13

The values were not correctly retreived from the comboboxes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/main-gtk.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/hdajackretask/main-gtk.c b/hdajackretask/main-gtk.c index 65d0f3a..4227669 100644 --- a/hdajackretask/main-gtk.c +++ b/hdajackretask/main-gtk.c @@ -183,14 +183,14 @@ static void update_user_pin_config(ui_data_t* ui) continue; if (ui->free_overrides) { - int i; + int j; int index; unsigned long val = 0; - for (i = 0; i < FREE_OVERRIDES_COUNT; i++) { - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); - if (index < 0) + for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); + if (index < 0) break; - val += get_free_override_list(i)[index].value; + val += get_free_override_list(j)[index].value; } if (index < 0) continue; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: snd-core: tidyup snd_soc_register_codec() fail case
by Kuninori Morimoto 01 Mar '13

01 Mar '13

kfree() on snd_soc_register_codec() was summarized to one place. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 689eb04..ceac893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3983,8 +3983,8 @@ int snd_soc_register_codec(struct device *dev, /* create CODEC component name */ codec->name = fmt_single_name(dev, &codec->id); if (codec->name == NULL) { - kfree(codec); - return -ENOMEM; + ret = -ENOMEM; + goto fail_codec; } if (codec_drv->compress_type) @@ -4023,7 +4023,7 @@ int snd_soc_register_codec(struct device *dev, reg_size, GFP_KERNEL); if (!codec->reg_def_copy) { ret = -ENOMEM; - goto fail; + goto fail_codec_name; } } } @@ -4057,8 +4057,9 @@ int snd_soc_register_codec(struct device *dev, dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); return 0; -fail: +fail_codec_name: kfree(codec->name); +fail_codec: kfree(codec); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm8960: Correct register 0 and 1 defaults
by Mark Brown 01 Mar '13

01 Mar '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb9273..4cb49eb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -53,8 +53,8 @@ * using 2 wire for device control, so we cache them instead. */ static const struct reg_default wm8960_reg_defaults[] = { - { 0x0, 0x0097 }, - { 0x1, 0x0097 }, + { 0x0, 0x00a7 }, + { 0x1, 0x00a7 }, { 0x2, 0x0000 }, { 0x3, 0x0000 }, { 0x4, 0x0000 }, -- 1.7.10.4

1 2

[alsa-devel] [PATCH] ALSA: hda - Use LPIB position fix for Acer AO751h
by David Henningsson 28 Feb '13

28 Feb '13

The user reports a regression (choppy audio), and that position_fix=1 (or position_fix=4) fixes the issue. This is a Poulsbo controller (8086:811b (rev 07)), so might be worth checking if other Poulsbo machines suffer from the same bug. Cc: stable(a)vger.kernel.org [v3.7+] BugLink: https://bugs.launchpad.net/bugs/1132562 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4cea6bb6..3505f1e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3114,6 +3114,7 @@ static bool check_hdmi_disabled(struct pci_dev *pci) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] = { + SND_PCI_QUIRK(0x1025, 0x0244, "Acer AO751h", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), -- 1.7.9.5

2 5

[alsa-devel] Query about SNDRV_PCM_FORMAT_S24_LE
by Patrick Lai 28 Feb '13

28 Feb '13

Hi, The DSP I am working with currently supports 24-bit PCM sample in 32-bit data space but with LSB padded. Based on the discussion I found on line, SNDRV_PCM_FORMAT_S24_LE is defined to be 24-bit PCM sample in 32-bit data space with MSB padded. Also, there is no format macro defined for the format which my DSP supports. Please correct me if I am wrong. Furthermore, which software layer in audio stack of a typical linux distribution such as ubuntu handles the padding. I don't think it's handled by the alsa library. Does application or pulseaudio handle the padding? I would think it's done in pulseaudio. I have not study pulseaudio codec extensively but would think someone on the mailing list can give a quick answer. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

[alsa-devel] [PATCH] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 27 Feb '13

27 Feb '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..9f33d90 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 10/12] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 27 Feb '13

27 Feb '13

With the generic DMA device tree helper supported by mxs-dma driver, client devices only need to call dma_request_slave_channel() for requesting a DMA channel from dmaengine. Call snd_dmaengine_pcm_open() with NULL filter_fn, and having parameters needed by dma_request_slave_channel() wrap into snd_dma_channel_params and pass it in as filter_data, so that dmaengine_pcm will call the helper rather than dma_request_channel() to request DMA channel. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- .../devicetree/bindings/sound/mxs-saif.txt | 18 ++++---- sound/soc/mxs/mxs-pcm.c | 43 ++------------------ sound/soc/mxs/mxs-pcm.h | 5 --- sound/soc/mxs/mxs-saif.c | 28 ++----------- sound/soc/mxs/mxs-saif.h | 3 +- 5 files changed, 20 insertions(+), 77 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..da221b4 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "saif". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,15 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "saif"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; - fsl,saif-master = <&saif0>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "saif"; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 564b5b6..74cb4b5 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -28,7 +28,6 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dmaengine.h> -#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/initval.h> @@ -39,11 +38,6 @@ #include "mxs-pcm.h" -struct mxs_pcm_dma_data { - struct mxs_dma_data dma_data; - struct mxs_pcm_dma_params *dma_params; -}; - static struct snd_pcm_hardware snd_mxs_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -64,22 +58,6 @@ static struct snd_pcm_hardware snd_mxs_hardware = { }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_data *pcm_dma_data = param; - struct mxs_pcm_dma_params *dma_params = pcm_dma_data->dma_params; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &pcm_dma_data->dma_data; - - return true; -} - static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -91,35 +69,22 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, static int snd_mxs_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mxs_pcm_dma_data *pcm_dma_data; + struct snd_dma_channel_params *params; int ret; - pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL); - if (pcm_dma_data == NULL) - return -ENOMEM; - - pcm_dma_data->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - pcm_dma_data->dma_data.chan_irq = pcm_dma_data->dma_params->chan_irq; - - ret = snd_dmaengine_pcm_open(substream, filter, pcm_dma_data); - if (ret) { - kfree(pcm_dma_data); + params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + ret = snd_dmaengine_pcm_open(substream, NULL, params); + if (ret) return ret; - } snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); - snd_dmaengine_pcm_set_data(substream, pcm_dma_data); - return 0; } static int snd_mxs_close(struct snd_pcm_substream *substream) { - struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream); - snd_dmaengine_pcm_close(substream); - kfree(pcm_dma_data); return 0; } diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 35ba2ca..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,11 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -struct mxs_pcm_dma_params { - int chan_irq; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 3a2aa1d..9f6abb8 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -659,7 +658,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; struct pinctrl *pinctrl; int ret = 0; @@ -721,22 +720,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -753,13 +736,8 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.chan_irq < 0) { - ret = saif->dma_param.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } + saif->dma_param.dev = saif->dev; + saif->dma_param.name = "saif"; platform_set_drvdata(pdev, saif); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..32b4310 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -108,6 +108,7 @@ #define MXS_SAIF_MCLK 0 +#include <sound/dmaengine_pcm.h> #include "mxs-pcm.h" struct mxs_saif { @@ -117,7 +118,7 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; + struct snd_dma_channel_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

1 0

[alsa-devel] [PATCH] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 27 Feb '13

27 Feb '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++------ sound/soc/davinci/davinci-pcm.c | 15 ++++++------ sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..ebd6ee0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -657,12 +657,14 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 stream_serializers = channels/2 + channels%2; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +682,44 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + ser = tx_ser; + } else { + ser = rx_ser; + } + + if (ser < stream_serializers) { + printk(KERN_WARNING "davinci-mcasp: stream has more channels " + "(%d) than are enabled in mcasp (%d)", + channels, ser * 2); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < stream_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < stream_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + 3); } } @@ -812,8 +842,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + davinci_hw_common_param(dev, substream->stream, channels); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +896,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = channels/2 + channels%2; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +971,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..9951973 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -193,7 +194,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) period_size); data_type = prtd->params->data_type; - count = period_size / data_type; + count = period_size / (data_type * serializers); if (fifo_level) count /= fifo_level; @@ -201,8 +202,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,9 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

2 2

[alsa-devel] [PATCH v4 0/3] ASoC davinci/tlv320aic3x updates, fixes & DT support
by Hebbar Gururaja 27 Feb '13

27 Feb '13

1. Convert tlv320aic3x mic bias to a supply widget and related machine driver 2. Add DT support for Davinci machine platform 3. Remove __dev* attributes This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130128 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://github.com/ghebbar/linux.git Branch : dev_next-20130128_davinci_asoc_dt_v3 Changes in v4 - updated pending machine driver to change in micbias -> supply widget - merge codec & machine driver change into a single commit Changes in v3 - drop mcasp pinctrl handler patch from series as it is now handled centrally. - Convert tlv320aic3x mic bias to a supply widget and related machine driver - remove __dev* attributes Changes in v2 - Remove reference to Linux & software details from DT binding Hebbar Gururaja (2): ASoC: tlv320aic3x: Convert mic bias to a supply widget ASoC: davinci: remove __dev* attributes Hebbar, Gururaja (1): ASoC: davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + include/sound/tlv320aic3x.h | 10 ++ sound/soc/codecs/tlv320aic3x.c | 83 ++++++++- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++--- sound/soc/omap/n810.c | 4 +- sound/soc/omap/rx51.c | 8 +- 8 files changed, 318 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

3 5

[alsa-devel] [PATCH 1/2] ASoC: wm5102: Correct AEC loopback enable definiton
by Mark Brown 26 Feb '13

26 Feb '13

Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c1ea75c..efd7317 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1083,7 +1083,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, -- 1.7.10.4

1 4

[alsa-devel] [PATCH - alsa-lib/mixer 1/1] snd_mixer_poll_descriptors_revents should loop over pollfd
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> Signed-off-by: Sander Jansen <s.jansen(a)gmail.com> diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..8deceac 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -743,7 +743,7 @@ int snd_mixer_poll_descriptors_revents(snd_mixer_t *mixer, struct pollfd *pfds, if (nfds == 0) return -EINVAL; res = 0; - for (idx = 0; idx < nfds; idx++) + for (idx = 0; idx < nfds; idx++, pfds++) res |= pfds->revents & (POLLIN|POLLERR|POLLNVAL); *revents = res; return 0; -- 1.8.1.3

2 1

[alsa-devel] [PATCH] Fix: pulse_pcm_poll_revents should return 0 on success
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> --- pulse/pcm_pulse.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 4226d30..0d3704b 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -651,6 +651,8 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, else *revents = 0; + err = 0; + finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); -- 1.8.1.4

1 0

[alsa-devel] [PATCH] pulse_pcm_poll_revents should return 0 on success
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> According to the DOCS, poll_revents should return 0 on success. Fix the pulse plugin to do so. Sander Jansen (1): Fix: pulse_pcm_poll_revents should return 0 on success pulse/pcm_pulse.c | 2 ++ 1 file changed, 2 insertions(+) -- 1.8.1.4

1 0

[alsa-devel] [PATCH] ASoC: dapm: Provide early event callbacks for power up and down
by Mark Brown 26 Feb '13

26 Feb '13

Some devices may benefit from being able to start some parts of the widget power up/down sequence earlier on in the sequence than the point at which the final power state is committed. Support these by providing events which are called before any power state changes are done. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 38fd652..7e4efa5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -311,6 +311,8 @@ struct device; #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ +#define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ +#define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ #define SND_SOC_DAPM_PRE_POST_PMD \ (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 11047ba..b69e9de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1255,6 +1255,14 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, ev_name = "POST_PMD"; power = 0; break; + case SND_SOC_DAPM_WILL_PMU: + ev_name = "WILL_PMU"; + power = 1; + break; + case SND_SOC_DAPM_WILL_PMD: + ev_name = "WILL_PMD"; + power = 0; + break; default: BUG(); return; @@ -1715,6 +1723,14 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); + list_for_each_entry(w, &down_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); + } + + list_for_each_entry(w, &up_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); + } + /* Power down widgets first; try to avoid amplifying pops. */ dapm_seq_run(dapm, &down_list, event, false); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] Allow to switch hardware sampe rate on EMU
by Mihail Zenkov 25 Feb '13

25 Feb '13

Commit 57e5c63007955838043e34c732d224b2cbbb128f "emu10k1: allow to disable the SRC" force hardware use only one rate (48000 hz), but EMU have support two hardware sampling rates (44100 and 48000 hz). Signed-off-by: Mihail Zenkov <mihail.zenkov(a)gmail.com> --- linux-3.8/sound/pci/emu10k1/emupcm.c +++ linux-3.8-1/sound/pci/emu10k1/emupcm.c @@ -1127,7 +1127,7 @@ struct snd_emu10k1_pcm *epcm; struct snd_emu10k1_pcm_mixer *mix; struct snd_pcm_runtime *runtime = substream->runtime; - int i, err; + int i, err, sample_rate; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); if (epcm == NULL) @@ -1146,7 +1146,11 @@ kfree(epcm); return err; } - err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (emu->card_capabilities->emu_model && emu->emu1010.internal_clock == 0) + sample_rate = 44100; + else + sample_rate = 48000; + err = snd_pcm_hw_rule_noresample(runtime, sample_rate); if (err < 0) { kfree(epcm); return err;

3 6

[alsa-devel] M-Audio Audiophile 192 (ice1724)'s broken spdif capture
by Jonas Petersen 25 Feb '13

25 Feb '13

Hi there, I'm trying to get spdif capture to work properly on the M-Audio Audiophile 192 (VT1724/Envy24HT). It is generally working, but I don't get a clean signal. I have "Multi Track Internal Clock" set to "IEC958 In" (spdif in). When I capture via Jack, arecord or Audacity I experience the following two phenomenons (always): 1.) The captured signal ends up 6 dB(FS) to loud (200%)! Everything above -6 dB is distorting. A 1 kHz sine test signal at -12 dB (25%) will be captured as -6 dB (50%). 2.) The left and right channels are shifted by one sample! When I feed a 1 kHz test signal (L+R are _exactly_ the same signal), the right channel will be offset by exactly one sample. Zooming into the waveform clearly shows that. Analyzing the signal with a goniometer shows a (slight but obvious) vertical eliptical shape and not the expected single vertical line. To make sure the hardware actually works fine, I did install a Windows 7 on the exact same machine, installed the drivers from M-Audio and did some recording with audacity. The result is as expected: the -12 dB signal ends up as -12 dB and the left and right channel exactly match each other. So the hardware is willing. I am running a Lubuntu 12.10 and I'm am able to compile and run the current alsa-driver source with the 3.5.0-22 kernel. I played around a bit with the alsa driver source (e.g. pci/ice1712/ice1724.c, pci/ice1712/revo.c) and I am able to compile and load a modified driver. So far I only was able to make the problem worse though. I'd now like to ask for some advice on how to approach the problem. I guess the fact that the left and right channel differ - even though they shouldn't - might be a thing to look for. This must be happening at some stage in the capturing. Is there a way to hook in at different places to narrow down what causes this? Maybe this even already rings somebody's bells? I'll be glad to deliver more information when needed. Thanks and regards, Jonas

3 29

[alsa-devel] [PATCH 1/1] M-Audio Audiophile192: Fix SPDIF input
by Jonas Petersen 25 Feb '13

25 Feb '13

This patch fixes initialization of the AK4114 chip so spdif capture is working properly. Worked out together with Pavel Hofman. Signed-off-by: Jonas Petersen <jnsptrsn1(a)gmail.com> diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 7641080..1112ec1 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -35,6 +35,7 @@ struct revo51_spec { struct snd_i2c_device *dev; struct snd_pt2258 *pt2258; + struct ak4114 *ak4114; }; static void revo_i2s_mclk_changed(struct snd_ice1712 *ice) @@ -359,9 +360,9 @@ static struct snd_ak4xxx_private akm_ap192_priv = { .cif = 0, .data_mask = VT1724_REVO_CDOUT, .clk_mask = VT1724_REVO_CCLK, - .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1, - .cs_addr = VT1724_REVO_CS1, - .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1, + .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS3, + .cs_addr = VT1724_REVO_CS3, + .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS3, .add_flags = VT1724_REVO_CCLK, /* high at init */ .mask_flags = 0, }; @@ -372,7 +373,7 @@ static struct snd_ak4xxx_private akm_ap192_priv = { * CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358) * CSN (pin 35) -- GPIO7 pin 59 */ -#define AK4114_ADDR 0x02 +#define AK4114_ADDR 0x00 static void write_data(struct snd_ice1712 *ice, unsigned int gpio, unsigned int data, int idx) @@ -426,7 +427,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) tmp = snd_ice1712_gpio_read(ice); tmp |= VT1724_REVO_CCLK; /* high at init */ tmp |= VT1724_REVO_CS0; - tmp &= ~VT1724_REVO_CS1; + tmp &= ~VT1724_REVO_CS3; snd_ice1712_gpio_write(ice, tmp); udelay(1); return tmp; @@ -434,7 +435,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp) { - tmp |= VT1724_REVO_CS1; + tmp |= VT1724_REVO_CS3; tmp |= VT1724_REVO_CS0; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -470,27 +471,32 @@ static unsigned char ap192_ak4114_read(void *private_data, unsigned char addr) static int ap192_ak4114_init(struct snd_ice1712 *ice) { static const unsigned char ak4114_init_vals[] = { - AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, + AK4114_RST | AK4114_PWN | AK4114_OCKS0, AK4114_DIF_I24I2S, AK4114_TX1E, - AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(1), + AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(0), 0, 0 }; static const unsigned char ak4114_init_txcsb[] = { 0x41, 0x02, 0x2c, 0x00, 0x00 }; - struct ak4114 *ak; int err; + struct revo51_spec *spec; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + err = snd_ak4114_create(ice->card, ap192_ak4114_read, ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, - ice, &ak); + ice, &spec->ak4114); /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ - ak->check_flags = AK4114_CHECK_NO_RATE; + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; return 0; /* error ignored; it's no fatal error */ } @@ -562,6 +568,9 @@ static int revo_init(struct snd_ice1712 *ice) ice); if (err < 0) return err; + err = ap192_ak4114_init(ice); + if (err < 0) + return err; /* unmute all codecs */ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, @@ -575,7 +584,7 @@ static int revo_init(struct snd_ice1712 *ice) static int revo_add_controls(struct snd_ice1712 *ice) { - struct revo51_spec *spec; + struct revo51_spec *spec = ice->spec; int err; switch (ice->eeprom.subvendor) { @@ -597,7 +606,9 @@ static int revo_add_controls(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; - err = ap192_ak4114_init(ice); + /* only capture SPDIF over AK4114 */ + err = snd_ak4114_build(spec->ak4114, NULL, + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); if (err < 0) return err; break; -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH] ALSA: emu10k1: Load firmware when it was already cached
by Takashi Iwai 25 Feb '13

25 Feb '13

At Sat, 23 Feb 2013 15:36:29 +0100, Florian Zeitz wrote: > > This expands the regression fix from > d28215996b0c3a900411769039aa3c54cf7008ab. > The firmware also needs to be loaded when it was already cached. > > Signed-off-by: Florian Zeitz <florob(a)babelmonkeys.de> Thanks, applied now. Takashi > --- > sound/pci/emu10k1/emu10k1_main.c | 11 ++++++----- > 1 file changed, 6 insertions(+), 5 deletions(-) > > > [2 0001-ALSA-emu10k1-Load-firmware-when-it-was-already-cache.patch <text/x-patch (7bit)>] > diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c > index 5115f11..e6b0166 100644 > --- a/sound/pci/emu10k1/emu10k1_main.c > +++ b/sound/pci/emu10k1/emu10k1_main.c > @@ -860,11 +860,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) > } > snd_printk(KERN_INFO "emu1010: firmware file = %s, size = 0x%zx\n", > filename, emu->firmware->size); > - err = snd_emu1010_load_firmware(emu); > - if (err != 0) { > - snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", filename); > - return err; > - } > + } > + > + err = snd_emu1010_load_firmware(emu); > + if (err != 0) { > + snd_printk(KERN_INFO "emu1010: Loading Firmware failed\n"); > + return err; > } > > /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ >

1 0

[alsa-devel] Bug#701498: hda: Internal mic does not work on Thinkpad T420s (CX20590 codec)
by Ben Hutchings 25 Feb '13

25 Feb '13

On Sat, 2013-02-23 at 20:42 +0100, Adrian Lang wrote: > Package: src:linux > Version: 3.2.35-2 > Severity: normal > > There are two microphones in alsamixer and Gnome 3 audio settings (i. e., > pulseaudio), but I can't get any sound using either of them. I think that's the > same problem as reported to [openSUSE] and [Ubuntu]. > > The problem persists with 3.7 (3.7.8-1~experimental.1) from > experimental. > > [openSUSE] > http://forums.opensuse.org/english/get-technical-help-here/laptop/469210-op… > [Ubuntu] > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/938644 I happen to have a Thinkpad T420 with the same audio codec, and the internal mic works fine. (There does seem to be a problem with external mics, now that I check, but let's ignore that for now.) So this is more board-specific, not chip-specific, behaviour. [...] > ** Model information > sys_vendor: LENOVO > product_name: 4171CTO > product_version: ThinkPad T420s [...] Does the T420s also have a single socket that can be used with either headphones or headset? [I'm copying your alsa-info.txt as an attachment for the benefit of alsa-devel.] Ben. -- Ben Hutchings Absolutum obsoletum. (If it works, it's out of date.) - Stafford Beer

3 2

[alsa-devel] [PATCH 1/5] ASoC: max98090: Remove unneeded version.h header include
by Sachin Kamat 25 Feb '13

25 Feb '13

version.h header file inclusion is not required as detected by versioncheck. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- All patches in this series are build tested and based on for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tree. --- sound/soc/codecs/max98090.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc17604..9ea73aa 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -23,8 +23,6 @@ #include <sound/max98090.h> #include "max98090.h" -#include <linux/version.h> - #define DEBUG #define EXTMIC_METHOD #define EXTMIC_METHOD_TEST -- 1.7.4.1

2 10

[alsa-devel] [ANNOUNCE]: tinycompress: release 0.1.0
by Vinod Koul 24 Feb '13

24 Feb '13

Hi, tinycompress library - the user mode library for using the ALSA compressed APIs has seen some good testing, which brings few fixes and new feature for gapless playback. This library supports both Android and Linux OSes. Along with we have a player, cplay- which can be used for mp3 offload playback Number of fixes and cleanups along the way as seen by below changelog and diffstat. So time to bump up the version to 0.1.0 Charles Keepax (4): compress: Extend limit on error messages compress: Return error when failing to allocate compress struct compress: Return error messages correctly from compress_open compress: Do not put newlines on error messages Namarta Kohli (1): compress: add compress_resume Richard Fitzgerald (12): compress: Add function to get timestamp in samples compress: Change compress_write() to take buffer as const void * compress: Remove unused buffer_size member from struct compress compress: Change running from bitfield to plain int compress: Fix incorrect error message on compress_resume() compress: Remove hardcoded limit on length of poll() wait compress: Handle case of pause() during compress_write() compress: Allow client to pass "don't care" for fragment size/number compress: Block if unable to write all data in single write() compress: Add missing header includes compress: Add makefile for building under standard linux cplay: support auto-configuration of fragment size and count Vinod Koul (9): initial creation compress: fix the timestamp calculation compress: write when we have a fragment to write compress: make variable 'running' as unsigned compress: remove unused variables compress: cache the config values passed tinycompress: add gapless meta data APIs tinycompres: remove unused variable tinycompress: version 0.1.0 Xuemin Su (1): compress: fix arithmetic exception in compress_get_hpointer Android.mk | 22 ++ README | 31 ++- compress.c | 539 +++++++++++++++++++++++++++++++++++ cplay.c | 368 ++++++++++++++++++++++++ include/sound/compress_offload.h | 95 ++++++ include/sound/compress_params.h | 230 +++++++++++++++ include/tinycompress/tinycompress.h | 270 ++++++++++++++++++ include/tinycompress/tinymp3.h | 106 +++++++ include/tinycompress/version.h | 70 +++++ makefile.linux | 35 +++ 10 files changed, 1765 insertions(+), 1 deletions(-) create mode 100644 Android.mk create mode 100644 compress.c create mode 100644 cplay.c create mode 100644 include/sound/compress_offload.h create mode 100644 include/sound/compress_params.h create mode 100644 include/tinycompress/tinycompress.h create mode 100644 include/tinycompress/tinymp3.h create mode 100644 include/tinycompress/version.h create mode 100644 makefile.linux -- ~Vinod

1 0

[alsa-devel] [PATCH] tinycompres: remove unused variable
by Vinod Koul 24 Feb '13

24 Feb '13

cplay.c:210: warning: unused variable i Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- cplay.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/cplay.c b/cplay.c index e52d641..5b4e83e 100644 --- a/cplay.c +++ b/cplay.c @@ -207,7 +207,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, FILE *file; char *buffer; int size, num_read, wrote; - unsigned int i, channels, rate, bits; + unsigned int channels, rate, bits; if (verbose) printf("%s: entry\n", __func__); -- 1.7.0.4

1 1

[alsa-devel] [PATCH] tinycompress: add gapless meta data APIs
by Vinod Koul 24 Feb '13

24 Feb '13

Based on newly introduced kernel API, we add gapless metatdata to the library. Since kernel can recieve only key/value pairs for metadata we add a structure for gapless_metdata and handle this in library Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 71 +++++++++++++++++++++++++++++++++++ include/sound/compress_offload.h | 18 ++++++++- include/tinycompress/tinycompress.h | 37 ++++++++++++++++++ include/tinycompress/version.h | 4 +- 4 files changed, 126 insertions(+), 4 deletions(-) diff --git a/compress.c b/compress.c index d44d5fe..5b600c3 100644 --- a/compress.c +++ b/compress.c @@ -85,6 +85,8 @@ struct compress { char error[COMPR_ERR_MAX]; struct compr_config *config; unsigned int running:1; + unsigned int gapless_metadata; + unsigned int next_track; }; static int oops(struct compress *compress, int e, const char *fmt, ...) @@ -121,6 +123,17 @@ int is_compress_ready(struct compress *compress) return (compress->fd > 0) ? 1 : 0; } +static int get_compress_version(struct compress *compress) +{ + int version = 0; + + if (ioctl(compress->fd, SNDRV_COMPRESS_IOCTL_VERSION, &version)) { + oops(compress, errno, "cant read version"); + return -1; + } + return version; +} + static bool _is_codec_supported(struct compress *compress, struct compr_config *config) { struct snd_compr_caps caps; @@ -212,6 +225,8 @@ struct compress *compress_open(unsigned int card, unsigned int device, return &bad_compress; } + compress->next_track = 0; + compress->gapless_metadata = 0; compress->config = calloc(1, sizeof(*config)); if (!compress->config) goto input_fail; @@ -401,6 +416,62 @@ int compress_drain(struct compress *compress) return 0; } +int compress_partial_drain(struct compress *compress) +{ + if (!is_compress_running(compress)) + return oops(compress, -ENODEV, "device not ready"); + + if (!compress->next_track) + return oops(compress, -EPERM, "next track not signalled"); + if (ioctl(compress->fd, SNDRV_COMPRESS_PARTIAL_DRAIN)) + return oops(compress, errno, "cannot drain the stream\n"); + compress->next_track = 0; + return 0; +} + +int compress_next_track(struct compress *compress) +{ + if (!is_compress_running(compress)) + return oops(compress, -ENODEV, "device not ready"); + + if (!compress->gapless_metadata) + return oops(compress, -EPERM, "metadata not set"); + if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) + return oops(compress, errno, "cannot set next track\n"); + compress->next_track = 1; + compress->gapless_metadata = 0; + return 0; +} + +int compress_set_gapless_metadata(struct compress *compress, + struct compr_gapless_mdata *mdata) +{ + struct snd_compr_metadata metadata; + int version; + + if (!is_compress_ready(compress)) + return oops(compress, -ENODEV, "device not ready"); + + version = get_compress_version(compress); + if (version <= 0) + return -1; + + if (version < SNDRV_PROTOCOL_VERSION(0, 1, 1)) + return oops(compress, -ENXIO, "gapless apis not supported in kernel"); + + metadata.key = SNDRV_COMPRESS_ENCODER_PADDING; + metadata.value[0] = mdata->encoder_padding; + if (ioctl(compress->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compress, errno, "can't set metadata for stream\n"); + + metadata.key = SNDRV_COMPRESS_ENCODER_DELAY; + metadata.value[0] = mdata->encoder_delay; + if (ioctl(compress->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compress, errno, "can't set metadata for stream\n"); + compress->gapless_metadata = 1; + return 0; +} + bool is_codec_supported(unsigned int card, unsigned int device, unsigned int flags, struct snd_codec *codec) { diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h index bbb09bc..17377df 100644 --- a/include/sound/compress_offload.h +++ b/include/sound/compress_offload.h @@ -16,7 +16,7 @@ #include <sound/asound.h> //#include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) struct snd_compressed_buffer { __u32 fragment_size; @@ -64,11 +64,23 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + #define SNDRV_COMPRESS_IOCTL_VERSION _IOR('C', 0x00, int) #define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x10, struct snd_compr_caps) #define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x11, struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14, struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15, struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -76,6 +88,8 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) - +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) #define SND_COMPR_TRIGGER_DRAIN 7 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 8 #endif diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 13dea3e..e129817 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -69,6 +69,11 @@ struct compr_config { struct snd_codec *codec; }; +struct compr_gapless_mdata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + #define COMPRESS_OUT 0x00000000 #define COMPRESS_IN 0x10000000 @@ -165,6 +170,38 @@ int compress_resume(struct compress *compress); int compress_drain(struct compress *compress); /* + * compress_next_track: set the next track for stream + * + * return 0 on success, negative on error + * + * @compress: compress stream to be transistioned to next track + */ +int compress_next_track(struct compress *compress); + +/* + * compress_partial_drain: drain will return after the last frame is decoded + * by DSP and will play the , All the data written into compressed + * ring buffer is decoded + * + * return 0 on success, negative on error + * + * @compress: compress stream to be drain + */ +int compress_partial_drain(struct compress *compress); + +/* + * compress_set_gapless_metadata: set gapless metadata of a compress strem + * + * return 0 on success, negative on error + * + * @compress: compress stream for which metadata has to set + * @mdata: metadata encoder delay and padding + */ + +int compress_set_gapless_metadata(struct compress *compress, + struct compr_gapless_mdata *mdata); + +/* * is_codec_supported:check if the given codec is supported * returns true when supported, false if not * diff --git a/include/tinycompress/version.h b/include/tinycompress/version.h index 89f0f4e..db191d6 100644 --- a/include/tinycompress/version.h +++ b/include/tinycompress/version.h @@ -56,7 +56,7 @@ #define TINYCOMPRESS_LIB_MAJOR 0 /* major number of library version */ #define TINYCOMPRESS_LIB_MINOR 0 /* minor number of library version */ -#define TINYCOMPRESS_LIB_SUBMINOR 1 /* subminor number of library version */ +#define TINYCOMPRESS_LIB_SUBMINOR 2 /* subminor number of library version */ /** library version */ #define TINYCOMPRESS_LIB_VERSION \ @@ -65,6 +65,6 @@ TINYCOMPRESS_LIB_SUBMINOR) /** library version (string) */ -#define TINYCOMPRESS_LIB_VERSION_STR "0.0.1" +#define TINYCOMPRESS_LIB_VERSION_STR "0.0.2" #endif -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ASoC: dapm: Fix handling of loops
by Mark Brown 22 Feb '13

22 Feb '13

Currently if a path loops back on itself we correctly skip over it to avoid going into an infinite loop but this causes us to ignore the need to power up the path as we don't count the loop for the purposes of counting inputs and outputs. This means that internal loopbacks within a device that have powered devices on them won't be powered up. Fix this by treating any path that is currently in the process of being recursed as having a single input or output so that it is counted for the purposes of power decisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 617ac36..38fd652 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -491,6 +491,7 @@ struct snd_soc_dapm_path { /* status */ u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ + u32 walking:1; /* path is in the process of being walked */ u32 weak:1; /* path ignored for power management */ int (*connected)(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1dc5b72..11047ba 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,6 +821,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); + path->walking = 0; return widget->outputs; } } @@ -831,6 +832,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +842,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +852,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +939,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +949,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +959,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } -- 1.7.10.4

2 1

[alsa-devel] [PATCH TINYCOMPRESS 1/14] compress: Add function to get timestamp in samples
by Richard Fitzgerald 22 Feb '13

22 Feb '13

The compress_get_hpointer() converts the timestamp into actual time. But Android needs it in samples. To avoid the inefficiency of using compress_get_hpointer() and converting into time and then back into samples, this change adds compress_get_tstamp() which returns the raw struct snd_compr_tstamp. diff --git a/compress.c b/compress.c index faef982..19b9366 100644 --- a/compress.c +++ b/compress.c @@ -287,6 +287,18 @@ int compress_get_hpointer(struct compress *compress, return 0; } +int compress_get_tstamp(struct compress *compress, + struct snd_compr_tstamp *tstamp ) +{ + if (!is_compress_ready(compress)) + return oops(compress, -ENODEV, "device not ready"); + + if (ioctl(compress->fd, SNDRV_COMPRESS_TSTAMP, tstamp)) + return oops(compress, errno, "cannot get avail"); + + return 0; +} + int compress_write(struct compress *compress, char *buf, unsigned int size) { struct snd_compr_avail avail; diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 13dea3e..8a98527 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -73,6 +73,7 @@ struct compr_config { #define COMPRESS_IN 0x10000000 struct compress; +struct snd_compr_tstamp; /* * compress_open: open a new compress stream @@ -103,6 +104,18 @@ void compress_close(struct compress *compress); */ int compress_get_hpointer(struct compress *compress, unsigned int *avail, struct timespec *tstamp); + + +/* + * compress_get_tstamp: get the raw hw timestamp + * return 0 on success, negative on error + * + * @compress: compress stream on which query is made + * @tstamp: timestamp information + */ +int compress_get_tstamp(struct compress *compress, + struct snd_compr_tstamp *tstamp ); + /* * compress_write: write data to the compress stream * return bytes written on success, negative on error -- 1.7.2.5

3 7

[alsa-devel] [PATCH TINYCOMPRESS 14/14] cplay: support auto-configuration of fragment size and count
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Default the fragment size and count to zero to take advantage of the auto-configuration support that was added to tinycompress. If user doesn't override these by specifying -f and -b values on the command line, they will be passed to tinycompress as zero and tinycompress will query the driver for size and count. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> diff --git a/cplay.c b/cplay.c index 3af6d13..60995a4 100644 --- a/cplay.c +++ b/cplay.c @@ -152,9 +152,9 @@ static int print_time(struct compress *compress) int main(int argc, char **argv) { char *file; - unsigned long buffer_size = 100*1024; + unsigned long buffer_size = 0; int c; - unsigned int card = 0, device = 0, frag = 4; + unsigned int card = 0, device = 0, frag = 0; if (argc < 2) @@ -276,8 +276,14 @@ void play_samples(char *name, unsigned int card, unsigned int device, codec.level = 0; codec.ch_mode = 0; codec.format = 0; - config.fragment_size = buffer_size/frag; - config.fragments = frag; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } config.codec = &codec; compress = compress_open(card, device, COMPRESS_IN, &config); -- 1.7.2.5

2 5

[alsa-devel] [PATCH TINYCOMPRESS 13/14] compress: Prevent divide-by-zero faults in get_hpointer()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Do not assume that the returned snd_compr_avail contains a snd_compr_tstamp with a valid sampling_rate. Check for zero before attempting to use it. diff --git a/compress.c b/compress.c index cfd0219..c0e81a3 100644 --- a/compress.c +++ b/compress.c @@ -297,10 +297,15 @@ int compress_get_hpointer(struct compress *compress, if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); *avail = (unsigned int)kavail.avail; - time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; - tstamp->tv_sec = time; - time = kavail.tstamp.pcm_io_frames % kavail.tstamp.sampling_rate; - tstamp->tv_nsec = time * 1000000000 / kavail.tstamp.sampling_rate; + if (kavail.tstamp.sampling_rate != 0) { + time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; + tstamp->tv_sec = time; + time = kavail.tstamp.pcm_io_frames % kavail.tstamp.sampling_rate; + tstamp->tv_nsec = time * 1000000000 / kavail.tstamp.sampling_rate; + } else { + tstamp->tv_sec = 0; + tstamp->tv_nsec = 0; + } return 0; } -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 12/14] compress: Add makefile for building under standard linux
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> diff --git a/makefile.linux b/makefile.linux new file mode 100644 index 0000000..60d6c04 --- /dev/null +++ b/makefile.linux @@ -0,0 +1,35 @@ +LIB = libtinycompress +BIN = cplay +VER = 0.1 +LIBSRC = compress.c +SRC = cplay.c +LIBOBJ = ${LIBSRC:.c=.o} +OBJ = ${SRC:.c=.o} + +CC = gcc +CROSS_COMPILE = + +CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include +LDFLAGS += -L. -ltinycompress +LIBLDFLAGS = -lasound + +all: libtinycompress cplay + +$(LIB): ${LIBOBJ} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} + +$(BIN): ${OBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + + +%.o: %.c + @echo " CC "$< + @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< + +clean: + @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + + +.PHONY: all clean -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 11/14] compress: Add missing header includes
by Richard Fitzgerald 22 Feb '13

22 Feb '13

This fixes build problems caused by dependencies that were not explicitly included. diff --git a/compress.c b/compress.c index 7ee287d..34936bd 100644 --- a/compress.c +++ b/compress.c @@ -66,6 +66,7 @@ #include <sys/time.h> #include <limits.h> +#include <linux/types.h> #include <linux/ioctl.h> #define __force #define __bitwise diff --git a/cplay.c b/cplay.c index d2a968c..3af6d13 100644 --- a/cplay.c +++ b/cplay.c @@ -51,6 +51,8 @@ * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. */ +#include <stdint.h> +#include <linux/types.h> #include <fcntl.h> #include <errno.h> #include <unistd.h> @@ -60,6 +62,7 @@ #include <signal.h> #include <stdbool.h> #include <getopt.h> +#include <sys/time.h> #define __force #define __bitwise #define __user -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 10/14] compress: Block if unable to write all data in single write()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Previously compress_write() would only block on poll() if the available space < fragment_size. If the device has a small fragment size this could lead to it never blocking and instead looping around doing many small writes. This is bad for power saving. If we were unable to write all the remaining data in a single write we want to block until the device reaches a buffer high water mark, to allow the CPU to sleep. This change will always attempt to issue the first write providing avail >= fragment_size. All subsequent loops will block on poll() before attempting another write() unless there is enough buffer space to write all remaining data. diff --git a/compress.c b/compress.c index 7c8ddc8..7ee287d 100644 --- a/compress.c +++ b/compress.c @@ -317,7 +317,8 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size { struct snd_compr_avail avail; struct pollfd fds; - int to_write, written, total = 0, ret; + int to_write = 0; /* zero indicates we haven't written yet */ + int written, total = 0, ret; const char* cbuf = buf; if (!(compress->flags & COMPRESS_IN)) @@ -333,24 +334,21 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) return oops(compress, errno, "cannot get avail"); - /* we will write only when avail > fragment size */ - if (avail.avail < compress->config.fragment_size) { - /* nothing to write so wait */ - ret = poll(&fds, 1, compress->max_poll_wait_ms); - /* A pause will cause -EBADFD or zero return from driver - * This is not an error, just stop writing + if ( (avail.avail < compress->config.fragment_size) + || ((to_write != 0) && (avail.avail < size)) ) { + /* not enough space for one fragment, or we have done + * a short write and there isn't enough space for all + * the remaining data */ + ret = poll(&fds, 1, compress->max_poll_wait_ms); + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop writing */ if ((ret == 0) || (ret == -EBADFD)) break; if (ret < 0) return oops(compress, errno, "poll error"); if (fds.revents & POLLOUT) { - if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) - return oops(compress, errno, "cannot get avail"); - if (avail.avail == 0) { - oops(compress, -EIO, "woken up even when avail is 0!!!"); - continue; - } + continue; } if (fds.revents & POLLERR) { return oops(compress, -EIO, "poll returned error!"); -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 9/14] compress: Allow client to pass "don't care" for fragment size/number
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Client cannot know what is a sensible fragment size and count for the device unless it has prior knowledge of the precise hardware. To allow for generic clients not hardcoded for particular hardware, this change allows fragment_size and fragments to be passed as 0, meaning "don't care" and tinycompress will choose default values read from the driver. diff --git a/compress.c b/compress.c index 7457550..7c8ddc8 100644 --- a/compress.c +++ b/compress.c @@ -123,19 +123,14 @@ int is_compress_ready(struct compress *compress) return (compress->fd > 0) ? 1 : 0; } -static bool _is_codec_supported(struct compress *compress, struct compr_config *config) +static bool _is_codec_supported(struct compress *compress, struct compr_config *config, + const struct snd_compr_caps *caps) { - struct snd_compr_caps caps; bool codec = false; unsigned int i; - if (ioctl(compress->fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { - oops(compress, errno, "cannot get device caps"); - return false; - } - - for (i = 0; i < caps.num_codecs; i++) { - if (caps.codecs[i] == config->codec->id) { + for (i = 0; i < caps->num_codecs; i++) { + if (caps->codecs[i] == config->codec->id) { /* found the codec */ codec = true; break; @@ -146,24 +141,24 @@ static bool _is_codec_supported(struct compress *compress, struct compr_config * return false; } - if (config->fragment_size < caps.min_fragment_size) { + if (config->fragment_size < caps->min_fragment_size) { oops(compress, -EINVAL, "requested fragment size %d is below min supported %d\n", - config->fragment_size, caps.min_fragment_size); + config->fragment_size, caps->min_fragment_size); return false; } - if (config->fragment_size > caps.max_fragment_size) { + if (config->fragment_size > caps->max_fragment_size) { oops(compress, -EINVAL, "requested fragment size %d is above max supported %d\n", - config->fragment_size, caps.max_fragment_size); + config->fragment_size, caps->max_fragment_size); return false; } - if (config->fragments < caps.min_fragments) { + if (config->fragments < caps->min_fragments) { oops(compress, -EINVAL, "requested fragments %d are below min supported %d\n", - config->fragments, caps.min_fragments); + config->fragments, caps->min_fragments); return false; } - if (config->fragments > caps.max_fragments) { + if (config->fragments > caps->max_fragments) { oops(compress, -EINVAL, "requested fragments %d are above max supported %d\n", - config->fragments, caps.max_fragments); + config->fragments, caps->max_fragments); return false; } @@ -213,9 +208,6 @@ struct compress *compress_open(unsigned int card, unsigned int device, if (!compress || !config) return &bad_compress; - /* take a local copy of config */ - compress->config = *config; - snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); compress->max_poll_wait_ms = DEFAULT_MAX_POLL_WAIT_MS; @@ -236,11 +228,27 @@ struct compress *compress_open(unsigned int card, unsigned int device, oops(compress, errno, "cannot open device '%s'", fn); goto input_fail; } + + struct snd_compr_caps caps; + if (ioctl(compress->fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { + oops(compress, errno, "cannot get device caps"); + goto codec_fail; + } + + /* If caller passed "don't care" fill in default values */ + if ((config->fragment_size == 0) || (config->fragments == 0)) { + config->fragment_size = caps.min_fragment_size; + config->fragments = caps.max_fragments; + } + + /* take a local copy of config */ + compress->config = *config; + #if 0 /* FIXME need to turn this On when DSP supports * and treat in no support case */ - if (_is_codec_supported(compress, config) == false) { + if (_is_codec_supported(compress, config, &caps) == false) { oops(compress, errno, "codec not supported\n"); goto codec_fail; } diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 1ab53c6..1309d65 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -58,6 +58,8 @@ extern "C" { #endif /* * struct compr_config: config structure, needs to be filled by app + * If fragment_size or fragments are zero, this means "don't care" + * and tinycompress will choose values that the driver supports * * @fragment_size: size of fragment requested, in bytes * @fragments: number of fragments @@ -78,11 +80,14 @@ struct snd_compr_tstamp; /* * compress_open: open a new compress stream * returns the valid struct compress on success, NULL on failure + * If config does not specify a requested fragment size, on return + * it will be updated with the size and number of fragments that + * were configured * * @card: sound card number * @device: device number * @flags: device flags can be COMPRESS_OUT or COMPRESS_IN - * @config: stream config requested + * @config: stream config requested. Returns actual fragment config */ struct compress *compress_open(unsigned int card, unsigned int device, unsigned int flags, struct compr_config *config); -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 8/14] compress: Handle case of pause() during compress_write()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

If stream is paused() during a compress_write() the poll() will return 0 or -EBADFD, or the write() will return -EBADFD. This is not an error so compress_write() should not pass an error code to the caller. It should abort the write and then return the number of bytes it had written up to that point. diff --git a/compress.c b/compress.c index b706f8b..7457550 100644 --- a/compress.c +++ b/compress.c @@ -329,10 +329,13 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (avail.avail < compress->config.fragment_size) { /* nothing to write so wait */ ret = poll(&fds, 1, compress->max_poll_wait_ms); + /* A pause will cause -EBADFD or zero return from driver + * This is not an error, just stop writing + */ + if ((ret == 0) || (ret == -EBADFD)) + break; if (ret < 0) return oops(compress, errno, "poll error"); - if (ret == 0) - return oops(compress, -EPIPE, "Poll timeout, Broken Pipe"); if (fds.revents & POLLOUT) { if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) return oops(compress, errno, "cannot get avail"); @@ -351,6 +354,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size else to_write = size; written = write(compress->fd, cbuf, to_write); + /* If play was paused the write returns -EBADFD */ + if (written == -EBADFD) + break; if (written < 0) return oops(compress, errno, "write failed!\n"); -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 7/14] compress: Remove hardcoded limit on length of poll() wait
by Richard Fitzgerald 22 Feb '13

22 Feb '13

For best power-saving we want to sleep on poll() for as long as possible, we don't want to wake unnecessarily for arbitrary time limits. Adds function compress_set_max_poll_wait() so that client can configure the poll() timeout. diff --git a/compress.c b/compress.c index 5235ea4..b706f8b 100644 --- a/compress.c +++ b/compress.c @@ -77,12 +77,16 @@ #define COMPR_ERR_MAX 32 +/* Default maximum time we will wait in a poll() - 20 seconds */ +#define DEFAULT_MAX_POLL_WAIT_MS 20000 + struct compress { int fd; unsigned int flags; char error[COMPR_ERR_MAX]; struct compr_config config; int running; + int max_poll_wait_ms; }; static int oops(struct compress *compress, int e, const char *fmt, ...) @@ -214,6 +218,8 @@ struct compress *compress_open(unsigned int card, unsigned int device, snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); + compress->max_poll_wait_ms = DEFAULT_MAX_POLL_WAIT_MS; + compress->flags = flags; if (!((flags & COMPRESS_OUT) || (flags & COMPRESS_IN))) { oops(compress, -EINVAL, "can't deduce device direction from given flags\n"); @@ -321,8 +327,8 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size /* we will write only when avail > fragment size */ if (avail.avail < compress->config.fragment_size) { - /* nothing to write so wait for 10secs */ - ret = poll(&fds, 1, 1000000); + /* nothing to write so wait */ + ret = poll(&fds, 1, compress->max_poll_wait_ms); if (ret < 0) return oops(compress, errno, "poll error"); if (ret == 0) @@ -432,3 +438,8 @@ bool is_codec_supported(unsigned int card, unsigned int device, return ret; } +void compress_set_max_poll_wait(struct compress *compress, int milliseconds) +{ + compress->max_poll_wait_ms = milliseconds; +} + diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index e285bd4..1ab53c6 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -189,6 +189,16 @@ int compress_drain(struct compress *compress); bool is_codec_supported(unsigned int card, unsigned int device, unsigned int flags, struct snd_codec *codec); +/* + * compress_set_max_poll_wait: set the maximum time tinycompress + * will wait for driver to signal a poll(). Interval is in + * milliseconds. + * Pass interval of -1 to disable timeout and make poll() wait + * until driver signals. + * If this function is not used the timeout defaults to 20 seconds. + */ +void compress_set_max_poll_wait(struct compress *compress, int milliseconds); + int is_compress_running(struct compress *compress); int is_compress_ready(struct compress *compress); -- 1.7.2.5

2 2

[alsa-devel] [PATCH TINYCOMPRESS 6/14] compress: Fix incorrect error message on compress_resume()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Was reporting compress_resume() errors as "cannot pause" diff --git a/compress.c b/compress.c index 871c19e..5235ea4 100644 --- a/compress.c +++ b/compress.c @@ -392,7 +392,7 @@ int compress_pause(struct compress *compress) int compress_resume(struct compress *compress) { if (ioctl(compress->fd, SNDRV_COMPRESS_RESUME)) - return oops(compress, errno, "cannot pause the stream\n"); + return oops(compress, errno, "cannot resume the stream\n"); return 0; } -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 5/14] compress: Take local copy of compr_config structure
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Take a copy of the compr_config structure passed by the client. This allows the client to use a structure in temporary memory and avoids the possibility that tinycompress can be broken by the client changing the content of its compr_config. diff --git a/compress.c b/compress.c index c4f215c..871c19e 100644 --- a/compress.c +++ b/compress.c @@ -81,7 +81,7 @@ struct compress { int fd; unsigned int flags; char error[COMPR_ERR_MAX]; - struct compr_config *config; + struct compr_config config; int running; }; @@ -209,7 +209,8 @@ struct compress *compress_open(unsigned int card, unsigned int device, if (!compress || !config) return &bad_compress; - compress->config = config; + /* take a local copy of config */ + compress->config = *config; snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); @@ -319,7 +320,7 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size return oops(compress, errno, "cannot get avail"); /* we will write only when avail > fragment size */ - if (avail.avail < compress->config->fragment_size) { + if (avail.avail < compress->config.fragment_size) { /* nothing to write so wait for 10secs */ ret = poll(&fds, 1, 1000000); if (ret < 0) -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 4/14] compress: Change running from bitfield to plain int
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Declaring running as a single-bit bitfield does not save any space or give any coding advantages, but does leave potential opportunities for the compiler to be inefficient in how it handles bitfields compared to how it would handle a machine int. diff --git a/compress.c b/compress.c index fa2d43f..c4f215c 100644 --- a/compress.c +++ b/compress.c @@ -82,7 +82,7 @@ struct compress { unsigned int flags; char error[COMPR_ERR_MAX]; struct compr_config *config; - int running:1; + int running; }; static int oops(struct compress *compress, int e, const char *fmt, ...) -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 3/14] compress: Remove unused buffer_size member from struct compress
by Richard Fitzgerald 22 Feb '13

22 Feb '13

diff --git a/compress.c b/compress.c index ee11988..fa2d43f 100644 --- a/compress.c +++ b/compress.c @@ -80,7 +80,6 @@ struct compress { int fd; unsigned int flags; - __u64 buffer_size; char error[COMPR_ERR_MAX]; struct compr_config *config; int running:1; -- 1.7.2.5

1 1

[alsa-devel] [PATCH TINYCOMPRESS 2/14] compress: Change compress_write() to take buffer as const void *
by Richard Fitzgerald 22 Feb '13

22 Feb '13

The input buffer to compress_write() should be const, and is better defined as void * so the caller doesn't have to cast. diff --git a/compress.c b/compress.c index 19b9366..ee11988 100644 --- a/compress.c +++ b/compress.c @@ -299,11 +299,12 @@ int compress_get_tstamp(struct compress *compress, return 0; } -int compress_write(struct compress *compress, char *buf, unsigned int size) +int compress_write(struct compress *compress, const void *buf, unsigned int size) { struct snd_compr_avail avail; struct pollfd fds; int to_write, written, total = 0, ret; + const char* cbuf = buf; if (!(compress->flags & COMPRESS_IN)) return oops(compress, -EINVAL, "Invalid flag set"); @@ -343,12 +344,12 @@ int compress_write(struct compress *compress, char *buf, unsigned int size) to_write = avail.avail; else to_write = size; - written = write(compress->fd, buf, to_write); + written = write(compress->fd, cbuf, to_write); if (written < 0) return oops(compress, errno, "write failed!\n"); size -= written; - buf += written; + cbuf += written; total += written; } return total; diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 8a98527..e285bd4 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -125,7 +125,7 @@ int compress_get_tstamp(struct compress *compress, * @buf: pointer to data * @size: number of bytes to be written */ -int compress_write(struct compress *compress, char *buf, unsigned int size); +int compress_write(struct compress *compress, const void *buf, unsigned int size); /* * compress_read: read data from the compress stream -- 1.7.2.5

1 1

[alsa-devel] [PATCH] Fix regression in emu1010 firmware loading
by Mihail Zenkov 22 Feb '13

22 Feb '13

This patch fix regression in emu1010 firmware loading after http://git.alsa-project.org/?p=alsa-kprivate.git;a=commitdiff;h=b209c4dfcd9… I just revert small part of this commit. Tested on emu1212m pci.

2 3

[alsa-devel] Axago ADA-X5
by Pavol Lajciak 22 Feb '13

22 Feb '13

Hello. I have USB connected Axago ADA-X5 http://www.axago.eu/cz/produkty/usb-adaptery/ADA-X5.html Chipset CMedia CM6206. Chipset is incorrectly detected as CM106. I have connected this device to speakers by optical toslik (spdif) connection. I hear in equal interval (period) reapeated jitter, when sound is playing. I use Advanced Linux Sound Architecture Driver Version 1.0.24. and Ubuntu 12.04.2 Here is ID of device - incorrectly detected chipset. Bus 002 Device 004: ID 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device S úctou Pavol Lajčiak

1 0

[alsa-devel] [PATCH] ALSA: hda - hdmi: Make jacks phantom, if they're not detectable
by David Henningsson 22 Feb '13

22 Feb '13

Just as for analog codecs, a jack that isn't suitable for detection (in this case, NO_PRESENCE was set) should be a phantom Jack instead of a normal one. Thanks to Raymond Yau for spotting. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/961286 BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=903869 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 21425fb..78e1827 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1640,6 +1640,9 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) if (pcmdev > 0) sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); + if (!is_jack_detectable(codec, per_pin->pin_nid)) + strncat(hdmi_str, " Phantom", + sizeof(hdmi_str) - strlen(hdmi_str) - 1); return snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str, 0); } -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ALSA: usb: Fix Processing Unit Descriptor parsers
by Takashi Iwai 21 Feb '13

21 Feb '13

At Thu, 21 Feb 2013 01:55:50 +0000, Pawel Moll wrote: > > Commit 99fc86450c439039d2ef88d06b222fd51a779176 "ALSA: usb-mixer: > parse descriptors with structs" introduced a set of useful parsers > for descriptors. Unfortunately the parses for the Processing Unit > Descriptor came with a very subtle bug... > > Functions uac_processing_unit_iProcessing() and > uac_processing_unit_specific() were indexing the baSourceID array > forgetting the fields before the iProcessing and process-specific > descriptors. > > The problem was observed with Sound Blaster Extigy mixer, > where nNrModes in Up/Down-mix Processing Unit Descriptor > was accessed at offset 10 of the descriptor (value 0) > instead of offset 15 (value 7). In result the resulting > control had interesting limit values: > > Simple mixer control 'Channel Routing Mode Select',0 > Capabilities: volume volume-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: 0 - -1 > Mono: -1 [100%] > > Fixed by starting from the bmControls, which was calculated > correctly, instead of baSourceID. > > Now the mentioned control is fine: > > Simple mixer control 'Channel Routing Mode Select',0 > Capabilities: volume volume-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: 0 - 6 > Mono: 0 [0%] > > Signed-off-by: Pawel Moll <mail(a)pawelmoll.com> > --- > include/uapi/linux/usb/audio.h | 6 ++++-- > 1 file changed, 4 insertions(+), 2 deletions(-) > > diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h > index ac90037..d2314be 100644 > --- a/include/uapi/linux/usb/audio.h > +++ b/include/uapi/linux/usb/audio.h > @@ -384,14 +384,16 @@ static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_de > int protocol) > { > __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); > - return desc->baSourceID[desc->bNrInPins + control_size]; > + return *(uac_processing_unit_bmControls(desc, protocol) > + + control_size); > } > > static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc, > int protocol) > { > __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); > - return &desc->baSourceID[desc->bNrInPins + control_size + 1]; > + return uac_processing_unit_bmControls(desc, protocol) > + + control_size + 1; > } Looks good to me. An alternative would be to change uac_processing_unit_bControlSize() to correct the header size, but I don't think it's good, as bControlSize fields in other units show the value including the header size. Daniel, what do you think? thanks, Takashi

2 2

[alsa-devel] [PATCH 1/4] compress: make variable 'running' as unsigned
by Vinod Koul 21 Feb '13

21 Feb '13

as we never expect this to be negative Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index 19d618a..a8964f4 100644 --- a/compress.c +++ b/compress.c @@ -83,7 +83,7 @@ struct compress { __u64 buffer_size; char error[COMPR_ERR_MAX]; struct compr_config *config; - int running:1; + unsigned int running:1; }; static int oops(struct compress *compress, int e, const char *fmt, ...) -- 1.7.0.4

1 4

[alsa-devel] [PATCH RFC 000/112] HD-audio generic parser improvements
by Takashi Iwai 21 Feb '13

21 Feb '13

Hi, this is a large series of patches to implement the really generic HD-audio codec parser usable for various codecs. It's based on the Realtek codec parser, so I started hacking patch_realtek.c for more generalization and improvements. Then it's merged to the existing generic parser code, hda_generic.c, and let Realtek parser just uses this new parser code. In the end, patch_realtek.c contains only the device specific fixups and the call of generic parser functions. The series contains a few improvements in the HD-audio core code, some of which aren't really dependent on the generic parser code. They are just there as I've developed this for a couple of weeks. After this patchset, most of other codecs can be migrated as well. Currently analog, ca0110, cirrus, cmedia, conexant and via codecs have been already rewritten (i.e. the own parser codes dropped). patch_sigmatel.c is still not converted yet, but can be done soon later. I'm going to send the conversion patches for these codecs right after this series. I don't expect to merge all conversions of codec parsers in 3.9 kernel, but at least, the conversion of Realtek and this generic parser code addition can be merged to 3.9. Maybe minor and easy ones like ca0110 and cmedia can be merged as well. Takashi

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[alsa-devel] Discovery of device characteristics
by Tim Hewett 21 Feb '13

21 Feb '13

I'm writing software looking to discover the available sample rates, channel formats etc. for a device. I have been taking the approach of testing a sample rate from a list of potential rates, then setting the device to that rate to then test the channel formats etc.. The structure is initialised using snd_pcm_hw_params_any() then snd_pcm_hw_params_test_rate() is called for a set of rates followed by snd_pcm_hw_params_set_rate() if the previous call is successful, then snd_pcm_hw_params_test_format() for a set of formats followed by snd_pcm_hw_params_set_format() if that is successful, then snd_pcm_hw_params_test_channels(), the point being to narrow down the characteristics available for given sample rates and channel formats. For example I have a device which supports up to 8 channels at sample rates of 44.1kHz and 48kHz but only 2 channels at 96kHz. It works ok usually but keeps failing to detect that the device supports 24-bit channel formats, both for SND_PCM_FORMAT_S24_LE/BE and SND_PCM_FORMAT_S24_3LE/3BE. It detects the 16-bit formats without any problem. Online source taken from http://download.atmark-techno.com/misc/demos/usb-audio-a500-howto/hw_params… does detect the 24-bit format but doesn't set the sample rate first - each characteristic is tested in isolation - snd_pcm_hw_params_any() is called before each one. I am wondering if my way is not the correct way to determine device characteristics.

2 3

Re: [alsa-devel] HD-audio generic parser merged to linux-next (desktop with internal speaker)
by Raymond Yau 21 Feb '13

21 Feb '13

> > > > > > > > hda-emu codecs/alc662-intel-d945gclf2 > > > > > > But it's irrelevant from whether you enable independent HP mode or > > > not. It's just the policy of badness evaluation. It's possible to > > > have a different result by adjusting the badness parameters. > > > > IOW, the patch like below will make surrounds back for the device, > > for the cost of losing individual headphone volume control. > > > > It seems that it would make more sense for having surrounds for such > > desktop setups, and the patch doesn't appear to affect too severely. > > Only ALC662/663 and one CMI9880 configs have been changed by this, and > > all look reasonable. So I'm going to apply it. > > BTW, I should mention beforehand that the previous patch won't suffice > for the device like ThinkCenter A58 where the headphone can't be > shared with the primary DAC. For allowing the surrounds on such > machines, we need to implement more complex workaround (i.e. make > surrounds exclusive, and turn off others while using surrounds). http://www.intel.com/content/www/us/en/motherboards/desktop-motherboards/de… Realtek ALC662* codec for 2+2ch (multi-streaming) Intel® High Definition Audio (Intel® HD Audio): Front panel microphone/headphone header S/PDIF digital audio header Internal mono speaker header https://launchpadlibrarian.net/92117241/Card0.Codecs.codec.0.txt The result of hda-emu seem OK for ThinkCenter M55 (AD1988) which have volume controls for the internal speaker https://bugzilla.redhat.com/attachment.cgi?id=335481 But have same problem with those 6-channel HDA codec M57/M57P(AD1882) https://launchpadlibrarian.net/63326460/Card0.Codecs.codec.2.txt

1 0

[alsa-devel] [PATCH v2 0/8] Add support for directly setting the FLL REFCLK
by Charles Keepax 20 Feb '13

20 Feb '13

This patch series adds support for specifying the source of REFCLK directly for Wolfson Arizona based devices. This should allow for improved performance by using the highest available REFCLK source. Should the REFCLK source not be directly specified then the patches will default to the older behaviour of using the 32kHz clock for REFCLK. REFCLK is configured using a call to snd_soc_codec_set_pll specifying one of the new WM51xx_FLLx_REFCLK defines as the pll_id. It is valid to clear the SYNCCLK source by setting the source to ARIZONA_FLL_SRC_NONE, however once a REFCLK source it can only be replaced with another valid REFCLK source. Charles Keepax (8): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK sound/soc/codecs/arizona.c | 191 ++++++++++++++++++++++++++++++-------------- sound/soc/codecs/arizona.h | 9 ++- sound/soc/codecs/wm5102.c | 6 ++ sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 6 ++ sound/soc/codecs/wm5110.h | 6 +- 6 files changed, 158 insertions(+), 66 deletions(-) -- 1.7.2.5

2 9

[alsa-devel] [PATCH] ASoC: max98088: Add TLV data for volume controls.
by Dylan Reid 20 Feb '13

20 Feb '13

Specify volumes as defined in the MAX98088/9 data sheet. Allows ALSA lib snd_mixer_selem_get_playback_dB_range and related functions to work. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index a4c16fd..3a7b7fd 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = { 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), }; +static const unsigned int max98088_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98088_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + static const struct snd_kcontrol_new max98088_snd_controls[] = { - SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L, - M98088_REG_3A_LVL_HP_R, 0, 31, 0), - SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L, - M98088_REG_3E_LVL_SPK_R, 0, 31, 0), - SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L, - M98088_REG_3C_LVL_REC_R, 0, 31, 0), + SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv), + SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv), SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L, M98088_REG_3A_LVL_HP_R, 7, 1, 1), -- 1.8.0

2 1

[alsa-devel] [PATCH 1/6] ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init
by Stephen Warren 20 Feb '13

20 Feb '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra only supports, and always enables, device tree. Remove all runtime checks for DT support from the driver. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index ba419f8..49861c6 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -176,11 +176,7 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (!dev->of_node) - /* non-DT is always Tegra20 */ - data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else - /* DT boot, but unknown SoC */ return -EINVAL; data->clk_pll_a = clk_get_sys(NULL, "pll_a"); -- 1.7.10.4

3 7

[alsa-devel] [PATCH 0/8] ASoC: arizona: Add support for directly setting the FLL REFCLK
by Charles Keepax 20 Feb '13

20 Feb '13

This patch series adds support for specifying the source of REFCLK directly for Wolfson Arizona based devices. This should allow for improved performance by using the highest available REFCLK source. Should the REFCLK source not be directly specified then the patches will default to the older behaviour of using the 32kHz clock for REFCLK. REFCLK is configured using a call to snd_soc_codec_set_pll specifying one of the new WM51xx_FLLx_REFCLK defines as the pll_id. It is valid to clear the SYNCCLK source by setting the source to ARIZONA_FLL_SRC_NONE, however once a REFCLK source it can only be replaced with another valid REFCLK source. Charles Keepax (8): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK sound/soc/codecs/arizona.c | 191 ++++++++++++++++++++++++++++++-------------- sound/soc/codecs/arizona.h | 9 ++- sound/soc/codecs/wm5102.c | 6 ++ sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 6 ++ sound/soc/codecs/wm5110.h | 6 +- 6 files changed, 158 insertions(+), 66 deletions(-) -- 1.7.2.5

2 12

[alsa-devel] [PATCH v5 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Just more fixups according to Takashi's comments. David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 52 +++++++++++----------- sound/pci/hda/hda_local.h | 28 +++++++----- sound/pci/hda/patch_hdmi.c | 105 ++++++++++++++++++++++++++++++++++---------- 3 files changed, 127 insertions(+), 58 deletions(-) -- 1.7.9.5

1 5

[alsa-devel] [PATCH v4 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Changes since v3: * snd_hdmi_get_eld now does not take a full hdmi_eld struct David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 52 +++++++++++----------- sound/pci/hda/hda_local.h | 28 +++++++----- sound/pci/hda/patch_hdmi.c | 102 ++++++++++++++++++++++++++++++++++---------- 3 files changed, 124 insertions(+), 58 deletions(-) -- 1.7.9.5

2 8

[alsa-devel] [PATCH v3 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Changes since v2: Added refactoring patch to split hdmi_eld into hdmi_eld and parsed_hdmi_eld. David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 47 ++++++++++++++---------- sound/pci/hda/hda_local.h | 23 +++++++----- sound/pci/hda/patch_hdmi.c | 86 +++++++++++++++++++++++++++++++++----------- 3 files changed, 107 insertions(+), 49 deletions(-) -- 1.7.9.5

2 8

[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by wendelin klimann 19 Feb '13

19 Feb '13

hello I try to implement an PCM3168 audio codec from TI to the BeagleBoard but actually i am stuck and it would be rally nice if one could help me a little bit. My ALSA driver version is 1.0.24 and i use an 3.2.18 kernel. As a startingpoint i used the files */linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3008.c* */linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3008.h* The files (see end of e-mail) */linux-mainline-3.2.18-r121b/git/sound/soc/omap/pcm3168_soundcard.c /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.c /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.h* could be compiled successfully and also the insmod loads the modules *$: insmod /lib/modules/3.2.18/kernel/sound/soc/omap/snd-soc-pcm3168_soundcard.ko* *$: lsmod* Module Size Used by snd_soc_pcm3168 1886 0 snd_soc_pcm3168_soundcard 2094 0 twl4 030_madc_hwmon 2569 0 snd_soc_omap3beagle 2239 0 rtc_twl 4382 0 snd_soc_twl4030 37616 1 rtc_ds1307 6642 0 snd_soc_omap 3521 1 snd_soc_omap_mcbsp 8374 1 snd_soc_core 107346 6 snd_soc_pcm3168,snd_soc_pcm3168_soundcard,snd_soc_omap3beagle,snd_soc_twl4030,snd_soc_omap,snd_soc_omap_mcbsp regmap_spi 1263 1 snd_soc_core snd_pcm 78039 4 snd_soc_twl4030,snd_soc_omap,snd_soc_omap_mcbsp,snd_soc_core snd_timer 18848 1 snd_pcm snd 57930 3 snd_soc_core,snd_pcm,snd_timer soundcore 6954 1 snd iptable_nat 3819 0 snd_page_alloc 4979 1 snd_pcm nf_nat 16515 1 iptable_nat nf_conntrack_ipv4 12018 3 iptable_nat,nf_nat nf_conntrack 71219 3 iptable_nat,nf_nat,nf_conntrack_ipv4 nf_defrag_ipv4 1379 1 nf_conntrack_ipv4 ip_tables 10537 1 iptable_nat x_tables 16505 2 iptable_nat,ip_tables gpio_keys 7117 0 autofs4 20492 2 but the new ALSA device is not recognized *$:aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: omap3beagle [omap3beagle], device 0: TWL4030 twl4030-hifi-0 [] Subdevices : 1/1 Subdevice #0: subdevice #0 So i tried to get some information in *$: cd /sys/kernel/debug/asoc* */sys/kernel/debug/asoc$: ls* codecs dais omap3beagle pcm3168 platforms */sys/kernel/debug/asoc$: cat codecs* pcm3168-codec.0 twl4030-codec */sys/kernel/debug/asoc$: cat dais* pcm3168-hifi twl4030-voice twl4030-hifi omap-mcbsp-dai.4 omap-mcbsp-dai.3 omap-mcbsp-dai.2 omap-mcbsp-dai.1 omap-mcbsp-dai.0 */sys/kernel/debug/asoc$: cat platforms* omap-pcm-audio snd-soc-dummy */sys/kernel/debug/asoc$:cd omap3beagle* */sys/kernel/debug/asoc/omap3beagle$:ls* dapm dapm_pop_time twl4030-codec */sys/kernel/debug/asoc/omap3beagle$:cd /sys/kernel/debug/asoc/pcm3168* */sys/kernel/debug/asoc/pcm3168$: ls* dapm_pop_time At this point i have no idea how to solve the problem or how to debug further and i would be very grateful for any hint. Thanks a lot Wendelin * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/omap/pcm3168_soundcard.c * ** *********************************************************************************************** */* * PCM3168 ASoC driver for BeagleBoard. * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> //#include <asm/dma.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <mach/edma.h> #include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/pcm3168.h" /* * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ /* * SND_SOC_DAIFMT_CBS_CFS * for codec in slave mode */ #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | \ SND_SOC_DAIFMT_IB_NF) static int pcm3168_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int fs; int ret = 0; printk("in hw_params -> by kli"); /* Fsref can be 8000 to 96000. */ fs = params_rate(params); /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); return ret; } pr_debug("pcm3168_hw_params: rate = %d Hz\n", fs); return 0; } static struct snd_soc_ops pcm3168_ops = { .hw_params = pcm3168_hw_params, }; /* omap-pcm3168 digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link pcm3168_dai = { .name = "PCM3168", /* Codec name */ .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "pcm3168-hifi", .codec_name = "pcm3168-codec", .platform_name = "omap-pcm-audio", .ops = &pcm3168_ops, }; /* omap-pcm3168 audio machine driver */ static struct snd_soc_card snd_soc_pcm3168 = { .name = "pcm3168", // -> "OMAP pcm3168" .owner = THIS_MODULE, .dai_link = &pcm3168_dai, .num_links = 1, }; /* pcm3168 audio private data */ static struct pcm3168_setup_data pcm3168_pcm3168_setup = { // .dem0_pin = GPIO(45), // .dem1_pin = GPIO(46), // .pdad_pin = GPIO(47), // .pdda_pin = GPIO(38), }; struct platform_device pcm3168_codec = { .name = "pcm3168-codec", .id = 0, .dev = { .platform_data = &pcm3168_pcm3168_setup, }, }; static struct platform_device *pcm3168_snd_device; static int __init pcm3168_init(void) { int ret; printk(KERN_ALERT "in init soundcard \n"); if (!(machine_is_omap3_beagle())) return -ENODEV; platform_device_register(&pcm3168_codec); pcm3168_snd_device = platform_device_alloc("soc-audio", 0); if (!pcm3168_snd_device) { printk(KERN_ERR "platform device allocation failed\n"); return -ENOMEM; } platform_set_drvdata(pcm3168_snd_device, &snd_soc_pcm3168); // platform_device_add_data(pcm3168_snd_device, &pcm3168_snd_data, // sizeof(pcm3168_snd_data)); /* ret = platform_device_add_resources(pcm3168_snd_device, pcm3168_snd_resources, ARRAY_SIZE(pcm3168_snd_resources)); if (ret) { printk(KERN_ERR "platform device add resources failed\n"); goto error; } */ ret = platform_device_add(pcm3168_snd_device); if (ret) goto error; return ret; error: platform_device_put(pcm3168_snd_device); return ret; } static void __exit pcm3168_exit(void) { platform_device_unregister(pcm3168_snd_device); platform_device_unregister(&pcm3168_codec); } module_init(pcm3168_init); module_exit(pcm3168_exit); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("PCM3168 Beagle ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:pcm3168-soc-audio"); * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.c * ** *********************************************************************************************** */* * ALSA Soc PCM3168 codec support * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin * * * Based on AC97 Soc codec, original copyright follow: * Copyright 2005 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * * Generic PCM3168 support. */ #include <linux/init.h> #include <linux/kernel.h> #include <linux/device.h> #include <linux/gpio.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> //#include <sound/soc-dapm.h> #include <sound/soc.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include <plat/mcbsp.h> #include "../omap/omap-mcbsp.h" #include "../omap/omap-pcm.h" #include "pcm3168.h" #define PCM3168_VERSION "0.1" #define PCM3168_RATES SNDRV_PCM_RATE_8000_96000 /*(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000)*/ static struct snd_soc_dai_driver pcm3168_dai = { .name = "pcm3168-hifi", .playback = { .stream_name = "PCM3168 Playback", .channels_min = 2, .channels_max = 8, .rates = PCM3168_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE }, .capture = { .stream_name = "PCM3168 Capture", .channels_min = 2, .channels_max = 6, .rates = PCM3168_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE }, }; static void pcm3168_gpio_free(struct pcm3168_setup_data *setup) { // gpio_free(setup->dem0_pin); // gpio_free(setup->dem1_pin); // gpio_free(setup->pdad_pin); // gpio_free(setup->pdda_pin); } static int pcm3168_soc_probe(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; int ret = 0; printk(KERN_ALERT "in SOC Probe -> kli \n"); printk(KERN_INFO "PCM3168 SoC Audio Codec %s\n", PCM3168_VERSION); return ret; gpio_err: // pcm3168_gpio_free(setup); return ret; } static int pcm3168_soc_remove(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; // pcm3168_gpio_free(setup); return 0; } #define pcm3168_soc_suspend NULL #define pcm3168_soc_resume NULL static struct snd_soc_codec_driver soc_codec_dev_pcm3168 = { .probe = pcm3168_soc_probe, .remove = pcm3168_soc_remove, .suspend = pcm3168_soc_suspend, .resume = pcm3168_soc_resume, }; static int __devinit pcm3168_codec_probe(struct platform_device *pdev) { int ret; printk(KERN_ALERT "probe pcm3168 -> kli \n"); ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3168, &pcm3168_dai, 1); printk(KERN_ALERT "probe_after pcm3168 -> kli = %d \n", ret); return ret; } static int __devexit pcm3168_codec_remove(struct platform_device *pdev) { printk(KERN_ALERT "remove pcm3168 -> kli \n"); snd_soc_unregister_codec(&pdev->dev); return 0; } MODULE_ALIAS("platform:pcm3168-codec"); static struct platform_driver pcm3168_codec_driver = { .probe = pcm3168_codec_probe, .remove = __devexit_p(pcm3168_codec_remove), .driver = { .name = "pcm3168-codec", .owner = THIS_MODULE, }, }; static int __init pcm3168_modinit(void) { printk(KERN_ALERT "in init of 3168 -> kli \n"); return platform_driver_register(&pcm3168_codec_driver); } module_init(pcm3168_modinit); static void __exit pcm3168_exit(void) { printk(KERN_ALERT "in exit of 3168 -> kli \n"); platform_driver_unregister(&pcm3168_codec_driver); } module_exit(pcm3168_exit); MODULE_DESCRIPTION("Soc PCM3168 driver"); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_LICENSE("GPL"); * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.h * *********************************************************************************************** * /* * PCM3168 ALSA SoC Layer * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_PCM3168_2_H #define __LINUX_SND_SOC_PCM3168_2_H struct pcm3168_setup_data { // unsigned dem0_pin; // unsigned dem1_pin; // unsigned pdad_pin; // unsigned pdda_pin; }; #endif

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[alsa-devel] [PATCH v2 0/4] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

New day, new patches. This time there is a proper lock around access to the ELD buffer. I hope this satisfies the previous comments. I've also tested the patches together with some PulseAudio patches I'm writing and the notification works just fine. The ELD data shows up 300 ms after hotplug, so it's likely the first repoll giving me new ELD data). David Henningsson (4): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: protect access to ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 12 ++++++-- sound/pci/hda/hda_local.h | 3 ++ sound/pci/hda/patch_hdmi.c | 67 +++++++++++++++++++++++++++++++++++--------- 3 files changed, 66 insertions(+), 16 deletions(-) -- 1.7.9.5

2 6

[alsa-devel] [PATCH 0/3] Make HDMI ELD usable with hotplug
by David Henningsson 19 Feb '13

19 Feb '13

I'm experimenting with using ELD information in PulseAudio and found that it didn't work very well with hotplugging. Neither the proc file nor the ELD kcontrol was cleared correctly after unplug, and there was no way (except polling) to find out when ELD has become available. Could use advice on which ones of these that should or should not be sent to stable. David Henningsson (3): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/patch_hdmi.c | 35 ++++++++++++++++++++++++++++------- 1 file changed, 28 insertions(+), 7 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Jack: add configurable option for irq_flag
by Omair Mohammed Abdullah 18 Feb '13

18 Feb '13

From: Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> For a jack using GPIO based detection, the framework sets it to trigger on both edges (rising and falling). Some codecs may provide only a falling or rising edge interrupt but not both. This patch provides a configurable option for irq_flag during snd_soc_jack_add_gpios(). If not configured, it sets the interrupt line to trigger on both edges. Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-jack.c | 7 +++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..d0286ad 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -585,6 +585,7 @@ struct snd_soc_jack_zone { * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms + * @irq_flag: Interrupt flags for GPIO-Irq line * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -597,6 +598,7 @@ struct snd_soc_jack_gpio { int report; int invert; int debounce_time; + unsigned long irq_flags; bool wake; struct snd_soc_jack *jack; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..c9f7953 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -318,10 +318,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; + if (!gpios[i].irq_flags) + gpios[i].irq_flags = + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING; + ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING, + gpios[i].irq_flags, gpios[i].name, &gpios[i]); if (ret < 0) -- 1.7.4.1

4 9

[alsa-devel] HDMI regression on close (was: ALSA: hda - Always turn on pins for HDMI/DP)
by Anssi Hannula 18 Feb '13

18 Feb '13

Hi! > From: Takashi Iwai <tiwai(a)suse.de> > Date: Fri, 14 Dec 2012 09:22:35 +0000 (+0100) > Subject: ALSA: hda - Always turn on pins for HDMI/DP > X-Git-Tag: v3.8-rc1~29^2~7 > X-Git-Url: http://git.kernel.org/?p=linux%2Fkernel%2Fgit%2Fstable%2Flinux-stable.git;a… > > ALSA: hda - Always turn on pins for HDMI/DP > > We've seen the broken HDMI *video* output on some machines with GM965, > and the debugging session pointed that the culprit is the disabled > audio output pins. Toggling these pins dynamically on demand caused > flickering of HDMI TV. > > This patch changes the behavior to keep the pin ON constantly. Unfortunately this seems to cause a regression on at least some "NVIDIA ION2" systems. When running snd_pcm_open() and snd_pcm_close() on an hdmi device on an affected system, the HDMI port seemingly continues to transmit an empty audio stream. However, when hw_params have been set and snd_pcm_prepare() called before closing the device (and optionally audio has been output), the HDMI port stops transmitting at snd_pcm_close() time. (I use "transmitting" here to mean that the HDMI sink/receiver receives/shows a an active audio stream) On systems _not_ affected by the issue, the HDMI port never starts transmitting if just snd_pcm_open()+snd_pcm_close() have been called (but note [1]). HDMI port continuing transmitting has the unfortunate effect that when the user has connected the system to an amplifier via HDMI (for video playback purposes), the amplifier may continue to play back empty audio from HDMI even when there is actual audio data on its S/PDIF input. An snd_pcm_open()+snd_pcm_close() sequence can be caused by e.g. device probing by an application. Several OpenELEC users have reported this issue on stable 3.7.x, here's some alsainfo: http://onse.fi/files/ion-regression-alsainfo-1.txt http://onse.fi/files/ion-regression-alsainfo-2.txt Both of these have the Nvidia GPU 0b HDMI/DP codecs: Vendor Id: 0x10de000b Subsystem Id: 0x10de0101 Revision Id: 0x100200 However, e.g. my non-ION2 laptop with 0b codecs (but different rev) is _not_ affected: Vendor Id: 0x10de000b Subsystem Id: 0x10de0101 Revision Id: 0x100100 http://www.alsa-project.org/db/?f=bc940d23d19ae7e2ee1e83895d85f711c2d0e5b1 I'm still working with the OpenELEC devs and users to pinpoint the issue further, but it is progressing rather slowly due to not many people having affected systems (especially in the affected dual HDMI + S/PDIF cabled configuration). Feel free to suggest anything we should try out if you have some ideas :) In the meantime, I'll continue investigating exactly why snd_pcm_close() doesn't stop the HDMI output on the affected systems (or why does a simple snd_pcm_open() even start outputting the audio, since that does not seem to happen on other systems). I'll keep you posted. [1] It seems that on my Intel HDMI system ( http://www.alsa-project.org/db/?f=bb3fc8680b372c0475719d42d7fc8b2bb7bfb4eb ) the HDMI port keeps transmitting in approximately reversed situation than in the above ION2 case. That is, if audio has been output in the port e.g. with a speaker-test command, closing the device will not cause the stream to stop (and therefore the above issue happens again, receives ignores S/PDIF stream as HDMI stream is given higher priority by default). I found out that the attached patch solves that one. What are your thoughts on that? (obviously the patch will not solve the ION2 regression, since cleanup() is not called if prepare() has not been called; or at least I assume so, way too tired to check that now...) Note that closing the device after audio has been output stops HDMI port transmission even without the patch on both the NVIDIA ION2 systems and on my NVIDIA laptop, it is just the Intel HDMI codec where that does not happen (well, of the tested ones, at least). > Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51421 > > Cc: <stable(a)vger.kernel.org> > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index 0fcfa6f..37dd06d 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -431,9 +431,11 @@ static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid) > if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) > snd_hda_codec_write(codec, pin_nid, 0, > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); > - /* Disable pin out until stream is active*/ > + /* Enable pin out: some machines with GM965 gets broken output when > + * the pin is disabled or changed while using with HDMI > + */ > snd_hda_codec_write(codec, pin_nid, 0, > - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); > + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); > } > > static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid) > @@ -1341,7 +1343,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, > struct hdmi_spec *spec = codec->spec; > int pin_idx = hinfo_to_pin_index(spec, hinfo); > hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; > - int pinctl; > bool non_pcm; > > non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); > @@ -1350,11 +1351,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, > > hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); > > - pinctl = snd_hda_codec_read(codec, pin_nid, 0, > - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); > - snd_hda_codec_write(codec, pin_nid, 0, > - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); > - > return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); > } > > @@ -1374,7 +1370,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, > int cvt_idx, pin_idx; > struct hdmi_spec_per_cvt *per_cvt; > struct hdmi_spec_per_pin *per_pin; > - int pinctl; > > if (hinfo->nid) { > cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); > @@ -1391,11 +1386,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, > return -EINVAL; > per_pin = &spec->pins[pin_idx]; > > - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, > - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); > - snd_hda_codec_write(codec, per_pin->pin_nid, 0, > - AC_VERB_SET_PIN_WIDGET_CONTROL, > - pinctl & ~PIN_OUT); > snd_hda_spdif_ctls_unassign(codec, pin_idx); > per_pin->chmap_set = false; > memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); -- Anssi Hannula

2 8

Re: [alsa-devel] [PATCH V2] ARM: dts: omap3-devkit8000: Enable audio support
by Peter Ujfalusi 18 Feb '13

18 Feb '13

On 02/16/2013 08:33 AM, Anil Kumar wrote: > Add the needed sections to enable audio support on > Devkit8000 when booted with DT blob. > > Signed-off-by: Anil Kumar <anilk4.v(a)gmail.com> Looks good: Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > --- > This patch is based on top of kernel 3.8-rc5 and > the following patches. > > Peter Ujfalusi:- > ASoC: twl4030: Correct the support for Voice port > ASoC: twl4030: Convert MICBIAS to SUPPLY widget > ASoC: omap-twl4030: Add support for routing, voice port and jack detect > > Anil Kumar:- > ARM: dts: add minimal DT support for DevKit8000 > https://patchwork.kernel.org/patch/2122461/ > > -Tested for playback and capture on Devkit8000. > > Test process:- > > #amixer set 'PredriveR Mixer AudioR2' on > #amixer set 'PredriveL Mixer AudioL2' on > #amixer set PreDriv 100 unmute > #amixer set 'DAC2 Digital Fine' 100 > #amixer cset numid=27 1 > #arecord | aplay > > For V2: > - Remove pin mux as done at device boot time by default. > > :100644 100644 6338993... c70116f... M arch/arm/boot/dts/omap3-devkit8000.dts > arch/arm/boot/dts/omap3-devkit8000.dts | 23 +++++++++++++++++++---- > 1 files changed, 19 insertions(+), 4 deletions(-) > > diff --git a/arch/arm/boot/dts/omap3-devkit8000.dts b/arch/arm/boot/dts/omap3-devkit8000.dts > index 6338993..c70116f 100644 > --- a/arch/arm/boot/dts/omap3-devkit8000.dts > +++ b/arch/arm/boot/dts/omap3-devkit8000.dts > @@ -42,6 +42,19 @@ > }; > > }; > + > + sound { > + compatible = "ti,omap-twl4030"; > + ti,model = "devkit8000"; > + > + ti,mcbsp = <&mcbsp2>; > + ti,codec = <&twl_audio>; > + ti,audio-routing = > + "Ext Spk", "PREDRIVEL", > + "Ext Spk", "PREDRIVER", > + "MAINMIC", "Main Mic", > + "Main Mic", "Mic Bias 1"; > + }; > }; > > &i2c1 { > @@ -51,6 +64,12 @@ > reg = <0x48>; > interrupts = <7>; /* SYS_NIRQ cascaded to intc */ > interrupt-parent = <&intc>; > + > + twl_audio: audio { > + compatible = "ti,twl4030-audio"; > + codec { > + }; > + }; > }; > }; > > @@ -86,10 +105,6 @@ > status = "disabled"; > }; > > -&mcbsp2 { > - status = "disabled"; > -}; > - > &mcbsp3 { > status = "disabled"; > }; > -- Péter

1 0

[alsa-devel] [PATCH] ALSA: hda - add quirks for mute LED on two HP machines
by David Henningsson 18 Feb '13

18 Feb '13

These two machines have no mute LED string in BIOS. BugLink: https://bugs.launchpad.net/bugs/1128934 Tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Should be sent to stable too, but I guess the patch have to be rewritten to fit. sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 48c9d10..e575685 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2726,6 +2726,18 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, } } +static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_nid = 0x18; + spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; + spec->gen.vmaster_mute_enum = 1; + } +} + static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2771,6 +2783,7 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_HP_MUTE_LED, + ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -2903,6 +2916,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic1, + }, [ALC269_FIXUP_HP_MUTE_LED_MIC2] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, @@ -2947,6 +2964,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), + SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove speaker clicks on CX20549
by David Henningsson 18 Feb '13

18 Feb '13

This chip needs the speaker pin to go to D3 to avoid clicks, so default_power_filter does not work here. This was found on Thinkpad R61i/T61i but I guess it applies to the entire chip. If not, quirks should be set for at least PCI SSID 17aa:20ac. Thanks to c4pp4 for testing. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7d941ef..941bf6c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3350,6 +3350,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; + codec->power_filter = NULL; /* Needs speaker amp to D3 to avoid click */ break; case 0x14f15047: codec->pin_amp_workaround = 1; -- 1.7.9.5

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Re: [alsa-devel] 3.7 HDMI channel map regression
by Takashi Iwai 17 Feb '13

17 Feb '13

At Sat, 16 Feb 2013 18:22:25 -0600, Shawn Bohrer wrote: > > On Mon, Jan 28, 2013 at 08:52:05PM -0600, Shawn Bohrer wrote: > > On Mon, Jan 28, 2013 at 09:56:33AM +0100, Takashi Iwai wrote: > > > At Sun, 27 Jan 2013 19:18:27 -0600, > > > Shawn Bohrer wrote: > > > > > > > > Hi Takashi, > > > > > > > > I recently updated my HTPC from 3.6.11 to 3.7.2 and this caused my RL > > > > and FC channels to swap, and my RR and LFE channels to swap for PCM > > > > audio. Doing a git bisect identified > > > > d45e6889ee69456a4d5b1bbb32252f460cd48fa9 "ALSA: hda - Provide the > > > > proper channel mapping for generic HDMI driver" as the commit that > > > > caused my channels to swap. The commit doesn't revert cleanly on > > > > 3.7.4, and I haven't really looked to see what the correct fix might > > > > be. > > > > > > > > Some info that may be relevant, the sound card is a: > > > > > > > > 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset > > > > Family High Definition Audio Controller (rev 04) > > > > > > > > The machine is running Fedora 18 and audio goes over HDMI to a 5.1 > > > > receiver. I'm not really sure what other info you might need, but > > > > let me know if you need something else or have any patches you would > > > > like me to test. > > > > > > OK, it's the first time to get a bug report about this. > > > Could you tell me how did you test it (i.e. which application, which > > > sound backend)? Can you confirm that it's reproduced via speaker-test > > > program in alsa-utils package? > > > > I originally noticed the problem when all of the dialog started coming > > out of my rear left speaker in MythTV after the kernel update. Then I > > started using the Gnome 3 sound configuration gui in the system > > settings which has a speaker test and I assume is using pulseaudio. > > Running 'speaker-test -c6 -l1 -twav' also reproduces the problem. > > > > For reference here are the versions of the various packages that I'm > > running: > > > > alsa-utils-1.0.26-1.fc18.x86_64 > > alsa-firmware-1.0.25-2.fc18.noarch > > alsa-plugins-pulseaudio-1.0.26-2.fc18.x86_64 > > alsa-lib-devel-1.0.26-2.fc18.x86_64 > > alsa-lib-1.0.26-2.fc18.x86_64 > > alsa-tools-firmware-1.0.26.1-1.fc18.x86_64 > > pulseaudio-gdm-hooks-2.1-5.fc18.x86_64 > > pulseaudio-libs-2.1-5.fc18.x86_64 > > pulseaudio-libs-glib2-2.1-5.fc18.x86_64 > > pulseaudio-module-x11-2.1-5.fc18.x86_64 > > pulseaudio-module-bluetooth-2.1-5.fc18.x86_64 > > pulseaudio-2.1-5.fc18.x86_64 > > pulseaudio-utils-2.1-5.fc18.x86_64 > > Hi Takashi, > > Any updates on this issue? I'd really like to see this issue fixed > and am happy to help in any way I can. Until this gets fixed I'm > stuck on a 3.6.* kernel. There is one fix in sound git tree regarding the HDMI channel map, but it's queued for 3.9 kernel (then backported to stable tree). Try sound.git tree or wait for a while until the upstream merge process above is done. Takashi

1 0

[alsa-devel] ASoC updates for v3.9
by Mark Brown 17 Feb '13

17 Feb '13

The following changes since commit 699ba4546cfe3f4eee73aff4f307bf16362f9232: Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next (2013-02-11 11:06:55 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.9-updates for you to fetch changes up to 54759313ececcbd07e43fad75e0683e39a1f2a25: Merge remote-tracking branch 'asoc/topic/da7213' into asoc-next (2013-02-14 17:28:00 +0000) ---------------------------------------------------------------- ASoC: Final updates for v3.9 A few more updates from the past week - a new driver from Dialog and some small fixes and tweaks. ---------------------------------------------------------------- Adam Thomson (1): ASoC: codecs: Add da7213 codec Fabio Estevam (1): ASoC: fsl: imx-audmux: Fix sparse warning Mark Brown (6): ASoC: wm8994: Fix variable double use ASoC: arizona: Automatically manage input mutes Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/da7213' into asoc-next Sebastien Guiriec (1): ASoC: omap-mcpdm: Remove useless ressource get. include/sound/da7213.h | 52 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 17 + sound/soc/codecs/da7213.c | 1599 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7213.h | 523 ++++++++++++++ sound/soc/codecs/wm5102.c | 13 - sound/soc/codecs/wm5110.c | 17 - sound/soc/codecs/wm8994.c | 10 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/omap/omap-mcpdm.c | 4 - 11 files changed, 2203 insertions(+), 40 deletions(-) create mode 100644 include/sound/da7213.h create mode 100644 sound/soc/codecs/da7213.c create mode 100644 sound/soc/codecs/da7213.h

2 1

Re: [alsa-devel] [PATCH V6 02/10] ASoC: SAMSUNG: Add DT support for i2s
by Mark Brown 17 Feb '13

17 Feb '13

On Thu, Feb 14, 2013 at 09:33:00PM +0100, Sylwester Nawrocki wrote: > My apologies for the late review. It's already sixth version of this patch > series... But I noticed it just now when applying it for Exynos4. The GPIO > part of the bindings is now incompatible with Exynos4 and the bindings will > need to be changed. Leaving the compatibility code in the I2S driver, when > we could have it done well right from the beginning. If I'm not mistaken > a file similar to arch/arm/boot/dts/exynos4x12-pinctrl.dtsi with at least > entries for the I2S devices is needed and an addition of related pinctrl > nodes in exynos5250.dtsi. But I guess this is not something that could be > fixed in the -rc period ? Yes, and incremental fix would be fine.

2 1

[alsa-devel] Takashi: please help (compiling alsa-driver)
by Jonas Petersen 15 Feb '13

15 Feb '13

Hi Takashi, I'm talking to you directly because this ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.… is the only alsa-driver source that I'm able to compile so far. I can unpack this and simply run ./configure && make && make install. It works great. (I compile against the kernel-headers from ubuntu). I spent hours in trying to compile from several other sources. * The plain1.0.25 release (complains about missing symbols). * The snapshots fromhttp://www.alsa-project.org/snapshot/ (will complain about missing 'uapi' stuff) * Btw. where did the snapshots fromhttp://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/ go (Advertised here:http://www.alsa-project.org/main/index.php/Download)? * Checked out alsa-driver from git. But no luck, tried many things. e.g.: Branch release, then autoconf, but not even ./configure succeeds. Id like to submit some patches and be able to work with the current state from git. How is that supposed to work? Btw. I often read that the alsa-kernel is needed to compile. But it is not neede with this above mentioned 'unstable-snapshot'. It will just compile against the ubuntu kernel-headers (Lubuntu 12.10). How do you generate these snapshots? Thanks and sorry if that's stupid questions.. Jonas

3 6

[alsa-devel] Allocating DMA buffer for non-PCM
by Adrian Knoth 15 Feb '13

15 Feb '13

Hi! Would you say that the following is the proper way to allocate a DMA buffer used to hold level data? struct hdspm { struct pci_dev *pci; /* and an pci info */ struct snd_dma_buffer dmaLevelBuffer; u32 *level_buffer; /* suitably aligned address */ [..] } /* allocate level buffer */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(hdspm->pci), MADIFX_LEVEL_BUFFER_SIZE, &hdspm->dmaLevelBuffer); if (err < 0) { /* error */ [..] } hdspm->level_buffer = snd_sgbuf_get_ptr(&(hdspm->dmaLevelBuffer), 0); memset(hdspm->level_buffer, 0, MADIFX_LEVEL_BUFFER_SIZE); This used to work on my development machine (kernel 3.6.x), but now a kernel 3.2.0 user reports a NULL pointer dereference of hdspm->level_buffer, so apparently, snd_sgbuf_get_ptr() returned NULL for him. How could this possibly happen? Am I missing something? Better use SNDRV_DMA_TYPE_DEV instead? TIA

3 8

Re: [alsa-devel] [PATCH] ASoC: codecs: Add da7213 codec
by Mark Brown 14 Feb '13

14 Feb '13

On Thu, Feb 14, 2013 at 12:02:51PM +0000, Adam Thomson wrote: > This patch adds support for the Dialog DA7213 audio codec. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA - au88x0 - Define channel map for au88x0
by Raymond Yau 14 Feb '13

14 Feb '13

Define channel map for playback, capture devices of au88x0 Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com>

2 1

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 14 Feb '13

14 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> ---- v6: - update ioctls GET_METADATA to IOWR - add documenetation of enum v5: - update metadata value with 8 words - add reset of new states in set_params - use statically allocated structs v4: - update metadata struct with single key/value v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation ----- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 31 ++++++++- sound/core/compress_offload.c | 96 +++++++++++++++++++++++++ 4 files changed, 180 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..d630163 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,27 @@ struct snd_compr_codec_caps { }; /** + * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the + * end of the track + * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the + * beginning of the track + */ +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +166,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +177,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..cbb592f 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -483,6 +483,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->metadata_set = false; + stream->next_track = false; } else { return -EPERM; } @@ -514,6 +516,49 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->get_metadata(stream, &metadata); + if (retval != 0) + return retval; + + if (copy_to_user((void __user *)arg, &metadata, sizeof(metadata))) + return -EFAULT; + + return 0; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->set_metadata(stream, &metadata); + stream->metadata_set = true; + + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +639,44 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +706,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +733,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 2

Re: [alsa-devel] [PATCH V6 02/10] ASoC: SAMSUNG: Add DT support for i2s
by Sylwester Nawrocki 14 Feb '13

14 Feb '13

On 01/18/2013 12:47 PM, Padmavathi Venna wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- ... > /* Lock for cross i/f checks */ > @@ -997,19 +1006,76 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) > return i2s; > } > > +#ifdef CONFIG_OF > +static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) > +{ > + struct device *dev = &i2s->pdev->dev; > + int index, gpio, ret; > + > + for (index = 0; index < 7; index++) { > + gpio = of_get_gpio(dev->of_node, index); > + if (!gpio_is_valid(gpio)) { > + dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); > + goto free_gpio; > + } > + > + ret = gpio_request(gpio, dev_name(dev)); > + if (ret) { > + dev_err(dev, "gpio [%d] request failed\n", gpio); > + goto free_gpio; > + } > + i2s->gpios[index] = gpio; > + } > + return 0; > + > +free_gpio: > + while (--index >= 0) > + gpio_free(i2s->gpios[index]); > + return -EINVAL; > +} > + > +static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) > +{ > + unsigned int index; > + for (index = 0; index < 7; index++) > + gpio_free(i2s->gpios[index]); > +} > +#else > +static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) > +{ > + return -EINVAL; > +} > + > +static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) > +{ > +} > + > +#endif NAK. Why we should leave with this temporary code when there is already pinctrl support for Exynos SoCs ? If the pinctrl driver for Exynos5 was updated you could instead just do devm_pinctrl_get_select_default(&i2s->pdev->dev);

2 1

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 14 Feb '13

14 Feb '13

this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v5: - update metadata value with 8 words - add reset of new states in set_params - use statically allocated structs v4: - update metadata struct with single key/value v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 25 ++++++- sound/core/compress_offload.c | 96 +++++++++++++++++++++++++ 4 files changed, 174 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..5081a66 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,21 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +160,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +171,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..cbb592f 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -483,6 +483,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->metadata_set = false; + stream->next_track = false; } else { return -EPERM; } @@ -514,6 +516,49 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->get_metadata(stream, &metadata); + if (retval != 0) + return retval; + + if (copy_to_user((void __user *)arg, &metadata, sizeof(metadata))) + return -EFAULT; + + return 0; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->set_metadata(stream, &metadata); + stream->metadata_set = true; + + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +639,44 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +706,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +733,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 4

Re: [alsa-devel] Use of of_parse_phandle()/of_node_put()
by Peter Ujfalusi 14 Feb '13

14 Feb '13

On 02/13/2013 06:02 PM, Anil Kumar wrote: > ------------8--------------- > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index b7e84a7..9000f4a 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -1044,6 +1044,13 @@ static void soc_remove_dai_links(struct > snd_soc_card *card) > soc_remove_link_dais(card, dai, order); > } > > + /* release cpu_of_node */ > + if(card->dai_link) { > + int i; > + for(i = 0; i < card->num_links; i++) > + of_node_put(card->dai_link[i].cpu_of_node); and the same for codec_of_node and platform_of_node for that matter. Mark: what do you think? Does it make sense to do this in the core or should we let the machine drivers to take care of this? > + } > + > -- Péter

1 0

Re: [alsa-devel] Use of of_parse_phandle()/of_node_put()
by Peter Ujfalusi 14 Feb '13

14 Feb '13

On 02/13/2013 05:40 PM, Anil Kumar wrote: > It looks issue is different here. > I have done some testing here with patch[1]. In this patch i tried to > get same "of_device" node pointer > again and found refcount for this dt node is 1 [Result]. > > As of_parse_phandle() says it "returns the device_node pointer with > refcount incremented". > But why refcount value is 1 [Result] again ? Documentation/kref.txt > > -----------------------------8--------------------- > Patch[1]:- > diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c > index fd98509..0828a5c 100644 > --- a/sound/soc/omap/omap-twl4030.c > +++ b/sound/soc/omap/omap-twl4030.c > @@ -297,6 +297,16 @@ static int omap_twl4030_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "McBSP node is not provided\n"); > return -EINVAL; > } > + printk(KERN_ERR"refcount 0x%x", > atomic_read(&dai_node->kref.refcount)); > + > + dai_node = of_parse_phandle(node, "ti,mcbsp", 0); > + if (!dai_node) { > + dev_err(&pdev->dev, "McBSP node is not provided\n"); > + return -EINVAL; > + } > + printk(KERN_ERR"refcount 0x%x", > atomic_read(&dai_node->kref.refcount)); > + > + > omap_twl4030_dai_links[0].cpu_dai_name = NULL; > omap_twl4030_dai_links[0].cpu_of_node = dai_node; > > [Result]:- > root@DevKit8000:/# insmod snd-soc-omap-twl4030.ko > [ 95.718109] refcount 0x1 > [ 95.720611] refcount 0x1 [ 95.818054] omap-twl4030 sound.20: > twl4030-hifi <-> 49022000.mcbsp mapping ok > > > Thanks, > Anil > [...] > -- Péter

1 0

[alsa-devel] Channel map of alsa-plugin (pulse, jack)
by Raymond Yau 14 Feb '13

14 Feb '13

Does channel map need to be implemented in alsa-plugin (e.g. alsa-pulse and alsa-jack) ?

1 0

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 13 Feb '13

13 Feb '13

this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> ---- v4: - update metadata struct witj single key/value v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation -- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 25 +++++- sound/core/compress_offload.c | 113 +++++++++++++++++++++++++ 4 files changed, 191 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..d7544b5 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,21 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +160,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +171,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..8d6812e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,66 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->get_metadata(stream, metadata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, metadata, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + +out: + kfree(metadata); + return retval; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->set_metadata(stream, metadata); + stream->metadata_set = true; + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +654,46 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +723,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +750,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 4

Re: [alsa-devel] Use of of_parse_phandle()/of_node_put()
by Peter Ujfalusi 13 Feb '13

13 Feb '13

On 02/13/2013 07:04 AM, Kumar, Anil wrote: > Hi Peter, > > Just trying to understand. > > In omap-twl4030.c file probe function :- > > dai_node = of_parse_phandle(node, "ti,mcbsp", 0); > if (!dai_node) { > dev_err(&pdev->dev, "McBSP node is not provided\n"); > return -EINVAL; > } > > Here “of_parse_phandle()” is used to get “of_device” node pointer. > of_parse_phandle() suggest to use of_node_put() on it when done. > > It looks when code request for an “of_device” node, kernel maintains > “refcount” for this. > It check “refcount” before giving pointer of of_device node and WARN_ON() > in case of refcount > 0 and increase it on success. > > Should this code need to use of_node_put() on the requested “of_device” when done > so that this can be get again ? Hrm, one thing or sure we should not call it of_node_put() while we have the card loaded since the node is used runtime by the core. However when we unload the machine driver it might be needed, but not sure about this. None of the existing machine drivers doing it (tegra, samsung, omap, etc). But if it is needed it might be better to be done by the core? > > I am taking reference of Linux-next kernel. > > I'm sorry if this question is very vague or missing something. > > Thanks, > Anil > > > > > > -- Péter

1 0

[alsa-devel] HD-audio generic parser merged to linux-next
by Takashi Iwai 13 Feb '13

13 Feb '13

Hi, as most of feedbacks about the new generic parser migration were positive, I decided to take a risk: now all commit that have been in test/hda-migrate branch in sound-unstable git tree are merged to for-next branch of sound git tree for better test coverage. It means that all changes are merged to linux-next, i.e. will be included in 3.9 kernel. For the early adapter people, I'm going to keep test/hda-migrate branch of sound-unstable git tree updated for a while, but the branch will be abandoned some time later. If you encounter any regression by the recent update, please let me know. thanks, Takashi

3 14

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 13 Feb '13

13 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 23 +++++- sound/core/compress_offload.c | 116 +++++++++++++++++++++++++ 4 files changed, 192 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..8903fe4 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,19 @@ struct snd_compr_codec_caps { }; /** + * struct snd_compr_metadata: compressed stream metadata + * @encoder_delay: no of samples inserted by the encoder at the beginning + * of the track + * @encoder_padding: no of samples appended by the encoder at the end + * of the track + */ +struct snd_compr_metadata { + __u32 encoder_delay; + __u32 encoder_padding; + __u32 reserved[12]; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +158,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +169,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..ba2f0ad 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,69 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->get_metadata(stream, metadata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, metadata, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + +out: + kfree(metadata); + return retval; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + pr_debug("metadata encoder delay=%x encoder padding=%x\n", + metadata->encoder_delay, metadata->encoder_padding); + + retval = stream->ops->set_metadata(stream, metadata); + stream->metadata_set = true; + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +657,46 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +726,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +753,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] ASoC: omap-dmic: Clean up with devm_request_and_ioremap
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up dmic code with devm_request_and_ioremap function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-dmic.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index ba49ccd..77e9e7e 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -493,16 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) goto err_put_clk; } - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(dmic->dev, "memory region already claimed\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); + dmic->io_base = devm_request_and_ioremap(&pdev->dev, res); if (!dmic->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); ret = -ENOMEM; goto err_put_clk; } -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Clean up with devm_* function
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up McPDM driver with devm_ function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 5ca11bd..079f277 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -369,7 +369,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -389,7 +389,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; @@ -465,14 +464,11 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), "McPDM")) - return -EBUSY; - - mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!mcpdm->io_base) + mcpdm->io_base = devm_request_and_ioremap(&pdev->dev, res); + if (!mcpdm->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); return -ENOMEM; + } mcpdm->irq = platform_get_irq(pdev, 0); if (mcpdm->irq < 0) -- 1.7.10.4

3 2

Re: [alsa-devel] [RFC]Connection list of hda widget
by Takashi Iwai 13 Feb '13

13 Feb '13

At Thu, 07 Feb 2013 16:09:35 +0100, David Henningsson wrote: > > On 02/07/2013 06:38 AM, Raymond Yau wrote: > > > > Treat vendor defined widget different from audio selector/audio input > > widget since hda_proc.c query the connection list of vendor defined widget > > > > ADI codecs have vendor defined widgets ( BIAS Power-Down and VREF > > Power-Down) which have connection list > > > > static void print_conn_list(struct snd_info_buffer *buffer, > > struct hda_codec *codec, hda_nid_t nid, > > unsigned int wid_type, hda_nid_t *conn, > > int conn_len) > > { > > int c, curr = -1; > > > > if (conn_len > 1 && > > wid_type != AC_WID_AUD_MIX && > > wid_type != AC_WID_VOL_KNB && > > + wide_type != AC_WID_VENDOR && > > wid_type != AC_WID_POWER) > > curr = snd_hda_codec_read(codec, nid, 0, > > AC_VERB_GET_CONNECT_SEL, 0); > > snd_iprintf(buffer, " Connection: %d\n", conn_len); > > > > I think there are some Sigmatel/IDT (or was it Analog Devices?) codecs > that actually have sensible connections for their vendor specific nodes, > so I would prefer not to change this. Right. The widget connection longer than 32 itself is OK. It's just ugly. That being said, it's basically a bug that we restrict the max connection size in the core code. So I applied the patch below. This removes the limit while we still keep the efficiency (except for proc output, which isn't too critical). The patch became bigger than I wanted originally, but it's life. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Remove limit of widget connections Currently we set the max number of connections to be 32, but there seems codec that gives longer connection lists like AD1988, and we see errors in proc output and else. (Though, in the case of AD1988, it's a list of all codecs connected to a single vendor widget, so this must be something fishy, but it's still valid from the h/w design POV.) This patch tries to remove this restriction. For efficiency, we still use the fixed size array in the parser, but takes a dynamic array when the size is reported to be greater than that. Now the fixed array size is found only in patch_hdmi.c, but it should be fine, as the codec itself can't support so many pins. Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 62 ++++++++++++++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 4 +-- sound/pci/hda/hda_proc.c | 19 +++++++++++--- sound/pci/hda/patch_hdmi.c | 3 +++ 4 files changed, 60 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f82a64d..3c92514 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -382,13 +382,23 @@ static void remove_conn_list(struct hda_codec *codec) /* read the connection and add to the cache */ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - hda_nid_t list[HDA_MAX_CONNECTIONS]; + hda_nid_t list[32]; + hda_nid_t *result = list; int len; len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); - if (len < 0) - return len; - return snd_hda_override_conn_list(codec, nid, len, list); + if (len == -ENOSPC) { + len = snd_hda_get_num_raw_conns(codec, nid); + result = kmalloc(sizeof(hda_nid_t) * len, GFP_KERNEL); + if (!result) + return -ENOMEM; + len = snd_hda_get_raw_connections(codec, nid, result, len); + } + if (len >= 0) + len = snd_hda_override_conn_list(codec, nid, len, result); + if (result != list) + kfree(result); + return len; } /** @@ -466,6 +476,27 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_get_connections); +/* return CONNLIST_LEN parameter of the given widget */ +static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); + if (parm == -1) + parm = 0; + return parm; +} + +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) +{ + return get_num_conns(codec, nid) & AC_CLIST_LENGTH; +} + /** * snd_hda_get_raw_connections - copy connection list without cache * @codec: the HDA codec @@ -483,19 +514,16 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; int i, conn_len, conns; unsigned int shift, num_elems, mask; - unsigned int wcaps; hda_nid_t prev_nid; int null_count = 0; if (snd_BUG_ON(!conn_list || max_conns <= 0)) return -EINVAL; - wcaps = get_wcaps(codec, nid); - if (!(wcaps & AC_WCAP_CONN_LIST) && - get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + parm = get_num_conns(codec, nid); + if (!parm) return 0; - parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { /* long form */ shift = 16; @@ -552,21 +580,13 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = n; } } else { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = val; } prev_nid = val; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c59272..0be1826 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -551,9 +551,6 @@ enum { AC_JACK_PORT_BOTH, }; -/* max. connections to a widget */ -#define HDA_MAX_CONNECTIONS 32 - /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f @@ -958,6 +955,7 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_get_connections(codec, nid, NULL, 0); } +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid); int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5e02f26..97cdbef 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -623,7 +623,7 @@ static void print_codec_info(struct snd_info_entry *entry, snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = get_wcaps_type(wid_caps); - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t *conn = NULL; int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, @@ -660,9 +660,18 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_type == AC_WID_VOL_KNB) wid_caps |= AC_WCAP_CONN_LIST; - if (wid_caps & AC_WCAP_CONN_LIST) - conn_len = snd_hda_get_raw_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); + if (wid_caps & AC_WCAP_CONN_LIST) { + conn_len = snd_hda_get_num_raw_conns(codec, nid); + if (conn_len > 0) { + conn = kmalloc(sizeof(hda_nid_t) * conn_len, + GFP_KERNEL); + if (!conn) + return; + if (snd_hda_get_raw_connections(codec, nid, conn, + conn_len) < 0) + conn_len = 0; + } + } if (wid_caps & AC_WCAP_IN_AMP) { snd_iprintf(buffer, " Amp-In caps: "); @@ -735,6 +744,8 @@ static void print_codec_info(struct snd_info_entry *entry, if (codec->proc_widget_hook) codec->proc_widget_hook(buffer, codec, nid); + + kfree(conn); } snd_hda_power_down(codec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85236da..899c4fb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,6 +64,9 @@ struct hdmi_spec_per_cvt { unsigned int maxbps; }; +/* max. connections to a widget */ +#define HDA_MAX_CONNECTIONS 32 + struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; -- 1.8.1.2

2 6

[alsa-devel] Channel map for Logitech Web Camera
by Raymond Yau 13 Feb '13

13 Feb '13

It seem that channel map is still missing for Logitech Web Camera since I can find the corresponding entries of hda-intel and the patched au8830 in asound.state except the Logitech Web Camera state.U0x46d0x992 { control.1 { iface MIXER name 'Mic Capture Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Mic Capture Volume' value 3041 comment { access 'read write' type INTEGER count 1 range '0 - 3072' dbmin 1800 dbmax 3000 dbvalue.0 2987 } } } Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 2 Streaming bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 38 bInCollection 1 baInterfaceNr( 0) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 16000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0044 1x 68 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined

1 0

[alsa-devel] [RFC] Fix for conflict of HDMI and SPDIF IEC958 controls
by Takashi Iwai 12 Feb '13

12 Feb '13

Hi, there is a long-standing problem in HD-audio regarding IEC958 controls. When both an SPDIF and an HDMI are created on the same card (e.g. one from analog codec and one from graphics chip), the driver assigns the IEC958 controls just with new indices, 0, 1, 2... The problem is that there is no way to connect between this index and the actual PCM device. Currently, alsa-lib HDA-Intel.conf has a fixed configuration: spdif -> IEC958 xxx index=0, PCM dev=1 hdmi,0 -> IEC958 xxx index=0, PCM dev=3 hdmi,1 -> IEC958 xxx index=1, PCM dev=7 hdmi,2 -> IEC958 xxx index=2, PCM dev=8 hdmi,3 -> IEC958 xxx index=3, PCM dev=9 So obviously spdif and the first hdmi conflict. Basically this can be fixed by reassigning each IEC958 control with the same device number corresponding to the PCM device. That is, for SPDIF, assign a control element with device=1, for HDMI, device=3,7... However, this obviously breaks the old configuration unless user upgrades the alsa-lib configuration. Thus this is no-go. Now here is a compromise: the IEC958 control for SPDIF is reassigned to device=1 *only* when SPDIF and HDMI coexist. Since the configuration is anyway broken as is now in such a case, it's no big deal to fix one side in an incompatible way. (The reason why SPDIF is re-assigned is that I guess majority of user require more HDMI than SPDIF in such a configuration.) In addition, we need a fix in alsa-lib. Also not for breaking the compatibility with older kernel, we need some fallback check. I did a quick hack to alsa-lib conf code and added "skip_rest" option to the hook element, so that only the first matching element is taken. Long story short, I cooked up three patches. One patch is for kernel, to add the workaround above, and the two are for alsa-lib, one clean up and one to introduce the skip_rest option (and application to HDA-Intel.conf). My plan is to merge this to 3.8 tree, so it's no urgent issue. But it's of course always good to fix something. thanks, Takashi

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Re: [alsa-devel] Xonar DG pci sound card driver incomplete
by Clemens Ladisch 12 Feb '13

12 Feb '13

kernel kernel wrote: > Requesting guidance on how to implement the missing mic input support > for this Asus Xonar card. I've downloaded the relevant datasheets but > am unsure how to proceed. Rip out the card, and look (or eletrically trace) how the CMI8788's GPIOs and the CS4245's inputs/outputs are connected. Test how to configure and set the GPIOs so that the mic works. Also check the register configuration of the CS4245. Regards, Clemens

1 0

Re: [alsa-devel] [PATCH 2/3] ALSA: hda - no-primary-hp is a quirk for model ALC889 not ALC882
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:54:16 +0900, Fernando Luis Vázquez Cao wrote: > > Substitute ALC889 for ALC882 in macro and function names. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> This fixup can work generically to all ALC882 variants, including ALC889. Unless you are really sure that this must be used exclusively only for ALC889, no reason to change the prefix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:52:40.301209823 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:58:54.695063675 +0900 > @@ -5030,7 +5030,7 @@ enum { > ALC889_FIXUP_MBP_VREF, > ALC889_FIXUP_IMAC91_VREF, > ALC882_FIXUP_INV_DMIC, > - ALC882_FIXUP_NO_PRIMARY_HP, > + ALC889_FIXUP_NO_PRIMARY_HP, > }; > > static void alc889_fixup_coef(struct hda_codec *codec, > @@ -5156,7 +5156,7 @@ static void alc889_fixup_imac91_vref(str > * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > -static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > +static void alc889_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > { > struct alc_spec *spec = codec->spec; > @@ -5350,9 +5350,9 @@ static const struct alc_fixup alc882_fix > .type = ALC_FIXUP_FUNC, > .v.func = alc_fixup_inv_dmic_0x12, > }, > - [ALC882_FIXUP_NO_PRIMARY_HP] = { > + [ALC889_FIXUP_NO_PRIMARY_HP] = { > .type = ALC_FIXUP_FUNC, > - .v.func = alc882_fixup_no_primary_hp, > + .v.func = alc889_fixup_no_primary_hp, > }, > }; > > @@ -5388,8 +5388,8 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > - SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC889_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC889_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > @@ -5432,7 +5432,7 @@ static const struct alc_model_fixup alc8 > {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, > {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, > {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, > - {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > + {.id = ALC889_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > {} > }; > > >

1 0

Re: [alsa-devel] [PATCH 2/3] ALSA: hda - update documentation for no-primary-hp fixup
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:49:46 +0900, Fernando Luis Vázquez Cao wrote: > > The problem addressed by this fixup is not specific to Vaio Z, affecting > some Vaio all-in-one desktop PCs too. Update the code comments accordingly. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> I applied this now to sound git tree but dropped Cc to stable because the patch is no longer cleanly applicable due to the recent code change. And this is just some minor documentation update, thus we can still live peacefully without this fix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt > --- linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt 2012-12-11 12:30:57.000000000 +0900 > +++ linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt 2013-02-09 22:44:28.614772054 +0900 > @@ -53,7 +53,7 @@ ALC882/883/885/888/889 > acer-aspire-8930g Acer Aspire 8330G/6935G > acer-aspire Acer Aspire others > inv-dmic Inverted internal mic workaround > - no-primary-hp VAIO Z workaround (for fixed speaker DAC) > + no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) > > ALC861/660 > ========== > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:39:11.745202168 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:50:10.924470402 +0900 > @@ -5153,7 +5153,8 @@ static void alc889_fixup_imac91_vref(str > } > > /* Don't take HP output as primary > - * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05 > + * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > + * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > >

1 0

Re: [alsa-devel] [PATCH 1/3] ALSA: hda - Workaround for silent output on Sony Vaio VGC-LN51JGB with ALC889
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:47:44 +0900, Fernando Luis Vázquez Cao wrote: > > Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by > the same issue that caused Vaio Z laptops to become silent: the speaker pin > must be connected to the first DAC even though the codec itself advertises > flexible routing through any of the DACs. > > Use the no-primary-hp fixup for choosing the speaker pin as the primary so > that the right DAC is assigned on this device. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> Thanks, applied now. Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-08 23:27:05.949484427 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-08 23:41:04.897644545 +0900 > @@ -5388,6 +5388,7 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > >

1 0

[alsa-devel] [patch] ALSA: hda/ca0132 - fix error handling for dsp_chip_to_dsp_addx()
by Dan Carpenter 12 Feb '13

12 Feb '13

The error handling here won't work on 64 bit systems. INVALID_CHIP_ADDRESS is an unsigned long defined like this: #define INVALID_CHIP_ADDRESS (~0UL) But we truncate the high bits away before returning and then we save it in an unsigned 32bit and then we compare it against the unsigned long version. I've changed everything to unsigned long. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 639a282..42114e9 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1581,7 +1581,7 @@ static int dsp_reset(struct hda_codec *codec) /* * Convert chip address to DSP address */ -static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx, +static unsigned long dsp_chip_to_dsp_addx(unsigned int chip_addx, bool *code, bool *yram) { *code = *yram = false; @@ -1596,7 +1596,7 @@ static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx, return Y_OFF(chip_addx); } - return (unsigned int)INVALID_CHIP_ADDRESS; + return INVALID_CHIP_ADDRESS; } /* @@ -1620,7 +1620,7 @@ static int dsp_dma_setup_common(struct hda_codec *codec, { int status = 0; unsigned int chnl_prop; - unsigned int dsp_addx; + unsigned long dsp_addx; unsigned int active; bool code, yram; @@ -1712,7 +1712,7 @@ static int dsp_dma_setup_common(struct hda_codec *codec, snd_printdd(KERN_INFO " dsp_dma_setup_common() Write IRQCNT"); snd_printdd( - "ChipA=0x%x,DspA=0x%x,dmaCh=%u, " + "ChipA=0x%x,DspA=0x%lx,dmaCh=%u, " "CHSEL=0x%x,CHPROP=0x%x,Active=0x%x\n", chip_addx, dsp_addx, dma_chan, port_map_mask, chnl_prop, active); @@ -1732,7 +1732,7 @@ static int dsp_dma_setup(struct hda_codec *codec, { int status = 0; bool code, yram; - unsigned int dsp_addx; + unsigned long dsp_addx; unsigned int addr_field; unsigned int incr_field; unsigned int base_cnt;

2 1

[alsa-devel] [PATCH] alsa-lib: Add sys/types.h to include list
by David Henningsson 12 Feb '13

12 Feb '13

This is needed by snd_pcm_format_silence* functions which return u_int*_t. It was discovered while trying to compile ALSA programs with eglibc 2.17. Credits to Richard Shaw, Gary Buhrmaster, Matthieu Baerts and Adam Conrad for this fix. BugLink: https://bugs.launchpad.net/bugs/1109298 BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=885306 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/asoundlib-head.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/asoundlib-head.h b/include/asoundlib-head.h index 20c8a68..6edbab0 100644 --- a/include/asoundlib-head.h +++ b/include/asoundlib-head.h @@ -31,6 +31,7 @@ #include <unistd.h> #include <stdio.h> #include <stdlib.h> +#include <sys/types.h> #include <string.h> #include <fcntl.h> #include <assert.h> -- 1.7.9.5

3 4

Re: [alsa-devel] Sound record on Devkit8000 with DT boot issue
by Peter Ujfalusi 11 Feb '13

11 Feb '13

Hi, On 02/11/2013 06:58 PM, Anil Kumar wrote: >> Here you should have something like: >> ti,audio-routing = >> "Ext Spk", "PREDRIVEL", >> "Ext Spk", "PREDRIVER", >> "MAINMIC", "Main Mic", >> "Main Mic", "Mic Bias 1"; >> >> So you connect the MicBias1 to Main Mic. When you start playback MicBias1 will >> be turned on for you. > > I tried with this routing (next-20130211 tag kernel). But it is not > working for me. > So i have gone through other board and found snd_soc_dapm_route for > > 1) zoom2.c > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Ext Mic"}, > > 2) omap3pandora.c > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Mic (internal)"}, > > But for omap-twl4030.c it is > {"MAINMIC", NULL, "Main Mic"}, > {"Main Mic", NULL, "Mic Bias 1"}, > > It Should will be as ? > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Main Mic"}, > > Because with this i am able to get audio on Devkit8000 as > ti,audio-routing = > "Ext Spk", "PREDRIVEL", > "Ext Spk", "PREDRIVER", > "MAINMIC", "Mic Bias 1", > "Mic Bias 1", "Main Mic"; In 3.9 this should not work... linux-next should not have zoom2 at least and I converted the machine drivers as well in regards to micbias handling: e04d6e5 ASoC: twl4030: Convert MICBIAS to SUPPLY widget 57d61b9 ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 -- Péter

1 0

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Remove useless ressource get.
by Sebastien Guiriec 11 Feb '13

11 Feb '13

Remove unused memory ressource get from McPDM driver. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 2fe8be2..5ca11bd 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -449,10 +449,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA; omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) - return -ENOMEM; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); if (!res) return -ENODEV; -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Fix sparse warning
by Fabio Estevam 11 Feb '13

11 Feb '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Fix the following sparse warning: sound/soc/fsl/imx-audmux.c:182:3: warning: symbol 'audmux_type' was not declared. Should it be static? Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..fab912e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -176,7 +176,7 @@ static inline void audmux_debugfs_remove(void) } #endif -enum imx_audmux_type { +static enum imx_audmux_type { IMX21_AUDMUX, IMX31_AUDMUX, } audmux_type; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 11 Feb '13

11 Feb '13

this adds new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 +++++++++ include/sound/compress_driver.h | 4 + include/uapi/sound/asound.h | 4 +- include/uapi/sound/compress_offload.h | 36 +++++++- sound/core/compress_offload.c | 129 ++++++++++++++++++++++++- 5 files changed, 215 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..95e955f 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -110,6 +110,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..88a6770 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -271,7 +271,9 @@ typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_PAUSED ((__force snd_pcm_state_t) 6) /* stream is paused */ #define SNDRV_PCM_STATE_SUSPENDED ((__force snd_pcm_state_t) 7) /* hardware is suspended */ #define SNDRV_PCM_STATE_DISCONNECTED ((__force snd_pcm_state_t) 8) /* hardware is disconnected */ -#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_DISCONNECTED +#define SNDRV_PCM_STATE_NEXT_TRACK ((__force snd_pcm_state_t) 9) /* stream will move to next track */ +#define SNDRV_PCM_STATE_PARTIAL_DRAIN ((__force snd_pcm_state_t) 10) /* stream is draining partially */ +#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_PARTIAL_DRAIN enum { SNDRV_PCM_MMAP_OFFSET_DATA = 0x00000000, diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..1d63993 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,32 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_keyvalue: compressed stream key/value pairs + * @key: key id + * @value: key value + */ + +struct snd_compr_keyvalue { + __u32 key; + __u32 value; +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @count: number of keys + * @keys: pointer to count keys + */ +struct snd_compr_metadata { + __u32 count; + struct snd_compr_keyvalue *keys; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +171,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +182,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..4928209 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -262,9 +262,14 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* write is allowed when stream is running or has been steup */ + /* write is allowed when stream is running or has been steup + * also stream cna be written when next_track info has been setup + * or its has partially drained + */ if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && - stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + stream->runtime->state != SNDRV_PCM_STATE_RUNNING && + stream->runtime->state != SNDRV_PCM_STATE_NEXT_TRACK && + stream->runtime->state != SNDRV_PCM_STATE_PARTIAL_DRAIN) { mutex_unlock(&stream->device->lock); return -EBADFD; } @@ -288,6 +293,9 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream->runtime->state = SNDRV_PCM_STATE_PREPARED; pr_debug("stream prepared, Houston we are good to go\n"); } + if (stream->runtime->state == SNDRV_PCM_STATE_NEXT_TRACK || + stream->runtime->state == SNDRV_PCM_STATE_PARTIAL_DRAIN) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; mutex_unlock(&stream->device->lock); return retval; @@ -514,6 +522,78 @@ out: return retval; } +static int +snd_compr_copy_metadata(struct snd_compr_metadata **arg, unsigned long user) +{ + struct snd_compr_metadata _mdata, *mdata; + int len; + + if (copy_from_user(&_mdata, (void __user *)user, + sizeof(_mdata))) + return -EFAULT; + + len = sizeof(_mdata.count) + _mdata.count * sizeof(*_mdata.keys); + + mdata = kmalloc(len, GFP_KERNEL); + if (!mdata) + return -ENOMEM; + + if (copy_from_user(mdata, (void __user *)arg, len)) { + kfree(mdata); + return -EFAULT; + } + *arg = mdata; + return len; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *mdata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + retval = snd_compr_copy_metadata(&mdata, arg); + if (retval <= 0) + return retval; + + retval = stream->ops->set_metadata(stream, mdata); + + kfree(mdata); + return retval; +} + +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *mdata; + int retval, len; + + if (!stream->ops->get_metadata) + return -ENXIO; + + len = snd_compr_copy_metadata(&mdata, arg); + if (len <= 0) + return len; + retval = stream->ops->get_metadata(stream, mdata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, mdata, len)) { + retval = -EFAULT; + goto out; + } + +out: + kfree(mdata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +674,38 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + + /* we can partially drain streams, only when next track info has been + * passed to the dsp + */ + if (stream->runtime->state != SNDRV_PCM_STATE_NEXT_TRACK) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval != 0) + return retval; + stream->runtime->state = SNDRV_PCM_STATE_PARTIAL_DRAIN; + return 0; +} + +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->runtime->state = SNDRV_PCM_STATE_NEXT_TRACK; + return 0; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +735,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +762,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 5

[alsa-devel] [PATCH] ALSA: core: don't return uninitialized snd_compr_tstamp
by Mark Brown 11 Feb '13

11 Feb '13

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp if the codec implements the pointer() function. If that happened the code was previously returning uninitialized garbage in the tstamp because it wasn't initialized anywhere. This change zero-fills the tstamp in the two places it is used before calling snd_compr_update_tstamp(), and also has snd_compr_update_tstamp() return an error indication if it can't provide a tstamp. For the case of snd_compr_calc_avail() it ignores this error because we still need to return info on the available buffer space even if we can't provide tstamp info - when the tstamp is not valid all fields are now guaranteed to be zero. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2d62068 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -144,16 +144,17 @@ static int snd_compr_free(struct inode *inode, struct file *f) return 0; } -static void snd_compr_update_tstamp(struct snd_compr_stream *stream, +static int snd_compr_update_tstamp(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { if (!stream->ops->pointer) - return; + return -ENOTSUPP; stream->ops->pointer(stream, tstamp); pr_debug("dsp consumed till %d total %d bytes\n", tstamp->byte_offset, tstamp->copied_total); stream->runtime->hw_pointer = tstamp->byte_offset; stream->runtime->total_bytes_transferred = tstamp->copied_total; + return 0; } static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, @@ -161,7 +162,9 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, { long avail_calc; /*this needs to be signed variable */ + memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); + /* Still need to return avail even if tstamp can't be filled in */ /* FIXME: This needs to be different for capture stream, available is # of compressed data, for playback it's @@ -517,11 +520,14 @@ out: static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_compr_tstamp tstamp; + struct snd_compr_tstamp tstamp = {0}; + int ret; - snd_compr_update_tstamp(stream, &tstamp); - return copy_to_user((struct snd_compr_tstamp __user *)arg, - &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + ret = snd_compr_update_tstamp(stream, &tstamp); + if (ret == 0) + ret = copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + return ret; } static int snd_compr_pause(struct snd_compr_stream *stream) -- 1.7.10.4

2 1

[alsa-devel] ASoC updates for v3.9
by Mark Brown 11 Feb '13

11 Feb '13

The following changes since commit 836dc9e3fbbab0c30aa6e664417225f5c1fb1c39: Linux 3.8-rc7 (2013-02-09 08:20:39 +1100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.9 for you to fetch changes up to 699ba4546cfe3f4eee73aff4f307bf16362f9232: Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next (2013-02-11 11:06:55 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.9 A fairly quiet release for ASoC: - Support for a wider range of hardware in the compressed stream code. - The ability to mute capture streams as well as playback streams while inactive. - DT support for AK4642, FSI, Samsung I2S and WM8962. - AC'97 support for Tegra. - New driver for max98090, replacing the stub which was there. Due to dependencies we've also got support for asynchronous I/O in regmap and DTification of DMA support for Samsung platforms (used only by the I2S driver and SPI) merged here as well. ---------------------------------------------------------------- Alexey Galakhov (1): ASoC: samsung: Fix compilation error on S3C2440 Andrey Smirnov (2): regmap: Add provisions to have user-defined read operation regmap: Add provisions to have user-defined write operation Axel Lin (2): ASoC: cs42l52: Convert to devm_input_allocate_device() regmap: Export regmap_async_complete_cb Benoît Thébaudeau (1): ASoC: tlv320aic3x: Remove mono support Bo Shen (4): ASoC: atmel_pcm: make it buildable as module ASoC: atmel_ssc_dai: remove error set private data ASoC: atmel_ssc_dai: correct sequence when unload ASoC: sam9g20_wm8731: disable clock and correct sequence when unload Charles Keepax (5): ASoC: wm_adsp: Release firmware on error ASoC: soc-compress: Add missing brackets around else ASoC: soc-compress: Serialise compressed ops ASoC: soc-compress: Initialise delayed work to power down audio ASoC: soc-compress: Add support for not memory mapped DSPs Chris Rattray (6): ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock ASoC: wm2200: Set system clock control register is adsp structs ASoC: wm2200: Add Rx ANC input select control ASoC: wm2200: correct IN2L and IN3L digital mute ASoC: wm2200: Provide platform data for MICBIAS configuration ASoC: wm_adsp: round to 4-byte boundary for coeff file blocks Chuansheng Liu (1): ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off Dan Carpenter (2): ASoC: pcm: delete some dead code ASoC: dwc: fix support for more than two channels Daniel Mack (1): ALSA: ASoC: cs4271: add optional soft reset workaround Fabio Estevam (6): ASoC: wm8962: Add device tree support ASoC: wm8804: Remove redundant check ASoC: wm8993: Refactor set_pll code to avoid GCC warnings ASoC: wm8985: Refactor set_pll code to avoid gcc warnings ASoC: mxs-saif: Remove platform data ASoC: mxs-saif: Use a signed integer for error value Hebbar Gururaja (1): ASoC: tlv320aic3x: Convert mic bias to a supply widget Jerry Wong (2): ASoC: Replace max98090 Device Driver ASoC: Replace max98090 Device Driver Joachim Eastwood (2): ASoC: atmel-ssc: make it buildable on other architectures ASoC: atmel-soc: make it buildable on other architectures Kuninori Morimoto (12): ASoC: fsi: remove platform depended .set_rate() callback support ASoC: fsi: remove SH_FSI_xxx_INV flags ASoC: fsi: cleanup sh_fsi.h ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: fsi: don't use platform info pointer on probe() ASoC: ak4642: add Device Tree support ASoC: simple-card: add asoc_simple_dai for initializing ASoC: fsi: add device tree support ASoC: SND_SOC_DAIFMT_NB_NF become 0 as default settings ASoC: add snd_soc_of_parse_daifmt() for DeviceTree ASoC: SND_SOC_DAIFMT_GATED become 0 as default settings ASoC: clock gating is decided by bool on snd_soc_of_parse_daifmt() Lars-Peter Clausen (1): ASoC: Constify ops and compr_ops fields of snd_soc_dai_link Lee Jones (1): ASoC: Ux500: Fix build error pertaining to missing include file Lucas Stach (4): ASoC: tegra: add function to set ac97 rate ASoC: tegra: setup DAP3<->DAC3 connection by default ASoC: tegra: add ac97 host driver ASoC: tegra: add tegra machine driver using wm9712 codec Mark Brown (97): ASoC: wm8962: Convert to devm_input_allocate_device() ASoC: wm8962: Unconditionally wait for the FLL to lock ASoC: wm5102: Split input PGA controls ASoC: wm5110: Split input PGA controls ASoC: arizona: Add noise gate hold time enumeration ASoC: wm5102: Add noise gate control ASoC: wm5110: Add noise gate control ASoC: arizona: Implement tristate support ASoC: wm2000: Integrate with clock API ASoC: wm2000: Use clock API integration to configure MCLK divisor Merge branch 'asoc-fix-cs4271' into asoc-cs4271 ASoC: wm5102: Implement routing and power management for ISRCs ASoC: arizona: Allow runtime reconfiguration of the output mode Merge branch 'fix/arizona' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: arizona: Factor out rate selection code Merge remote-tracking branch 'asoc/fix/pxa' into asoc-pxa ASoC: wm_adsp: Add support for parsing algorithms ASoC: wm_adsp: Factor out calculation of memory base addresses ASoC: wm_asdp: Validate sanity of algorithm count ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks ASoC: wm_adsp: Add basic firmware selection support Merge remote-tracking branch 'asoc/topic/adsp' into asoc-arizona Merge remote-tracking branch 'asoc/fix/arizona' into asoc-arizona Merge branch 'topic/fsi' of git://git.kernel.org/.../broonie/sound into asoc-simple-card ASoC: wm_adsp: Add initialisation function for ADSP1 Merge remote-tracking branch 'asoc/topic/adsp' into asoc-wm2200 ASoC: wm2200: Initialise the ADSPs ASoC: arizona: Don't request FLL lock IRQ Merge remote-tracking branch 'asoc/fix/arizona' into asoc-arizona ASoC: wm_adsp: Implement support for coefficeint file format 1 ASoC: wm2200: Add controls for firmware enumeration ASoC: wm5102: Add controls for firmware selection Merge remote-tracking branch 'asoc/topic/adsp' into asoc-wm2200 ASoC: wm5100: Implement DRC, EQ and LHPF coefficient configuration ASoC: wm2200: Implement EQ and LHPF coefficient configuration ASoC: wm_adsp: Make region identification errors more informative ASoC: wm_adsp: Use GFP_DMA for algorithm readback ASoC: arizona: Support clearing clocks Merge remote-tracking branch 'asoc/fix/adsp' into asoc-adsp ASoC: wm_adsp: Provide explicit trace of coefficient writes ASoC: arizona: Allow number of channels clocked to be restricted ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory ASoC: wm_adsp: Correct handling of some coefficeint blocks ASoC: wm2000: Expose some additional registers ASoC: wm_adsp: Add speaker Tx as a firmware option ASoC: wm_adsp: Accept 0 as a parameter block address ASoC: wm_adsp: Ensure ADSP2 DMAs are quiesced when DSP is halted Merge remote-tracking branch 'asoc/fix/adsp' into asoc-adsp regmap: Add asynchronous I/O support regmap: spi: Support asynchronous I/O for SPI ASoC: wm_adsp: Fix support for firmware switching on DSP2 and higher ASoC: wm2000: Lock state machine updates ASoC: wm2000: Expose ANC gain adjustment Merge tag 'async' of git://git.kernel.org/.../broonie/regmap into asoc-adsp ASoC: wm_adsp: Use asynchronous I/O to write firmware and coefficients regmap: Export regmap_async_complete() Merge tag 'async' of git://git.kernel.org/.../broonie/regmap into asoc-adsp ASoC: arizona: Fix debug logging level for FLLs and AIFs ASoC: compress: Only mute playback streams ASoC: core: Allow digital mute for capture Merge remote-tracking branch 'asoc/fix/adsp' into asoc-next Merge remote-tracking branch 'asoc/fix/arizona' into asoc-next Merge remote-tracking branch 'asoc/fix/davinci' into asoc-next Merge remote-tracking branch 'asoc/fix/dwc' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/fix/ux500' into asoc-next Merge remote-tracking branch 'asoc/fix/wm2000' into asoc-next Merge remote-tracking branch 'asoc/fix/wm2200' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l52' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/of' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/simple-card' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320dac33' into asoc-next Merge remote-tracking branch 'asoc/topic/twl' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2200' into asoc-next Merge remote-tracking branch 'asoc/topic/wm5100' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8350' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8804' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8985' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next Michal Bachraty (1): ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback Mike Dunn (3): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe ALSA: pxa27x: rename pxa27x_assert_ac97reset() Misael Lopez Cruz (1): ASoC: twl6040: Convert PLUGINT to no-suspend irq Padmavathi Venna (10): ASoC: samsung: Rename samsung i2s secondary device name ASoC: Samsung: Add I2S S/W RST in startup function ASoC: samsung: Add DT support for i2s ARM: SAMSUNG: Make dma request compatible to generic dma bindings. spi: s3c64xx: Modify SPI driver to use generic DMA DT support ARM: dts: Add nodes for i2s controllers for Samsung Exynos5 platforms ARM: EXYNOS: Enable platform support for I2S controllers ASoC: SMDK: WM8994: Add device tree support for machine file ARM: dts: Modify SPI nodes according generic DMA DT bindings ASoC: Samsung: Update Kconfig for I2S,SPDIF and PCM audio Peter Ujfalusi (15): ASoC: twl4030: Correct the support for Voice port ASoC: zoom2: No need to configure the Voice port anymore ASoC: sdp3430: No need to configure the Voice port anymore ASoC: twl4030: Convert MICBIAS to SUPPLY widget ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use ASoC: sdp3430: No need to configure pin mux for extmute ASoC: omap-twl4030: Add support for routing, voice port and jack detect ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 ASoC: twl4030: Remove suspend/resume soc driver operations ASoC: twl6040: Only set the bias_level once in twl6040_resume() ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation ASoC: tlv320dac33: Remove suspend/resume soc driver operations ASoC: omap-pcm: No need to set constraint at open time R. Chandrasekar (1): ASoC: Samsung: I2S: Add support for runtime S2R Ricardo Neri (1): ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word Ryo Tsutsui (1): ASoC: arizona: Fixed a bug in FLL fractional calculation Sachin Kamat (4): ASoC: tegra_wm9712: Remove __devinitconst attribute ASoC: tegra20_ac97: Remove __devinitconst attribute ASoC: tegra: Use NULL instead of 0 for pointers ASoC: tegra: Staticize some functions in tegra30_i2s.c Stephen Rothwell (1): ASoC: ak4642: remove __devinitconst annotation Stephen Warren (1): regmap: include linux/sched.h to fix build Tejun Heo (1): ASoC: wm8350: don't use [delayed_]work_pending() Thierry Reding (1): ASoC: fsi: Remove __devinitconst Documentation/devicetree/bindings/sound/ak4642.txt | 17 + Documentation/devicetree/bindings/sound/cs4271.txt | 12 + .../bindings/sound/nvidia,tegra-audio-wm9712.txt | 51 + .../bindings/sound/nvidia,tegra20-ac97.txt | 22 + .../devicetree/bindings/sound/omap-twl4030.txt | 46 + .../devicetree/bindings/sound/renesas,fsi.txt | 26 + .../bindings/sound/samsung,smdk-wm8994.txt | 14 + .../devicetree/bindings/sound/samsung-i2s.txt | 63 + .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + Documentation/devicetree/bindings/sound/wm8962.txt | 16 + arch/arm/boot/dts/exynos5250-smdk5250.dts | 26 + arch/arm/boot/dts/exynos5250.dtsi | 44 +- arch/arm/mach-exynos/mach-exynos5-dt.c | 6 + arch/arm/mach-pxa/pxa27x.c | 20 +- arch/arm/mach-shmobile/board-ap4evb.c | 42 +- arch/arm/mach-shmobile/board-armadillo800eva.c | 34 +- arch/arm/mach-shmobile/board-kzm9g.c | 20 +- arch/arm/mach-shmobile/board-mackerel.c | 42 +- arch/arm/plat-samsung/dma-ops.c | 10 +- arch/arm/plat-samsung/include/plat/dma-ops.h | 3 +- arch/arm/plat-samsung/s3c-dma-ops.c | 3 +- arch/sh/boards/mach-ecovec24/setup.c | 27 +- arch/sh/boards/mach-se/7724/setup.c | 29 +- drivers/base/regmap/internal.h | 18 + drivers/base/regmap/regmap-spi.c | 52 + drivers/base/regmap/regmap.c | 301 ++- drivers/misc/Kconfig | 2 +- drivers/misc/atmel-ssc.c | 2 +- drivers/spi/spi-s3c64xx.c | 77 +- include/linux/mfd/arizona/pdata.h | 9 + include/linux/regmap.h | 28 + include/sound/cs4271.h | 15 + include/sound/max98090.h | 29 + include/sound/saif.h | 16 - include/sound/sh_fsi.h | 70 +- include/sound/simple_card.h | 12 +- include/sound/soc-dai.h | 8 +- include/sound/soc.h | 6 +- include/sound/tlv320aic3x.h | 10 + include/sound/wm2000.h | 3 - include/sound/wm2200.h | 22 +- sound/arm/pxa2xx-ac97-lib.c | 8 +- sound/soc/atmel/Kconfig | 6 +- sound/soc/atmel/atmel-pcm-pdc.c | 4 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel-pcm.h | 6 +- sound/soc/atmel/atmel_ssc_dai.c | 14 +- sound/soc/atmel/sam9g20_wm8731.c | 6 +- sound/soc/codecs/ak4642.c | 33 +- sound/soc/codecs/arizona.c | 247 +- sound/soc/codecs/arizona.h | 8 +- sound/soc/codecs/cs4271.c | 34 + sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/max98090.c | 2685 ++++++++++++++++---- sound/soc/codecs/max98090.h | 1549 +++++++++++ sound/soc/codecs/tlv320aic3x.c | 87 +- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/codecs/tlv320dac33.c | 16 - sound/soc/codecs/twl4030.c | 85 +- sound/soc/codecs/twl6040.c | 62 +- sound/soc/codecs/wm2000.c | 66 +- sound/soc/codecs/wm2000.h | 3 + sound/soc/codecs/wm2200.c | 62 +- sound/soc/codecs/wm5100.c | 13 + sound/soc/codecs/wm5102.c | 172 +- sound/soc/codecs/wm5110.c | 111 +- sound/soc/codecs/wm8350.c | 10 +- sound/soc/codecs/wm8804.c | 3 - sound/soc/codecs/wm8962.c | 37 +- sound/soc/codecs/wm8983.c | 41 +- sound/soc/codecs/wm8985.c | 43 +- sound/soc/codecs/wm_adsp.c | 525 +++- sound/soc/codecs/wm_adsp.h | 18 + sound/soc/codecs/wmfw.h | 15 +- sound/soc/davinci/davinci-evm.c | 6 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/dwc/designware_i2s.c | 4 + sound/soc/generic/simple-card.c | 63 +- sound/soc/mxs/mxs-saif.c | 49 +- sound/soc/omap/Kconfig | 19 +- sound/soc/omap/Makefile | 4 - sound/soc/omap/n810.c | 4 +- sound/soc/omap/omap-hdmi.c | 2 + sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-twl4030.c | 204 +- sound/soc/omap/omap3pandora.c | 8 +- sound/soc/omap/rx51.c | 8 +- sound/soc/omap/sdp3430.c | 278 -- sound/soc/omap/zoom2.c | 207 -- sound/soc/pxa/palm27x.c | 38 +- sound/soc/samsung/Kconfig | 6 +- sound/soc/samsung/dma.c | 3 +- sound/soc/samsung/dma.h | 1 + sound/soc/samsung/i2s.c | 267 +- sound/soc/samsung/i2s.h | 7 - sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 7 +- sound/soc/samsung/smdk_wm8994.c | 30 +- sound/soc/sh/fsi.c | 242 +- sound/soc/soc-compress.c | 123 +- sound/soc/soc-core.c | 135 +- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-pcm.c | 19 +- sound/soc/tegra/Kconfig | 19 + sound/soc/tegra/Makefile | 4 + sound/soc/tegra/tegra20_ac97.c | 480 ++++ sound/soc/tegra/tegra20_ac97.h | 95 + sound/soc/tegra/tegra20_das.c | 13 + sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/tegra/tegra_asoc_utils.c | 53 + sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_wm9712.c | 176 ++ sound/soc/ux500/mop500.c | 2 +- 114 files changed, 7824 insertions(+), 2023 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,fsi.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt create mode 100755 include/sound/max98090.h delete mode 100644 include/sound/saif.h mode change 100644 => 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h delete mode 100644 sound/soc/omap/sdp3430.c delete mode 100644 sound/soc/omap/zoom2.c create mode 100644 sound/soc/tegra/tegra20_ac97.c create mode 100644 sound/soc/tegra/tegra20_ac97.h create mode 100644 sound/soc/tegra/tegra_wm9712.c

2 1

Re: [alsa-devel] Sound record on Devkit8000 with DT boot issue
by Peter Ujfalusi 11 Feb '13

11 Feb '13

On 02/10/2013 04:21 PM, Anil Kumar wrote: > Hi All, > > I am working on Devkit8000 and adding DT support for that. I am trying > to add sound DT node support for this boards. > I have nearly done with this but facing issue on record. I do not have > much knowledge of twl4030 audio codec but with > small Hack[1] (as MainMic connected to the input jack) i am able to > record sound on Devkit8000. > I do not know that MICBIAS1_EN can be enable by amixer commands. > Could you please help me to get out of this ? You need to pass the routing information via DT for Devkit8000. This is not done for the BeagleBoard since it works without it (no need for MICBIAS for example) > > > 1) Hack [1] > > diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c > index 63b280b..6780681 100644 > --- a/sound/soc/codecs/twl4030.c > +++ b/sound/soc/codecs/twl4030.c > @@ -1473,7 +1473,7 @@ static const struct snd_soc_dapm_widget > twl4030_dapm_widgets[] = { > SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0, > NULL, 0), > > - SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), > + SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 1), Yes, this is HACK and it is wrong at the same time. > SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), > SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), > > > 2) Sound DT node patch for Devkit8000 > > diff --git a/arch/arm/boot/dts/omap3- > devkit8000.dts b/arch/arm/boot/dts/omap3-devkit8000.dts > index dc59272..5f3eec0 100644 > --- a/arch/arm/boot/dts/omap3-devkit8000.dts > +++ b/arch/arm/boot/dts/omap3-devkit8000.dts > @@ -44,11 +44,22 @@ > }; > > }; > + > + sound { > + compatible = "ti,omap-twl4030"; > + ti,model = "devkit8000"; > + > + ti,mcbsp = <&mcbsp2>; > + ti,codec = <&twl_audio>; Here you should have something like: ti,audio-routing = "Ext Spk", "PREDRIVEL", "Ext Spk", "PREDRIVER", "MAINMIC", "Main Mic", "Main Mic", "Mic Bias 1"; So you connect the MicBias1 to Main Mic. When you start playback MicBias1 will be turned on for you. > + }; > }; > > &omap3_pmx_core { > pinctrl-names = "default"; > - pinctrl-0 = <&i2c1_pins>; > + pinctrl-0 = < > + &i2c1_pins > + &mcbsp2_pins > + >; > > leds_pins: pinmux_led_pins { > pinctrl-single,pins = < > @@ -65,6 +76,21 @@ > 0x18c 0x118 /* I2C1_SDA */ > >; > }; > + > + mcbsp2_pins: pinmux_mcbsp2_pins { > + pinctrl-single,pins = < > + /* > + * MCBSP2_FSX, OMAP_MUX_MODE0 | OMAP_PIN_INPUT, > + * MCBSP2_CLKX, OMAP_MUX_MODE0 | OMAP_PIN_INPUT > + */ > + 0x10c 0x01000100 > + /* > + * MCBSP2_DR, OMAP_MUX_MODE0 | OMAP_PIN_INPUT, > + * MCBSP2_DX, OMAP_MUX_MODE0 | OMAP_PIN_OUTPUT > + */ > + 0x110 0x00000100 > + >; > + }; > }; > > &i2c1 { > @@ -74,6 +100,14 @@ > reg = <0x48>; > interrupts = <7>; /* SYS_NIRQ cascaded to intc */ > interrupt-parent = <&intc>; > + > + twl_audio: audio { > + compatible = "ti,twl4030-audio"; > + codec { > + ti,ramp_delay_value = <3>; > + ti,offset_cncl_path = <0x31>; > + }; > + }; > }; > }; > > 3) Playback amixer commands > amixer set 'PredriveR Mixer AudioR2' on > amixer set 'PredriveL Mixer AudioL2' on > amixer set PreDriv 100 unmute > amixer set 'DAC2 Digital Fine' 100 > > Thanks, > Anil > -- Péter

1 0

[alsa-devel] [PATCH RFC 00/20] More patches for HD-audio generic auto-parser
by Takashi Iwai 11 Feb '13

11 Feb '13

Hi, this is a series of patches for more enhancements of HD-audio generic parser I posted shortly ago. All patches are found in sound-unstable git tree test/hda-gen-parser branch (and merged to master branch, too). git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Takashi

2 22

[alsa-devel] [PATCH] ASoC: wm_adsp: round to 4-byte boundary for coeff file blocks
by Chris Rattray 11 Feb '13

11 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index be45e2b..f3f7e75 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -711,6 +711,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; + int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -842,7 +843,12 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - pos += le32_to_cpu(blk->len) + sizeof(*blk); + tmp = le32_to_cpu(blk->len) % 4; + if (tmp) + pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); + else + pos += le32_to_cpu(blk->len) + sizeof(*blk); + blocks++; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH] Add M2Tech hiFace USB-SPDIF driver
by Antonio Ospite 11 Feb '13

11 Feb '13

Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- Hi, this is my first ALSA driver I still have some doubts and I marked them in the comments below with a XXX prefix, could someone please take a look at them? I plan on submitting a version 2 of the driver after I get some feedback, especially on those doubts. I also noticed that many drivers under sound/usb/ have text in the Kconfig entry stating "Say Y here to include support for ..." but I only see "[N/m/?]" as options when I run "make oldconfig" maybe because of some dependency, I am not sure, should such text be removed in order to avoid confusion? Thanks, Antonio sound/usb/Kconfig | 33 +++ sound/usb/Makefile | 2 +- sound/usb/hiface/Makefile | 2 + sound/usb/hiface/chip.c | 331 ++++++++++++++++++++++ sound/usb/hiface/chip.h | 34 +++ sound/usb/hiface/pcm.c | 666 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/hiface/pcm.h | 24 ++ 7 files changed, 1091 insertions(+), 1 deletion(-) create mode 100644 sound/usb/hiface/Makefile create mode 100644 sound/usb/hiface/chip.c create mode 100644 sound/usb/hiface/chip.h create mode 100644 sound/usb/hiface/pcm.c create mode 100644 sound/usb/hiface/pcm.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd7..2b206e8 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,38 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Say Y here to include support for M2Tech hiFace USB-SPDIF interface. + + M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output + Interface, see http://www.m2tech.biz/hiface.html + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc..abe668f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 0000000..463b136 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 0000000..dce74e9 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,331 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/initval.h> + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_AUTHOR("Antonio Ospite <ao2(a)amarulasolutions.com>"); +MODULE_DESCRIPTION("M2Tech HiFace USB audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{HiFace, Evo}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "HiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static struct hiface_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_dev_free(struct snd_device *device) +{ + struct hiface_chip *chip = device->device_data; + kfree(chip); + return 0; +} + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + static struct snd_device_ops ops = { + .dev_free = hiface_dev_free, + }; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot create alsa card.\n"); + return ret; + } + + strcpy(card->driver, DRIVER_NAME); + + if (quirk && quirk->device_name) + strcpy(card->shortname, quirk->device_name); + else + strcpy(card->shortname, "M2Tech generic audio"); + + sprintf(card->longname, "%s at %d:%d", card->shortname, + device->bus->busnum, device->devnum); + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_card_free(card); + return -ENOMEM; + } + + chip->dev = device; + chip->index = idx; + chip->card = card; + + ret = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (ret < 0) { + kfree(chip); + snd_card_free(card); + return ret; + } + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + pr_info("Probe " DRIVER_NAME " driver.\n"); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + snd_printk(KERN_ERR "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(&register_mutex); + for (i = 0; i < SNDRV_CARDS; i++) { + if (chips[i] && chips[i]->dev == device) { + if (chips[i]->shutdown) { + snd_printk(KERN_ERR CARD_NAME " device is in the shutdown state, cannot create a card instance\n"); + ret = -ENODEV; + goto err; + } + chip = chips[i]; + break; + } + } + if (!chip) { + /* it's a fresh one. + * now look for an empty slot and create a new card instance + */ + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i] && !chips[i]) { + ret = hiface_chip_create(device, i, quirk, + &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + break; + } + if (!chip) { + snd_printk(KERN_ERR "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + } + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + chips[chip->index] = chip; + chip->intf_count++; + + mutex_unlock(&register_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(&register_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + pr_debug("%s: called.\n", __func__); + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + chip->intf_count--; + if (chip->intf_count <= 0) { + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + mutex_lock(&register_mutex); + chips[chip->index] = NULL; + mutex_unlock(&register_mutex); + + chip->shutdown = true; + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); + } +} + +static struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 0000000..c2e9fda --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,34 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include <linux/usb.h> +#include <sound/core.h> + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + int intf_count; /* number of registered interfaces */ + int index; /* index in module parameter arrays */ + bool shutdown; + + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 0000000..34fe0ed --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,666 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/slab.h> +#include <sound/pcm.h> + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_MAX_PACKET_SIZE 4096 +#define MAX_BUFSIZE (2 * PCM_N_URBS * PCM_MAX_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 rate; /* one of PCM_RATE_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates) - 2, /* by default rates up to 192000 are supported */ + .list = rates, + .mask = 0, +}; + +static const int rates_alsaid[] = { + SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000, + SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000, + SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000, + SNDRV_PCM_RATE_KNOT, SNDRV_PCM_RATE_KNOT }; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_BUFSIZE, + .period_bytes_min = PCM_MAX_PACKET_SIZE, + .period_bytes_max = MAX_BUFSIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + u8 rate_value[] = { 0x43, 0x4b, 0x42, 0x4a, 0x40, 0x48, 0x58, 0x68 }; + int ret; + struct usb_device *device = rt->chip->dev; + + /* We are already sure that the rate is supported here thanks to the + * contraints set in hiface_pcm_open(), but we have to retrieve the + * index to access rate_value. + */ + for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + if (rate == rates[rt->rate]) + break; + + if (unlikely(rt->rate == ARRAY_SIZE(rates))) { + pr_err("Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + 0xb0, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value[rt->rate], 0, NULL, 0, 100); + if (ret < 0) { + snd_printk(KERN_ERR "Error setting samplerate %d.\n", + rates[rt->rate]); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream( + struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + pr_debug("Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0, i; + + if (rt->stream_state == STREAM_DISABLED) { + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_MAX_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + pr_debug("%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + + +/* The hardware wants word-swapped 32-bit values */ +void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +static int hiface_pcm_playback(struct pcm_substream *sub, + struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + u8 *source; + unsigned int pcm_buffer_size; + + /* XXX Can an invalid format make it to here? + * Isn't ALSA checking pcm_hw.formats ? + */ + if (alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE) { + pr_err("Unsupported sample format\n"); + return -EINVAL; + } + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_MAX_PACKET_SIZE <= pcm_buffer_size) { + pr_debug("%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_MAX_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + pr_debug("%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_MAX_PACKET_SIZE - len); + } + sub->dma_off += PCM_MAX_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_MAX_PACKET_SIZE; + + return 0; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (usb_urb->status || rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) { + int ret; + + ret = hiface_pcm_playback(sub, out_urb); + if (ret < 0) { + spin_unlock_irqrestore(&sub->lock, flags); + goto out_fail; + } + if (sub->period_off >= sub->instance->runtime->period_size) { + sub->period_off %= sub->instance->runtime->period_size; + spin_unlock_irqrestore(&sub->lock, flags); + snd_pcm_period_elapsed(sub->instance); + } else { + spin_unlock_irqrestore(&sub->lock, flags); + } + } else { + memset(out_urb->buffer, 0, PCM_MAX_PACKET_SIZE); + spin_unlock_irqrestore(&sub->lock, flags); + } +out_fail: + usb_submit_urb(&out_urb->instance, GFP_ATOMIC); +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + /* XXX can we avoid setting hw.rates below? + * + * FWIU setting alsa_rt->hw.rates in the open callback can be + * useful in two cases: + * + * - different hardware revisions have different + * capabilities + * + * - different stream types must have the same capabilities + * + * but in our case it should not be needed as we only have one + * playback stream and we define the capabilities via + * constraints already to avoid invalid rates to be used. + * + * Moreover, following the code stolen from 6fire it seems + * impossible to get to setting alsa_rt->hw.rates here: + * + * hiface_pcm_init: + * rt->rate = ARRAY_SIZE(rates); + * + * hiface_pcm_open: + * if (rt->rate < ARRAY_SIZE(rates)) + * alsa_rt->hw.rates = rates_alsaid[rt->rate]; + * // So it does not get here on the first open + * + * hiface_pcm_prepare: + * hiface_pcm_set_rate: + * for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + * if (rate == rates[rt->rate]) + * break; + * + * hiface_pcm_close: + * rt->rate = ARRAY_SIZE(rates); + * // So it won't set alsa_rt->hw.rates at the next open either + * + * So the only way to set alsa_rt->hw.rates is to have two + * calls to the open callback, is that even possible? ALSA + * does not allow sharing PCM streams AFAIK and we use only + * one playback stream. + * + * Maybe the 6fire driver was setting alsa_rt->hw.rates in + * order to have the playback stream and the capture stream + * use the same rate? I.e. the first open,prepare,set_rate() + * sequence on one type of stream will decide the rate and the + * second open on the other type would restrict the supported + * rate to the one already in use, but I am not sure. + * + * If the rationale makes sense then we don't need to set + * alsa_rt->hw.rates and we can remove this and related code. + */ + if (rt->rate < ARRAY_SIZE(rates)) + alsa_rt->hw.rates = rates_alsaid[rt->rate]; + + sub = &rt->playback; + } + + if (!sub) { + mutex_unlock(&rt->stream_mutex); + pr_err("Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rate_max = 384000; + constraints_rates.count = ARRAY_SIZE(rates); + } + + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + pr_debug("%s: called.\n", __func__); + + mutex_lock(&rt->stream_mutex); + if (sub) { + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + /* all substreams closed? if so, stop streaming */ + if (!rt->playback.instance) { + hiface_pcm_stream_stop(rt); + rt->rate = ARRAY_SIZE(rates); + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_free_pages(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&sub->lock, flags); + sub->active = true; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&sub->lock, flags); + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_STATE_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_MAX_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + if (rt->playback.instance) { + snd_pcm_stop(rt->playback.instance, + SNDRV_PCM_STATE_XRUN); + } + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + pr_debug("%s: called.\n", __func__); + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + rt->rate = ARRAY_SIZE(rates); + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + pr_err("Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strcpy(pcm->name, "USB-SPDIF Audio"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MAX_BUFSIZE, MAX_BUFSIZE); + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 0000000..ddc9fe7 --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH - add support for M-Audio FT C600 1/1] ALSA: usb-audio: add support for M-Audio FT C600
by Matt Gruskin 10 Feb '13

10 Feb '13

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e90daf8..638e7f7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -807,6 +807,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, { switch (cval->mixer->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (strcmp(kctl->id.name, "Effect Duration") == 0) { cval->min = 0x0000; cval->max = 0xffff; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0e2ed3d..cc2dd1f 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -380,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .selector_map = c400_selectors, }, { + .id = USB_ID(0x0763, 0x2031), + .selector_map = c400_selectors, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 15520de..497d274 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -637,7 +637,7 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, } /* M-Audio FastTrack Ultra quirks */ -/* FTU Effect switch (also used by C400) */ +/* FTU Effect switch (also used by C400/C600) */ struct snd_ftu_eff_switch_priv_val { struct usb_mixer_interface *mixer; int cached_value; @@ -1029,32 +1029,45 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } -/* M-Audio Fast Track C400 */ -/* C400 volume controls, this control needs a volume quirk, see mixer.c */ +/* M-Audio Fast Track C400/C600 */ +/* C400/C600 volume controls, this control needs a volume quirk, see mixer.c */ static int snd_c400_create_vol_ctls(struct usb_mixer_interface *mixer) { char name[64]; unsigned int cmask, offset; int out, chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - for (out = 0; out < 6; out++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + for (out = 0; out < num_outs; out++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "PCM%d-Out%d Playback Volume", chan + 1, out + 1); } else { snprintf(name, sizeof(name), "In%d-Out%d Playback Volume", - chan - 5, out + 1); + chan - num_outs + 1, out + 1); } cmask = (out == 0) ? 0 : 1 << (out - 1); - offset = chan * 6; + offset = chan * num_outs; err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1110,20 +1123,33 @@ static int snd_c400_create_effect_vol_ctls(struct usb_mixer_interface *mixer) char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x42; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "Effect Send DOut%d", chan + 1); } else { snprintf(name, sizeof(name), "Effect Send AIn%d", - chan - 5); + chan - num_outs + 1); } cmask = (chan == 0) ? 0 : 1 << (chan - 1); @@ -1142,20 +1168,33 @@ static int snd_c400_create_effect_ret_vol_ctls(struct usb_mixer_interface *mixer char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int offset = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - const int chan_id[6] = { 0, 7, 2, 9, 4, 0xb }; - const unsigned int offset = 0x3c; - /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ - for (chan = 0; chan < 6; chan++) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + offset = 0x3c; + /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + offset = 0x70; + /* { 0x70, 0x79, 0x72, 0x7b, 0x74, 0x7d, 0x76, 0x7f } */ + break; + } + + for (chan = 0; chan < num_outs; chan++) { snprintf(name, sizeof(name), "Effect Return %d", chan + 1); - cmask = (chan_id[chan] == 0) ? 0 : 1 << (chan_id[chan] - 1); + cmask = (chan == 0) ? 0 : + 1 << (chan + (chan % 2) * num_outs - 1); err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1299,6 +1338,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); break; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d82e378..749752a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -363,6 +363,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (is_playback) { implicit_fb = 1; ep = 0x81; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 64d25a7..4901400 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2326,6 +2326,77 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0763, 0x2031), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + /* .vendor_name = "M-Audio", */ + /* .product_name = "Fast Track C600", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* Capture */ + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* MIDI */ + { + .ifnum = -1 /* Interface = 4 */ + } + } + } +}, +{ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c97185..7d7ad0b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -863,13 +863,14 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 4; /* - * M-Audio Fast Track C400 - when packets are not skipped, real world - * latency varies by approx. +/- 50 frames (at 96KHz) each time the - * stream is (re)started. When skipping packets 16 at endpoint start - * up, the real world latency is stable within +/- 1 frame (also + * M-Audio Fast Track C400/C600 - when packets are not skipped, real + * world latency varies by approx. +/- 50 frames (at 96KHz) each time + * the stream is (re)started. When skipping packets 16 at endpoint + * start up, the real world latency is stable within +/- 1 frame (also * across power cycles). */ - if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) && + if ((ep->chip->usb_id == USB_ID(0x0763, 0x2030) || + ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; } -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 2/5] hda_intel: Add Device IDs for Intel Wellsburg PCH
by Takashi Iwai 10 Feb '13

10 Feb '13

At Fri, 8 Feb 2013 17:29:40 -0800, James Ralston wrote: > > This patch adds the HD Audio Device IDs for the Intel Wellsburg PCH > > Signed-off-by: James Ralston <james.d.ralston(a)intel.com> Thanks, applied. Takashi > --- > sound/pci/hda/hda_intel.c | 5 +++++ > 1 files changed, 5 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index c78286f..3218f4a 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -3611,6 +3611,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > + /* Wellsburg */ > + { PCI_DEVICE(0x8086, 0x8d20), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > + { PCI_DEVICE(0x8086, 0x8d21), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > /* Lynx Point-LP */ > { PCI_DEVICE(0x8086, 0x9c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > -- > 1.7.7.6 >

1 0

[alsa-devel] [PATCH - add support for M-Audio FT C600 1/1] ALSA: usb-audio: add support for M-Audio FT C600
by Matt Gruskin 09 Feb '13

09 Feb '13

Adds quirks and mixer support for the M-Audio Fast Track C600 USB audio interface. This device is very similar to the C400 - the C600 simply has some more inputs and outputs, so the existing C400 support is extended to support this device as well. Signed-off-by: Matt Gruskin <matthew.gruskin(a)gmail.com> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e90daf8..638e7f7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -807,6 +807,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, { switch (cval->mixer->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (strcmp(kctl->id.name, "Effect Duration") == 0) { cval->min = 0x0000; cval->max = 0xffff; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0e2ed3d..cc2dd1f 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -380,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .selector_map = c400_selectors, }, { + .id = USB_ID(0x0763, 0x2031), + .selector_map = c400_selectors, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 15520de..497d274 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -637,7 +637,7 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, } /* M-Audio FastTrack Ultra quirks */ -/* FTU Effect switch (also used by C400) */ +/* FTU Effect switch (also used by C400/C600) */ struct snd_ftu_eff_switch_priv_val { struct usb_mixer_interface *mixer; int cached_value; @@ -1029,32 +1029,45 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } -/* M-Audio Fast Track C400 */ -/* C400 volume controls, this control needs a volume quirk, see mixer.c */ +/* M-Audio Fast Track C400/C600 */ +/* C400/C600 volume controls, this control needs a volume quirk, see mixer.c */ static int snd_c400_create_vol_ctls(struct usb_mixer_interface *mixer) { char name[64]; unsigned int cmask, offset; int out, chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - for (out = 0; out < 6; out++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + for (out = 0; out < num_outs; out++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "PCM%d-Out%d Playback Volume", chan + 1, out + 1); } else { snprintf(name, sizeof(name), "In%d-Out%d Playback Volume", - chan - 5, out + 1); + chan - num_outs + 1, out + 1); } cmask = (out == 0) ? 0 : 1 << (out - 1); - offset = chan * 6; + offset = chan * num_outs; err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1110,20 +1123,33 @@ static int snd_c400_create_effect_vol_ctls(struct usb_mixer_interface *mixer) char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x42; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "Effect Send DOut%d", chan + 1); } else { snprintf(name, sizeof(name), "Effect Send AIn%d", - chan - 5); + chan - num_outs + 1); } cmask = (chan == 0) ? 0 : 1 << (chan - 1); @@ -1142,20 +1168,33 @@ static int snd_c400_create_effect_ret_vol_ctls(struct usb_mixer_interface *mixer char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int offset = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - const int chan_id[6] = { 0, 7, 2, 9, 4, 0xb }; - const unsigned int offset = 0x3c; - /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ - for (chan = 0; chan < 6; chan++) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + offset = 0x3c; + /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + offset = 0x70; + /* { 0x70, 0x79, 0x72, 0x7b, 0x74, 0x7d, 0x76, 0x7f } */ + break; + } + + for (chan = 0; chan < num_outs; chan++) { snprintf(name, sizeof(name), "Effect Return %d", chan + 1); - cmask = (chan_id[chan] == 0) ? 0 : 1 << (chan_id[chan] - 1); + cmask = (chan == 0) ? 0 : + 1 << (chan + (chan % 2) * num_outs - 1); err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1299,6 +1338,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); break; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d82e378..749752a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -363,6 +363,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (is_playback) { implicit_fb = 1; ep = 0x81; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 64d25a7..4901400 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2326,6 +2326,77 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0763, 0x2031), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + /* .vendor_name = "M-Audio", */ + /* .product_name = "Fast Track C600", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* Capture */ + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* MIDI */ + { + .ifnum = -1 /* Interface = 4 */ + } + } + } +}, +{ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c97185..7d7ad0b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -863,13 +863,14 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 4; /* - * M-Audio Fast Track C400 - when packets are not skipped, real world - * latency varies by approx. +/- 50 frames (at 96KHz) each time the - * stream is (re)started. When skipping packets 16 at endpoint start - * up, the real world latency is stable within +/- 1 frame (also + * M-Audio Fast Track C400/C600 - when packets are not skipped, real + * world latency varies by approx. +/- 50 frames (at 96KHz) each time + * the stream is (re)started. When skipping packets 16 at endpoint + * start up, the real world latency is stable within +/- 1 frame (also * across power cycles). */ - if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) && + if ((ep->chip->usb_id == USB_ID(0x0763, 0x2030) || + ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; } -- 1.7.9.5

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[alsa-devel] [PATCH 3/3] ALSA: hda - Yet another fix for broken HSW HDMI pin connections
by mengdong.lin＠intel.com 08 Feb '13

08 Feb '13

From: Takashi Iwai <tiwai(a)suse.de> A Haswell test machine showed that the invalid connection list, but this time it has only a single pin on the codec, thus the former fixup code doesn't work as it assumes the three pins blindly. This patch splits the former fixup code to two parts: - Enable eDP 1.2 for Haswell codec - Fix the connection list of pins on Haswell codec; the converter list is recorded dynamically in hdmi_add_cvt(), and applied in hdmi_add_pin() Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 54243c4..b9af281b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -84,6 +84,7 @@ struct hdmi_spec_per_pin { struct hdmi_spec { int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; + hda_nid_t cvt_nids[MAX_HDMI_CVTS]; int num_pins; struct hdmi_spec_per_pin pins[MAX_HDMI_PINS]; @@ -1197,6 +1198,9 @@ static void hdmi_repoll_eld(struct work_struct *work) hdmi_present_sense(per_pin, per_pin->repoll_count); } +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + hda_nid_t nid); + static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) { struct hdmi_spec *spec = codec->spec; @@ -1216,6 +1220,9 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (snd_BUG_ON(spec->num_pins >= MAX_HDMI_PINS)) return -E2BIG; + if (codec->vendor_id == 0x80862807) + intel_haswell_fixup_connect_list(codec, pin_nid); + pin_idx = spec->num_pins; per_pin = &spec->pins[pin_idx]; @@ -1263,7 +1270,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) if (err < 0) return err; - spec->num_cvts++; + spec->cvt_nids[spec->num_cvts++] = cvt_nid; return 0; } @@ -1691,27 +1698,22 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; -static void intel_haswell_fixup_connect_list(struct hda_codec *codec) -{ - unsigned int vendor_param; - hda_nid_t list[3] = {0x2, 0x3, 0x4}; - vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); - if (vendor_param == -1 || vendor_param & 0x02) - return; - - /* enable DP1.2 mode */ - vendor_param |= 0x02; - snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + hda_nid_t nid) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t conns[4]; + int nconns; - vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); - if (vendor_param == -1 || !(vendor_param & 0x02)) + nconns = snd_hda_get_connections(codec, nid, conns, ARRAY_SIZE(conns)); + if (nconns == spec->num_cvts && + !memcmp(conns, spec->cvt_nids, spec->num_cvts * sizeof(hda_nid_t))) return; - /* override 3 pins connection list */ - snd_hda_override_conn_list(codec, 0x05, 3, list); - snd_hda_override_conn_list(codec, 0x06, 3, list); - snd_hda_override_conn_list(codec, 0x07, 3, list); + /* override pins connection list */ + snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); + snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } #define INTEL_VENDOR_NID 0x08 @@ -1742,6 +1744,22 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, return; } +static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) +{ + unsigned int vendor_param; + + vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + INTEL_GET_VENDOR_VERB, 0); + if (vendor_param == -1 || vendor_param & INTEL_EN_DP12) + return; + + /* enable DP1.2 mode */ + vendor_param |= INTEL_EN_DP12; + snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0, + INTEL_SET_VENDOR_VERB, vendor_param); +} + + /* available models for fixup */ enum { @@ -1780,7 +1798,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); if (codec->vendor_id == 0x80862807) - intel_haswell_fixup_connect_list(codec); + intel_haswell_fixup_enable_dp12(codec); if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; -- 1.7.10.4

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[alsa-devel] [PATCH 2/3] [ALSA] hda - Add fixup for Haswell to enable all pin and convertor widgets
by mengdong.lin＠intel.com 08 Feb '13

08 Feb '13

From: Mengdong Lin <mengdong.lin(a)intel.com> Some Haswell machines support more than one display outputs (HDMI or DP), but its BIOS may not enable the codec's 2nd and 3rd pin and output cvt widgets. This patch implements a board-specific fixup for Intel Haswell Machines: If the hidden pins are not enabled by BIOS, the driver will enable them and call common code to update the codec tree. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 899c4fb..54243c4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1714,6 +1714,57 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec) snd_hda_override_conn_list(codec, 0x07, 3, list); } +#define INTEL_VENDOR_NID 0x08 +#define INTEL_GET_VENDOR_VERB 0xf81 +#define INTEL_SET_VENDOR_VERB 0x781 +#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ +#define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ + +static void intel_haswell_enable_all_pins(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + INTEL_GET_VENDOR_VERB, 0); + if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) + return; + + vendor_param |= INTEL_EN_ALL_PIN_CVTS; + vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + INTEL_SET_VENDOR_VERB, vendor_param); + if (vendor_param == -1) + return; + + snd_hda_codec_update_widgets(codec); + return; +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_enable_all_pins, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { @@ -1725,6 +1776,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (codec->vendor_id == 0x80862807) intel_haswell_fixup_connect_list(codec); -- 1.7.10.4

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[alsa-devel] channel names
by Adrian Knoth 08 Feb '13

08 Feb '13

Hi! Am I missing something or doesn't ALSA support channel names? When dealing with multichannel interfaces, it's pretty inconvenient to remember whether channel 193 is on MADI 3, AES-EBU or phones. We can address this on the jackd level, however, since OSX allows audio drivers to name their channels, I wonder how to do the same with ALSA? Cheers

4 5

[alsa-devel] [PATCH 1/3] [ALSA] hda - Support rereading widgets under the function group
by mengdong.lin＠intel.com 08 Feb '13

08 Feb '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A codec may allow software to hide some unused pin/cvt widgets. Sometimes BIOS does not enable the hidden widgets properly although they are needed for the board. Thus the driver need to enable them as a board-specific fixup and the whole tree will change. This patch implements a common code for rereading codec widgets. So the fixup code can call it after enabling the hidden widgets. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3c92514..e80f835 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1441,6 +1441,30 @@ int snd_hda_codec_new(struct hda_bus *bus, } EXPORT_SYMBOL_HDA(snd_hda_codec_new); +int snd_hda_codec_update_widgets(struct hda_codec *codec) +{ + hda_nid_t fg; + int err; + + /* Assume the function group node does not change, + * only the widget nodes may change. + */ + kfree(codec->wcaps); + fg = codec->afg ? codec->afg : codec->mfg; + err = read_widget_caps(codec, fg); + if (err < 0) { + snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); + return err; + } + + snd_array_free(&codec->init_pins); + err = read_pin_defaults(codec); + + return err; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_update_widgets); + + /** * snd_hda_codec_configure - (Re-)configure the HD-audio codec * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0be1826..e8c9442 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -935,6 +935,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); +int snd_hda_codec_update_widgets(struct hda_codec *codec); /* * low level functions -- 1.7.10.4

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Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 07 Feb '13

07 Feb '13

On Thu, Feb 07, 2013 at 11:37:38AM +0000, Richard Fitzgerald wrote: > > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > > > 2) Send SNDRV_COMPRESS_DRAIN > > The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which > > expects the decoder to completely drain its buffers and come to complete halt. > > This would also mean the framework will treat a drained stream as stopped and > > needs a new start. Certainly we dont want that in this case. So we can't use > > SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that > > would overtly complicate this. If we are not doing proper drian lets not > > call it that. > > Ok, so let's keep the partial drain but split out the NEXT_TRACK hint so we keep > the two operations separate. It's clearer to understand. NEXT_TRACK tells DSP to > prepare for data for new track, PARTIAL_DRAIN asks DSP when it has played all > data of current track. I don't think that application _must_ drain when doing > gapless, only if it cares to know when DSP reaches the join between tracks. If we split this then I dont see why next write/meta_data call should wait. Then the PARTIAL_DRAIN should be used for knowing in userpsace that switch happened, right? > Possible problem - application calls NEXT_TRACK and PARTIAL_DRAIN when DSP has > already reached end of current track, so what does PARTIAL_DRAIN mean in this > case? It should just return immediately, and if your output is starved you are in error state, that can happen when you dont give data as well. > Also can we make metadata a key-pair list as Takashi suggested, with unlimited > length. Something like > > struct snd_compr_metadata_pair { > enum snd_compr_metadata_key key; > u32 value; > }; No enums... Thats not how kernel ABI is designed. I am leaning towards key/value pair, but is this fair assumption that value is always 32bit, what if we have larger single value, then should we split? > > struct snd_compr_metadata { > int count; /* number of actual entries in following array */ > struct snd_compr_metadata_pair *pairs; > }; > > Should we union the u32 value with a u8* buf to allow for any case where the > value needs to be a buffer of binary data? Is there any known case like this? Again nope, that aint good from ABI POV. What I think it should be is: enum { SNDRV_COMPRESS_ENCODER_PADDING = 1, SNDRV_COMPRESS_ENCODER_DELAY = 2, }; struct snd_compr_metadata { u32 key; u32 value; }; -- ~Vinod

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Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 07 Feb '13

07 Feb '13

On Thu, Feb 07, 2013 at 01:15:54AM +0000, Richard Fitzgerald wrote: > > +- partial drain > > +This is called when end of file is reached. The userspace can inform DSP that > > +EOF is reached and now DSP can start skipping padding delay. Also next write > > +data would belong to next track > > We're really doing two different tasks inside the "partial drain". The name, > and the reason we are doing two tasks in one, comes from a particular higher- > layer scenario, but there's no reason the driver API need use the same > terminology as one particular application of the functionality. > The two tasks are: > > 1) Tell the DSP that we have sent all data for the current track and following > data will be for the next track. This lets the DSP lay down a marker for where > it should strip padding at the end of a track, and know it should be expecting > more data for another track to follow gaplessly where it must strip the > encoder delay from the start. > > 2) Ask for notification when DSP has reached the changeover point between the > playback of the two tracks > > I think it would be more logical and less confusing not to combine the two into > a single ioctl. Instead, add only one new ioctl specifically to provide the > track changeover hint, something like SNDRV_COMPRESS_NEXT_TRACK (meaning DSP > should expect data for next track to follow). Don't add a new drain, just use > the existing SNDRV_COMPRESS_DRAIN - the driver/DSP can make a decision whether > it needs to do something special with the drain if we have told it that we > will be sending it some more data for a following track. > > So the SNDRV_COMPRESS_PARTIAL_DRAIN in this patch would become > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > 2) Send SNDRV_COMPRESS_DRAIN The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which expects the decoder to completely drain its buffers and come to complete halt. This would also mean the framework will treat a drained stream as stopped and needs a new start. Certainly we dont want that in this case. So we can't use SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that would overtly complicate this. If we are not doing proper drian lets not call it that. But I think I like the idea of splitting the two up to do a cleaner interface. Let me check this... > When we reach the final track we just do SNDRV_COMPRESS_DRAIN to wait for it > to finish. > > If we setup a next track by doing SNDRV_COMPRESS_NEXT_TRACK and then change > our mind and decide that this is going to be the last track, we do a > SNDRV_COMPRESS_DRAIN without sending any next track data, then we do > SNDRV_COMPRESS_STOP Nope that is where we would have issue. You dont call SNDRV_COMPRESS_STOP for compressed streams. You have to wait till DSP has decoded and rendered data for the last track which can be done by using SNDRV_COMPRESS_DRAIN only. SNDRV_COMPRESS_STOP should not be called in this case. -- ~Vinod

2 2

[alsa-devel] [PATCH] hda-analyzer: Make parsing code more robust against future proc file changes
by David Henningsson 07 Feb '13

07 Feb '13

Historically, new additions to the codec proc file has crashed hda-analyzer. With this patch, new additions will instead be ignored. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-analyzer/hda_proc.py | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/hda-analyzer/hda_proc.py b/hda-analyzer/hda_proc.py index 3d544c8..10aa662 100644 --- a/hda-analyzer/hda_proc.py +++ b/hda-analyzer/hda_proc.py @@ -644,9 +644,8 @@ class HDACodecProc(HDACodec, HDABaseProc): if line == 'Invalid AFG subtree': print "Invalid AFG subtree for codec %s?" % self.proc_codec_id return - while line.startswith('Power-Map: ') or \ - line.startswith('Analog Loopback: '): - print 'Sigmatel specific "%s" verb ignored for the moment' % line + while not line.startswith('Node '): + print 'Vendor specific "%s" verb ignored at the moment' % line idx += 1 line = lines[idx].strip() node = None @@ -728,7 +727,7 @@ class HDACodecProc(HDACodec, HDABaseProc): elif line.startswith(' Volume-Knob: '): node.add_volknob(line[15:]) else: - self.wrongfile(line) + print 'Node line "%s" ignored at the moment' % line idx += 1 if idx < len(lines): line = lines[idx] -- 1.7.9.5

1 0

[alsa-devel] [PATCH] hda-emu: Add sanify check of initial pinctl
by David Henningsson 07 Feb '13

07 Feb '13

The pinctl in alsa-info is not the initial pinctl but the current, so sanify the current pinctl before accepting it as the initial pinctl. (The initial pinctl is used during pm test.) Several false positives in the test tool is removed with this improvement. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 64 +++++++++++++++++++++++++++++++++++---------------- hda-parse.c | 2 ++ include/hda-types.h | 4 ++++ 3 files changed, 50 insertions(+), 20 deletions(-) diff --git a/hda-int.c b/hda-int.c index c25af56..fbd3cee 100644 --- a/hda-int.c +++ b/hda-int.c @@ -400,29 +400,37 @@ static int get_channel_streamid(struct xhda_codec *codec, return node->streamid; } -static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, - unsigned int cmd) +void hda_verify_pin_ctl(struct xhda_node *node, int log, + unsigned int *sanified_pinctl) { - if (!node) - return 0; - node->pinctl = cmd & 0xff; + int pinctl = node->pinctl; /* sanity checks */ if ((node->pinctl & AC_PINCTL_OUT_EN) && - !(node->pincap & AC_PINCAP_OUT)) - hda_log(HDA_LOG_ERR, - "setting OUT_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_OUT)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting OUT_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_OUT_EN; + } if ((node->pinctl & AC_PINCTL_HP_EN) && - !(node->pincap & AC_PINCAP_HP_DRV)) - hda_log(HDA_LOG_ERR, - "setting HP_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_HP_DRV)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting HP_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_HP_EN; + } if ((node->pinctl & AC_PINCTL_IN_EN) && - !(node->pincap & AC_PINCAP_IN)) - hda_log(HDA_LOG_ERR, - "setting IN_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_IN)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting IN_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_IN_EN; + } + if (node->pinctl & AC_PINCTL_IN_EN) { unsigned int cap_set, cap_check; const char *vref; @@ -455,11 +463,27 @@ static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, if (!cap_set) cap_set = AC_PINCAP_VREF_HIZ; if (cap_check && !(cap_set & cap_check)) { - hda_log(HDA_LOG_ERR, - "setting VREF %s to pin 0x%x without caps 0x%x\n", - vref, node->nid, node->pincap); + if (log) + hda_log(HDA_LOG_ERR, + "setting VREF %s to pin 0x%x without caps 0x%x\n", + vref, node->nid, node->pincap); + pinctl &= ~AC_PINCTL_VREFEN; } } + + if (sanified_pinctl) + *sanified_pinctl = pinctl; +} + +static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, + unsigned int cmd) +{ + if (!node) + return 0; + node->pinctl = cmd & 0xff; + + hda_verify_pin_ctl(node, 1, NULL); + return 0; } diff --git a/hda-parse.c b/hda-parse.c index c75b03b..d38bc04 100644 --- a/hda-parse.c +++ b/hda-parse.c @@ -240,6 +240,8 @@ static int parse_node_items(const char *buf) &node->orig_amp_out_vals); } else if ((p = strmatch(head, "Pin-ctls: "))) { node->pinctl = strtoul(p, NULL, 0); + if (node->pincap) + hda_verify_pin_ctl(node, 0, &node->pinctl); node->orig_pinctl = node->pinctl; } else if ((p = strmatch(head, "Pincap "))) { const char *end; diff --git a/include/hda-types.h b/include/hda-types.h index 3754bca..68cfaa8 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -146,6 +146,10 @@ void hda_free_route_lists(struct xhda_route_list *list); void hda_show_routes(int nid, unsigned int flags); +void hda_verify_pin_ctl(struct xhda_node *node, int log, + unsigned int *sanified_pinctl); + + void *xalloc(size_t size); #endif /* __HDA_TYPES_H */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: compress: Only mute playback streams
by Mark Brown 07 Feb '13

07 Feb '13

Otherwise capture activity on a compressed DAI would mute any playback on the same DAI. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index c81aeec..35726cb 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,13 +116,12 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; + snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } - snd_soc_dai_digital_mute(codec_dai, 1); - cpu_dai->active--; codec_dai->active--; codec->active--; @@ -179,10 +178,16 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cmd == SNDRV_PCM_TRIGGER_START) - snd_soc_dai_digital_mute(codec_dai, 0); - else if (cmd == SNDRV_PCM_TRIGGER_STOP) - snd_soc_dai_digital_mute(codec_dai, 1); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1); + break; + } + } out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: core: Allow digital mute for capture
by Mark Brown 07 Feb '13

07 Feb '13

Help avoid noise from the power up of the capture path propagating through into the start of the recording (especially noise caused by the ramp of microphone biases) by keeping the capture muted until after we've finished powering things up with DAPM in the same manner we do for playback. This allows us to take advantage of soft mute support in the hardware more effectively and is more consistent. The core code using the existing digital mute operation is updated to take advantage of this. Some additional cases in the soc-pcm code and suspend will need separate handling but these are less practically relevant than the main runtime stream start/stop case. Rather than refactor the digital mute function in every single driver a new operation is added for drivers taking advantage of this functionality, the old operation should be phased out over time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 3 ++- sound/soc/soc-compress.c | 19 +++++++++---------- sound/soc/soc-core.c | 12 ++++++++++-- sound/soc/soc-dapm.c | 6 ++++-- sound/soc/soc-pcm.c | 7 +++---- 5 files changed, 28 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 8bcdd71..6afdaf6 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction); struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 35726cb..b5b3db7 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,12 +116,13 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; - snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + cpu_dai->active--; codec_dai->active--; codec->active--; @@ -178,15 +179,13 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - snd_soc_dai_digital_mute(codec_dai, 0); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_soc_dai_digital_mute(codec_dai, 1); - break; - } + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0, cstream->direction); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + break; } out: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcee552..1283d86 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3547,12 +3547,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); * snd_soc_dai_digital_mute - configure DAI system or master clock. * @dai: DAI * @mute: mute enable + * @direction: stream to mute * * Mutes the DAI DAC. */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) { - if (dai->driver && dai->driver->ops->digital_mute) + if (!dai->driver) + return -ENOTSUPP; + + if (dai->driver->ops->mute_stream) + return dai->driver->ops->mute_stream(dai, mute, direction); + else if (direction == SNDRV_PCM_STREAM_PLAYBACK && + dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else return -ENOTSUPP; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 17dba18..1dc5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3255,14 +3255,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0); + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); ret = 0; break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1); + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); ret = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 786c2e7..a8d72a9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -383,8 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -488,7 +487,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -586,7 +585,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) -- 1.7.10.4

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Re: [alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Mark Brown 07 Feb '13

07 Feb '13

On Wed, Feb 06, 2013 at 11:02:33AM -0800, Jerry Wong wrote: > From: Jerry Wong <jerry.wong(a)maximintegrated.com> > > This patch removes the existing max98090 driver prior to installing a more > complete one. Applied both, thanks.

1 0

Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Takashi Iwai 07 Feb '13

07 Feb '13

At Thu, 7 Feb 2013 11:37:38 +0000, Richard Fitzgerald wrote: > > > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > > > 2) Send SNDRV_COMPRESS_DRAIN > > The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which > > expects the decoder to completely drain its buffers and come to complete halt. > > This would also mean the framework will treat a drained stream as stopped and > > needs a new start. Certainly we dont want that in this case. So we can't use > > SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that > > would overtly complicate this. If we are not doing proper drian lets not > > call it that. > > Ok, so let's keep the partial drain but split out the NEXT_TRACK hint so we keep > the two operations separate. It's clearer to understand. NEXT_TRACK tells DSP to > prepare for data for new track, PARTIAL_DRAIN asks DSP when it has played all > data of current track. I don't think that application _must_ drain when doing > gapless, only if it cares to know when DSP reaches the join between tracks. > Possible problem - application calls NEXT_TRACK and PARTIAL_DRAIN when DSP has > already reached end of current track, so what does PARTIAL_DRAIN mean in this > case? Which track is it draining? Maybe define that it always refers to the > track _before_ the most recent NEXT_TRACK call? > > Also can we make metadata a key-pair list as Takashi suggested, with unlimited > length. Something like > > struct snd_compr_metadata_pair { > enum snd_compr_metadata_key key; > u32 value; > }; > > struct snd_compr_metadata { > int count; /* number of actual entries in following array */ > struct snd_compr_metadata_pair *pairs; > }; > > Should we union the u32 value with a u8* buf to allow for any case where the > value needs to be a buffer of binary data? Oh no, don't do it. Never. In general, - Never put types that depending on the architecture in struct passed to ioctls (e.g. long, pointer) - Never use bit fields for ABI - If 64bit value is used, think of alignment; safer to add packed attribute So, in general, use only u32, s16, or whatever, the types with explicit size. Takashi

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Re: [alsa-devel] mmotm 2013-02-06-17-13 uploaded (alsa)
by Takashi Iwai 07 Feb '13

07 Feb '13

At Wed, 06 Feb 2013 21:59:37 -0800, Randy Dunlap wrote: > > On 02/06/13 17:14, akpm(a)linux-foundation.org wrote: > > The mm-of-the-moment snapshot 2013-02-06-17-13 has been uploaded to > > > > http://www.ozlabs.org/~akpm/mmotm/ > > > > mmotm-readme.txt says > > > > README for mm-of-the-moment: > > > > http://www.ozlabs.org/~akpm/mmotm/ > > > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > > more than once a week. > > > > on i386: (from linux-next.patch) > > > CC [M] sound/pci/hda/patch_via.o > sound/pci/hda/patch_via.c: In function 'is_aa_path_mute': > sound/pci/hda/patch_via.c:360:27: error: 'struct hda_gen_spec' has no member named 'num_loopbacks' > sound/pci/hda/patch_via.c:361:17: error: 'struct hda_gen_spec' has no member named 'loopback_list' > make[4]: *** [sound/pci/hda/patch_via.o] Error 1 > > > > Full randconfig file is attached. Oh, it's again an unsatisfied CONFIG_PM ifdef. Fixed now. Thanks! Takashi

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[alsa-devel] [patch] ALSA: ice1712: fix boundary check in snd_wm8766_write()
by Dan Carpenter 07 Feb '13

07 Feb '13

The wm->regs[] array has WM8766_REG_COUNT (16) elements not WM8766_REG_RESET (31). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is a static checker thing, I can't test it. diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 8072ade..e473f8a 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -31,7 +31,7 @@ static void snd_wm8766_write(struct snd_wm8766 *wm, u16 addr, u16 data) { - if (addr < WM8766_REG_RESET) + if (addr < WM8766_REG_COUNT) wm->regs[addr] = data; wm->ops.write(wm, addr, data); }

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fix debug logging level for FLLs and AIFs
by Mark Brown 06 Feb '13

06 Feb '13

Use _dbg for debug messages. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ba3bd03..5e77a60 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -57,14 +57,14 @@ #define arizona_fll_warn(_fll, fmt, ...) \ dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_dbg(_fll, fmt, ...) \ - dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_aif_err(_dai, fmt, ...) \ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_warn(_dai, fmt, ...) \ dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_dbg(_dai, fmt, ...) \ - dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, -- 1.7.10.4

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[alsa-devel] PandaBoard ES Audio Problems
by Brent Weatherall 06 Feb '13

06 Feb '13

Hello. While developing with OpenMax and ALSA on the PandaBoard ES (OMAP 4460 armhf), I ran into a strange problem. At a high level, when configuring the built in audio device (3.5 mm jack), the device accepts ANY rate requested (via snd_pcm_set_rate_near). it accepts unsupported rates without alteration. If the rate is actually not supported, then 'snd_pcm_hw_params' fails with error '-22'. After this happens, it is not possible to reconfigure the rate without closing the device and starting over to completely re-configure from the start (calling snd_pcm_open, etc...). It is not possible to simply try another rate. Technical details below: When 'snd_pcm_set_rate_near' is called, during the min and max rate probing, the code reaches interval_inline.h:71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) During this call, line '74 return i->min;' executes. This function ALWAYS returns the same rate value requested in 'snd_pcm_set_rate_near' as the minimum possible rate. If I request a rate near 44100, the minimum rates comes back as 44100. If I request a rate near 8000, the minimum is returned as 8000 and so forth. To recap, the 'snd_pcm_set_rate_near' call ALWAYS succeeds without altering the rate passed to it the first time. The only way to find out if the rate is actually supported is to call 'snd_pcm_hw_params' and then see if that fails with err == -22 (invalid argument). Furthermore, once the failure occurs, another rate value cannot be tried unless the PCM device is closed and the entire configuration process is repeated. After the failure, subsequent calls to 'snd_pcm_set_rate_near' with ANY different value always get set back to the original bad value. Example, I request 44100 and it fails. So I try 8000 and it returns 44100 (and of course fails again if I try 'snd_pcm_hw_params'). I am not sure yet, but is the value at 'interval_inline.h:74 return i->min;' determined directly by the hardware? It seems like it is, but I don't know the code well enough yet to make that assertion. I also haven't yet examined the 'snd_interval_max' command, but my gut tells me it is behaving the same way, since after a bad rate is set, any higher or lower rate setting attempts ALWAYS come back as the original rate. How would i determine if it is hardware or maybe something bad with the hardware.software interface? I checked the PCM state value and it was correct the entire time. Here are short snippets of two gdb sessions where I found the issue and below that is test program output showing the state values as I try to configure two different rates: Code snippet in ALSA lib file interval_inline.h: 71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) 72 { 73 assert(!snd_interval_empty(i)); 74 return i->min; 75 } Run #1 with 44100 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $6 = 44100 Run #2 with 8000 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $8 = 8000 Here is a log of the high level test program: snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_channels succeeded. cannot set buffer time (Invalid argument) cannot set buffer period time (Invalid argument) Before commit, state is: SND_PCM_STATE_OPEN cannot set parameters (Invalid argument) After failure, state is: SND_PCM_STATE_OPEN Trying to re-configure rate param. Before calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN cannot drop (Input/output error) After calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN Trying rate 8000. snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_buffer_time_near succeeded. Buffer size actually set to 22050. cannot set buffer period time (Invalid argument) cannot set parameters (Invalid argument) cannot prepare audio interface for use (Input/output error) Rate 44100 supported: FAIL Please let me know your thoughts on this and if there is a good way to prove if this is a hardware error or a software error. I am running 'Linux version 3.4.0-1487-omap4' and using the standard repositories with apt-get to install ALSA. I used 'apt-get source' to obtain 'alsa-lib-1.0.25' source files and I am running 'libasound2-dbg' to enable GDB debugging. Regards, Brent

6 17

[alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 06 Feb '13

06 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Documentation/sound/alsa/compress_offload.txt | 22 +++++++++++ include/sound/compress_driver.h | 2 + include/uapi/sound/compress_offload.h | 18 ++++++++- sound/core/compress_offload.c | 51 +++++++++++++++++++++++++ 4 files changed, 92 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..071c4b6 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,28 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +The decoder needs to know the encoder delay and encoder padding. So we need to +pass this to DSP. Also DSP and userspace needs to switch from one track to +another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..42d6d7c 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -110,6 +110,8 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..65ca2f1 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,18 @@ struct snd_compr_codec_caps { }; /** + * struct snd_compr_metadata: compressed stream metadata + * @encoder_delay: no of samples inserted by the encoder at the beginning + * of the track + * @encoder_padding: no of samples appended by the encoder at the end + * of the track + */ +struct snd_compr_metadata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +157,8 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +166,12 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x35) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 8 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2928971 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,37 @@ out: return retval; } +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + pr_debug("metadata encoder delay=%x encoder padding=%x\n", + metadata->encoder_delay, metadata->encoder_padding); + + retval = stream->ops->set_metadata(stream, metadata); + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +625,20 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + else + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +668,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +692,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

4 13

[alsa-devel] ASoC updates for v3.8
by Mark Brown 06 Feb '13

06 Feb '13

The following changes since commit 949db153b6466c6f7cad5a427ecea94985927311: Linux 3.8-rc5 (2013-01-25 11:57:28 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.8-rc6 for you to fetch changes up to 5074e8eca643dd19e5397e638c7e9ca0ba9b4e24: Merge remote-tracking branch 'asoc/fix/wm2200' into tmp (2013-02-06 10:54:35 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.8-rc7 I've been a bit slack sending the pull requests for this so there's rather more driver specific updates here than is desirable but everything here is nicely localised so hopefully we're OK. ---------------------------------------------------------------- Alexey Galakhov (1): ASoC: samsung: Fix compilation error on S3C2440 Benoît Thébaudeau (1): ASoC: tlv320aic3x: Remove mono support Charles Keepax (1): ASoC: wm_adsp: Release firmware on error Chris Rattray (1): ASoC: wm2200: correct IN2L and IN3L digital mute Dan Carpenter (1): ASoC: dwc: fix support for more than two channels Lee Jones (1): ASoC: Ux500: Fix build error pertaining to missing include file Mark Brown (13): ASoC: wm_adsp: Ensure ADSP2 DMAs are quiesced when DSP is halted ASoC: wm2000: Lock state machine updates Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/davinci' into tmp Merge remote-tracking branch 'asoc/fix/dwc' into tmp Merge remote-tracking branch 'asoc/fix/fsl' into tmp Merge remote-tracking branch 'asoc/fix/lrg' into tmp Merge remote-tracking branch 'asoc/fix/samsung' into tmp Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into tmp Merge remote-tracking branch 'asoc/fix/ux500' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Michal Bachraty (1): ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback Ryo Tsutsui (1): ASoC: arizona: Fixed a bug in FLL fractional calculation Shawn Guo (2): Revert "ASoC: fsl: fix multiple definition of init_module" ASoC: fsl: fix snd-soc-imx-pcm module build MAINTAINERS | 2 +- sound/soc/codecs/arizona.c | 16 ++++++++++++++-- sound/soc/codecs/tlv320aic3x.c | 4 ++-- sound/soc/codecs/wm2000.c | 31 ++++++++++++++++++++++++++++--- sound/soc/codecs/wm2200.c | 7 ++----- sound/soc/codecs/wm5102.c | 3 +-- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm_adsp.c | 20 ++++++++++++++------ sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/dwc/designware_i2s.c | 4 ++++ sound/soc/fsl/imx-pcm-dma.c | 21 +-------------------- sound/soc/fsl/imx-pcm-fiq.c | 22 +--------------------- sound/soc/fsl/imx-pcm.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.h | 18 ++++++++++++++++++ sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/soc-dapm.c | 12 ++++++++++-- sound/soc/ux500/mop500.c | 2 +- 17 files changed, 132 insertions(+), 69 deletions(-)

2 5

[alsa-devel] hda_analyzer failed
by Raymond Yau 06 Feb '13

06 Feb '13

>> >> > File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile >> > raise ValueError, "wrong proc file format (%s)" % msg >> > ValueError: wrong proc file format (integer decode 'State of AFG node >> 0x01:' ('Node ')) >> > >> > Any ideas? >> >> This patch add lines which make the parser of hda_proc.py fail >> >> http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=f4… >> > > Sigh, this is not the first time new stuff in the proc file crashes hda_analyzer. > > Jaroslav, can you please fix hda_analyzer so that it ignores new stuff in the proc file instead of crashing, so we don't have this problem over and over again? > I don`t understand why the parser have to ignore new stuff BTW, why hda_proc query the connection list of vendor defined widget ? http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob;f=codecs… Node 0x23 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: -22

3 3

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 06 Feb '13

06 Feb '13

At Wed, 6 Feb 2013 08:47:41 +0800, Daniel J Blueman wrote: > > On 6 February 2013 01:40, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 05 Feb 2013 17:34:15 +0100, > > Takashi Iwai wrote: > >> > >> At Wed, 6 Feb 2013 00:29:54 +0800, > >> Daniel J Blueman wrote: > >> > > >> > On 6 February 2013 00:16, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > > At Wed, 6 Feb 2013 00:10:30 +0800, > >> > > Daniel J Blueman wrote: > >> > >> > >> > >> Hi Takashi, > >> > >> > >> > >> The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > >> > >> arrangement on my Macbook Pro 10,1 - only the higher-frequency > >> > >> speakers work despite the front and rear channels being exposed in the > >> > >> mixer. > >> > > > >> > > Interesting. So you have a machine with 4.0 built-in speaker instead > >> > > of 2.1? Then we need to add a device-specific flag for it. Currently > >> > > the driver assumes 2.1 system blindly because majority of machines > >> > > have that. > >> > > > >> > > FWIW, the codec parser code has been totally rewritten for 3.9, so any > >> > > patch to 3.8 won't be applied to 3.9 (and vice versa)... > >> > > > >> > > Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to > >> > > cook it for 3.9 (and maybe backport to 3.8). > >> > > >> > Here's the output from the current alsa-info.sh on 3.8-rc6 with the > >> > two cited patches reverted [1]; let me know if you'd like 3.8-rc6 > >> > pure. > >> > >> Thanks. > >> > >> > The sound is pretty impressive for a laptop when the low-frequecy > >> > speakers are enabled. > >> > >> Which program are you using for testing the surrounds? > >> I'm interested in it because the commit you reverted is basically > >> providing only an additional information for the channel map, and it > >> doesn't change anything else. It implies that some applications are > >> really referring to the chmap info. > > > > Or, it might be that the mixer value is simply not set correct. > > > > To be sure, could you try again 3.8-rc6 without reversing patches, > > adjust "Speaker" and "Bass Speaker" volumes properly, and retest? > > If it still doesn't work, please take alsa-info.sh snapshot at this > > state for comparing with the previous result. > > My apologies! We do now have "Bass Speaker" which affects both bass > speakers. It was always being restored to level 0 and works when set > up. OK, good to know. > There is a "Subwoofer" slider in addition to "Balance" and "Fade" in > the GNOME mixer UI which is greyed out; presumably this is intended as > the same mixer control? I have no idea about GNOME stuff... David, do you know of this? Takashi

2 1

[alsa-devel] [PATCH 00/15] ARM build regressions in v3.8
by Arnd Bergmann 06 Feb '13

06 Feb '13

I know this comes late, but we have a number of broken configurations in ARM in v3.8 that were still building in v3.7, and I'd like to get them all fixed in the final 3.8 release. It would be nice if the respective maintainers could have a look at these patches and apply them directly when they are happy with them. The first patch in the series is strictly speaking not a build error but just a warning, but it is a particularly annoying one that came in through the latest binutils release rather than a kernel change. The same binutils update also broke the samsung and w90x900 platforms. A few of the other changes are the result of the imx multiplatform conversion. I'm not really fixing those here, just picking up the pieces. It would be much nicer if we could actually get those drivers to work again with CONFIG_MULTIPLATFORM enabled rather than just disabling them, but it may be much too late for that. At least the drivers don't seem to be too essential, as they are only built in allyesconfig but not in any of the defconfigs. Arnd Arnd Bergmann (15): ARM: compressed/head.S: work around new binutils warning ARM: mvebu: build coherency_ll.S for arch=armv7-a ARM: samsung: fix assembly syntax for new gas ARM: w90x900: fix legacy assembly syntax ASoC: fsl: fiq and dma cannot both be modules clk: export __clk_get_name drm/exynos: don't include plat/gpio-cfg.h drm/exynos: fimd and ipp are broken on multiplatform media: coda: don't build on multiplatform mfd/vexpress: export vexpress_config_func_{put,get} mtd: davinci_nand: fix OF support USB: gadget/freescale: disable non-multiplatform drivers USB: ehci: make orion and mxc bus glues coexist samples/seccomp: be less stupid about cross compiling staging/omapdrm: don't build on multiplatform arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/compressed/head.S | 12 ++++++++++++ arch/arm/mach-mvebu/coherency_ll.S | 1 + arch/arm/mach-s3c24xx/include/mach/debug-macro.S | 12 ++++++------ arch/arm/mach-s3c24xx/include/mach/entry-macro.S | 4 ++-- arch/arm/mach-s3c24xx/pm-h1940.S | 2 +- arch/arm/mach-s3c24xx/sleep-s3c2410.S | 12 ++++++------ arch/arm/mach-s3c24xx/sleep-s3c2412.S | 12 ++++++------ arch/arm/mach-w90x900/include/mach/entry-macro.S | 4 ++-- arch/arm/plat-samsung/include/plat/debug-macro.S | 18 +++++++++--------- drivers/clk/clk.c | 1 + drivers/gpu/drm/exynos/Kconfig | 4 ++-- drivers/gpu/drm/exynos/exynos_hdmi.c | 1 - drivers/media/platform/Kconfig | 2 +- drivers/mfd/vexpress-config.c | 3 ++- drivers/mtd/nand/davinci_nand.c | 2 +- drivers/staging/omapdrm/Kconfig | 2 +- drivers/usb/gadget/Kconfig | 3 ++- drivers/usb/host/ehci-hcd.c | 16 +++++++++++++++- samples/seccomp/Makefile | 2 ++ sound/soc/fsl/Kconfig | 3 +++ 21 files changed, 76 insertions(+), 42 deletions(-) -- 1.7.10.4 Cc: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: James Morris <james.l.morris(a)oracle.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> Cc: Mike Turquette <mturquette(a)linaro.org> Cc: Rob Clark <rob(a)ti.com> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Shawn Guo <shawn.guo(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-media(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org

3 8

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 05 Feb '13

05 Feb '13

At Wed, 6 Feb 2013 00:29:54 +0800, Daniel J Blueman wrote: > > On 6 February 2013 00:16, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 6 Feb 2013 00:10:30 +0800, > > Daniel J Blueman wrote: > >> > >> Hi Takashi, > >> > >> The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > >> arrangement on my Macbook Pro 10,1 - only the higher-frequency > >> speakers work despite the front and rear channels being exposed in the > >> mixer. > > > > Interesting. So you have a machine with 4.0 built-in speaker instead > > of 2.1? Then we need to add a device-specific flag for it. Currently > > the driver assumes 2.1 system blindly because majority of machines > > have that. > > > > FWIW, the codec parser code has been totally rewritten for 3.9, so any > > patch to 3.8 won't be applied to 3.9 (and vice versa)... > > > > Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to > > cook it for 3.9 (and maybe backport to 3.8). > > Here's the output from the current alsa-info.sh on 3.8-rc6 with the > two cited patches reverted [1]; let me know if you'd like 3.8-rc6 > pure. Thanks. > The sound is pretty impressive for a laptop when the low-frequecy > speakers are enabled. Which program are you using for testing the surrounds? I'm interested in it because the commit you reverted is basically providing only an additional information for the channel map, and it doesn't change anything else. It implies that some applications are really referring to the chmap info. Takashi

1 1

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 05 Feb '13

05 Feb '13

At Wed, 6 Feb 2013 00:10:30 +0800, Daniel J Blueman wrote: > > Hi Takashi, > > The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > arrangement on my Macbook Pro 10,1 - only the higher-frequency > speakers work despite the front and rear channels being exposed in the > mixer. Interesting. So you have a machine with 4.0 built-in speaker instead of 2.1? Then we need to add a device-specific flag for it. Currently the driver assumes 2.1 system blindly because majority of machines have that. FWIW, the codec parser code has been totally rewritten for 3.9, so any patch to 3.8 won't be applied to 3.9 (and vice versa)... Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to cook it for 3.9 (and maybe backport to 3.8). Takashi > > Reverting f37bc7 [1] restored the correct behaviour ([2] was reverted > to compile, but the problem still occurs with only [2] reverted). > > How can I help to debug this and find the right approach? > > Thanks, > Daniel > > --- [1] > > commit f37bc7a88d374448a1f4bba9267d308606d78bf2 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Nov 8 15:59:23 2012 +0100 > > ALSA: hda - Give standard "Bass Speaker" mixer for 2.1 speakers > > When two built-in speakers are found on the machine, we can suppose > it's rather a 2.1 speaker system with a bass output instead of > front/surround channels. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- [2] > > commit ee81abb623cb5e03c182d16871bb4fb34fdc9b4f > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Nov 8 17:12:10 2012 +0100 > > ALSA: hda - Apply a proper chmap for built-in 2.1 speakers > > When 2.1 speakers are detected, use the corresponding channel map > instead of the standard map with front+rear surrounds. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > -- > Daniel J Blueman >

1 1

[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 05 Feb '13

05 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..bc7ab1a 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 1, + WM2200_MBIAS_LVL_1V8 = 2, + WM2200_MBIAS_LVL_1V9 = 3, + WM2200_MBIAS_LVL_2V0 = 4, + WM2200_MBIAS_LVL_2V2 = 5, + WM2200_MBIAS_LVL_2V4 = 6, + WM2200_MBIAS_LVL_2V5 = 7, + WM2200_MBIAS_LVL_2V6 = 8, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..bbd8383 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 05 Feb '13

05 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Change-Id: I65da7c50baf7134d20e0bfeecc601d82828a8de1 --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..bc7ab1a 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 1, + WM2200_MBIAS_LVL_1V8 = 2, + WM2200_MBIAS_LVL_1V9 = 3, + WM2200_MBIAS_LVL_2V0 = 4, + WM2200_MBIAS_LVL_2V2 = 5, + WM2200_MBIAS_LVL_2V4 = 6, + WM2200_MBIAS_LVL_2V5 = 7, + WM2200_MBIAS_LVL_2V6 = 8, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..bbd8383 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

2 1

[alsa-devel] alsa wiki spam
by Eliot Blennerhassett 05 Feb '13

05 Feb '13

Hundreds of spam bot registrations dominate the recent changes: http://www.alsa-project.org/main/index.php?title=Special:RecentChanges&limi… no sign so far of spam content, but it is a bit of a worry.

2 1

[alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 05 Feb '13

05 Feb '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch removes the existing max98090 driver prior to installing a more complete one. --- The original author is okay to replace his recently submitted driver with this one. "... Would you mind if we began this now by replacing the existing driver with the one attached? ..." - Jerry "... But of course I agree to replace it to your code. Please send it to upstream... " - Morimoto-san sound/soc/codecs/Kconfig | 4 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/max98090.c | 577 ------------------------------------------- 3 files changed, 0 insertions(+), 583 deletions(-) delete mode 100644 sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..0e368d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -268,9 +267,6 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate -config SND_SOC_MAX98090 - tristate - config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..aa56312 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,7 +34,6 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o -snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -158,7 +157,6 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o -obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c deleted file mode 100644 index c9772ca..0000000 --- a/sound/soc/codecs/max98090.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * max98090.c -- MAX98090 ALSA SoC Audio driver - * based on Rev0p8 datasheet - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * Based on - * - * max98095.c - * Copyright 2011 Maxim Integrated Products - * - * https://github.com/hardkernel/linux/commit/\ - * 3417d7166b17113b3b33b0a337c74d1c7cc313df#sound/soc/codecs/max98090.c - * Copyright 2011 Maxim Integrated Products - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/i2c.h> -#include <linux/module.h> -#include <linux/regmap.h> -#include <sound/soc.h> -#include <sound/tlv.h> - -/* - * - * MAX98090 Registers Definition - * - */ - -/* RESET / STATUS / INTERRUPT REGISTERS */ -#define MAX98090_0x00_SW_RESET 0x00 -#define MAX98090_0x01_INT_STS 0x01 -#define MAX98090_0x02_JACK_STS 0x02 -#define MAX98090_0x03_INT_MASK 0x03 - -/* QUICK SETUP REGISTERS */ -#define MAX98090_0x04_SYS_CLK 0x04 -#define MAX98090_0x05_SAMPLE_RATE 0x05 -#define MAX98090_0x06_DAI_IF 0x06 -#define MAX98090_0x07_DAC_PATH 0x07 -#define MAX98090_0x08_MIC_TO_ADC 0x08 -#define MAX98090_0x09_LINE_TO_ADC 0x09 -#define MAX98090_0x0A_ANALOG_MIC_LOOP 0x0A -#define MAX98090_0x0B_ANALOG_LINE_LOOP 0x0B - -/* ANALOG INPUT CONFIGURATION REGISTERS */ -#define MAX98090_0x0D_INPUT_CONFIG 0x0D -#define MAX98090_0x0E_LINE_IN_LVL 0x0E -#define MAX98090_0x0F_LINI_IN_CFG 0x0F -#define MAX98090_0x10_MIC1_IN_LVL 0x10 -#define MAX98090_0x11_MIC2_IN_LVL 0x11 - -/* MICROPHONE CONFIGURATION REGISTERS */ -#define MAX98090_0x12_MIC_BIAS_VOL 0x12 -#define MAX98090_0x13_DIGITAL_MIC_CFG 0x13 -#define MAX98090_0x14_DIGITAL_MIC_MODE 0x14 - -/* ADC PATH AND CONFIGURATION REGISTERS */ -#define MAX98090_0x15_L_ADC_MIX 0x15 -#define MAX98090_0x16_R_ADC_MIX 0x16 -#define MAX98090_0x17_L_ADC_LVL 0x17 -#define MAX98090_0x18_R_ADC_LVL 0x18 -#define MAX98090_0x19_ADC_BIQUAD_LVL 0x19 -#define MAX98090_0x1A_ADC_SIDETONE 0x1A - -/* CLOCK CONFIGURATION REGISTERS */ -#define MAX98090_0x1B_SYS_CLK 0x1B -#define MAX98090_0x1C_CLK_MODE 0x1C -#define MAX98090_0x1D_ANY_CLK1 0x1D -#define MAX98090_0x1E_ANY_CLK2 0x1E -#define MAX98090_0x1F_ANY_CLK3 0x1F -#define MAX98090_0x20_ANY_CLK4 0x20 -#define MAX98090_0x21_MASTER_MODE 0x21 - -/* INTERFACE CONTROL REGISTERS */ -#define MAX98090_0x22_DAI_IF_FMT 0x22 -#define MAX98090_0x23_DAI_TDM_FMT1 0x23 -#define MAX98090_0x24_DAI_TDM_FMT2 0x24 -#define MAX98090_0x25_DAI_IO_CFG 0x25 -#define MAX98090_0x26_FILTER_CFG 0x26 -#define MAX98090_0x27_DAI_PLAYBACK_LVL 0x27 -#define MAX98090_0x28_EQ_PLAYBACK_LVL 0x28 - -/* HEADPHONE CONTROL REGISTERS */ -#define MAX98090_0x29_L_HP_MIX 0x29 -#define MAX98090_0x2A_R_HP_MIX 0x2A -#define MAX98090_0x2B_HP_CTR 0x2B -#define MAX98090_0x2C_L_HP_VOL 0x2C -#define MAX98090_0x2D_R_HP_VOL 0x2D - -/* SPEAKER CONFIGURATION REGISTERS */ -#define MAX98090_0x2E_L_SPK_MIX 0x2E -#define MAX98090_0x2F_R_SPK_MIX 0x2F -#define MAX98090_0x30_SPK_CTR 0x30 -#define MAX98090_0x31_L_SPK_VOL 0x31 -#define MAX98090_0x32_R_SPK_VOL 0x32 - -/* ALC CONFIGURATION REGISTERS */ -#define MAX98090_0x33_ALC_TIMING 0x33 -#define MAX98090_0x34_ALC_COMPRESSOR 0x34 -#define MAX98090_0x35_ALC_EXPANDER 0x35 -#define MAX98090_0x36_ALC_GAIN 0x36 - -/* RECEIVER AND LINE_OUTPUT REGISTERS */ -#define MAX98090_0x37_RCV_LOUT_L_MIX 0x37 -#define MAX98090_0x38_RCV_LOUT_L_CNTL 0x38 -#define MAX98090_0x39_RCV_LOUT_L_VOL 0x39 -#define MAX98090_0x3A_LOUT_R_MIX 0x3A -#define MAX98090_0x3B_LOUT_R_CNTL 0x3B -#define MAX98090_0x3C_LOUT_R_VOL 0x3C - -/* JACK DETECT AND ENABLE REGISTERS */ -#define MAX98090_0x3D_JACK_DETECT 0x3D -#define MAX98090_0x3E_IN_ENABLE 0x3E -#define MAX98090_0x3F_OUT_ENABLE 0x3F -#define MAX98090_0x40_LVL_CTR 0x40 -#define MAX98090_0x41_DSP_FILTER_ENABLE 0x41 - -/* BIAS AND POWER MODE CONFIGURATION REGISTERS */ -#define MAX98090_0x42_BIAS_CTR 0x42 -#define MAX98090_0x43_DAC_CTR 0x43 -#define MAX98090_0x44_ADC_CTR 0x44 -#define MAX98090_0x45_DEV_SHUTDOWN 0x45 - -/* REVISION ID REGISTER */ -#define MAX98090_0xFF_REV_ID 0xFF - -#define MAX98090_REG_MAX_CACHED 0x45 -#define MAX98090_REG_END 0xFF - -/* - * - * MAX98090 Registers Bit Fields - * - */ - -/* MAX98090_0x06_DAI_IF */ -#define MAX98090_DAI_IF_MASK 0x3F -#define MAX98090_RJ_M (1 << 5) -#define MAX98090_RJ_S (1 << 4) -#define MAX98090_LJ_M (1 << 3) -#define MAX98090_LJ_S (1 << 2) -#define MAX98090_I2S_M (1 << 1) -#define MAX98090_I2S_S (1 << 0) - -/* MAX98090_0x45_DEV_SHUTDOWN */ -#define MAX98090_SHDNRUN (1 << 7) - -/* codec private data */ -struct max98090_priv { - struct regmap *regmap; -}; - -static const struct reg_default max98090_reg_defaults[] = { - /* RESET / STATUS / INTERRUPT REGISTERS */ - {MAX98090_0x00_SW_RESET, 0x00}, - {MAX98090_0x01_INT_STS, 0x00}, - {MAX98090_0x02_JACK_STS, 0x00}, - {MAX98090_0x03_INT_MASK, 0x04}, - - /* QUICK SETUP REGISTERS */ - {MAX98090_0x04_SYS_CLK, 0x00}, - {MAX98090_0x05_SAMPLE_RATE, 0x00}, - {MAX98090_0x06_DAI_IF, 0x00}, - {MAX98090_0x07_DAC_PATH, 0x00}, - {MAX98090_0x08_MIC_TO_ADC, 0x00}, - {MAX98090_0x09_LINE_TO_ADC, 0x00}, - {MAX98090_0x0A_ANALOG_MIC_LOOP, 0x00}, - {MAX98090_0x0B_ANALOG_LINE_LOOP, 0x00}, - - /* ANALOG INPUT CONFIGURATION REGISTERS */ - {MAX98090_0x0D_INPUT_CONFIG, 0x00}, - {MAX98090_0x0E_LINE_IN_LVL, 0x1B}, - {MAX98090_0x0F_LINI_IN_CFG, 0x00}, - {MAX98090_0x10_MIC1_IN_LVL, 0x11}, - {MAX98090_0x11_MIC2_IN_LVL, 0x11}, - - /* MICROPHONE CONFIGURATION REGISTERS */ - {MAX98090_0x12_MIC_BIAS_VOL, 0x00}, - {MAX98090_0x13_DIGITAL_MIC_CFG, 0x00}, - {MAX98090_0x14_DIGITAL_MIC_MODE, 0x00}, - - /* ADC PATH AND CONFIGURATION REGISTERS */ - {MAX98090_0x15_L_ADC_MIX, 0x00}, - {MAX98090_0x16_R_ADC_MIX, 0x00}, - {MAX98090_0x17_L_ADC_LVL, 0x03}, - {MAX98090_0x18_R_ADC_LVL, 0x03}, - {MAX98090_0x19_ADC_BIQUAD_LVL, 0x00}, - {MAX98090_0x1A_ADC_SIDETONE, 0x00}, - - /* CLOCK CONFIGURATION REGISTERS */ - {MAX98090_0x1B_SYS_CLK, 0x00}, - {MAX98090_0x1C_CLK_MODE, 0x00}, - {MAX98090_0x1D_ANY_CLK1, 0x00}, - {MAX98090_0x1E_ANY_CLK2, 0x00}, - {MAX98090_0x1F_ANY_CLK3, 0x00}, - {MAX98090_0x20_ANY_CLK4, 0x00}, - {MAX98090_0x21_MASTER_MODE, 0x00}, - - /* INTERFACE CONTROL REGISTERS */ - {MAX98090_0x22_DAI_IF_FMT, 0x00}, - {MAX98090_0x23_DAI_TDM_FMT1, 0x00}, - {MAX98090_0x24_DAI_TDM_FMT2, 0x00}, - {MAX98090_0x25_DAI_IO_CFG, 0x00}, - {MAX98090_0x26_FILTER_CFG, 0x80}, - {MAX98090_0x27_DAI_PLAYBACK_LVL, 0x00}, - {MAX98090_0x28_EQ_PLAYBACK_LVL, 0x00}, - - /* HEADPHONE CONTROL REGISTERS */ - {MAX98090_0x29_L_HP_MIX, 0x00}, - {MAX98090_0x2A_R_HP_MIX, 0x00}, - {MAX98090_0x2B_HP_CTR, 0x00}, - {MAX98090_0x2C_L_HP_VOL, 0x1A}, - {MAX98090_0x2D_R_HP_VOL, 0x1A}, - - /* SPEAKER CONFIGURATION REGISTERS */ - {MAX98090_0x2E_L_SPK_MIX, 0x00}, - {MAX98090_0x2F_R_SPK_MIX, 0x00}, - {MAX98090_0x30_SPK_CTR, 0x00}, - {MAX98090_0x31_L_SPK_VOL, 0x2C}, - {MAX98090_0x32_R_SPK_VOL, 0x2C}, - - /* ALC CONFIGURATION REGISTERS */ - {MAX98090_0x33_ALC_TIMING, 0x00}, - {MAX98090_0x34_ALC_COMPRESSOR, 0x00}, - {MAX98090_0x35_ALC_EXPANDER, 0x00}, - {MAX98090_0x36_ALC_GAIN, 0x00}, - - /* RECEIVER AND LINE_OUTPUT REGISTERS */ - {MAX98090_0x37_RCV_LOUT_L_MIX, 0x00}, - {MAX98090_0x38_RCV_LOUT_L_CNTL, 0x00}, - {MAX98090_0x39_RCV_LOUT_L_VOL, 0x15}, - {MAX98090_0x3A_LOUT_R_MIX, 0x00}, - {MAX98090_0x3B_LOUT_R_CNTL, 0x00}, - {MAX98090_0x3C_LOUT_R_VOL, 0x15}, - - /* JACK DETECT AND ENABLE REGISTERS */ - {MAX98090_0x3D_JACK_DETECT, 0x00}, - {MAX98090_0x3E_IN_ENABLE, 0x00}, - {MAX98090_0x3F_OUT_ENABLE, 0x00}, - {MAX98090_0x40_LVL_CTR, 0x00}, - {MAX98090_0x41_DSP_FILTER_ENABLE, 0x00}, - - /* BIAS AND POWER MODE CONFIGURATION REGISTERS */ - {MAX98090_0x42_BIAS_CTR, 0x00}, - {MAX98090_0x43_DAC_CTR, 0x00}, - {MAX98090_0x44_ADC_CTR, 0x06}, - {MAX98090_0x45_DEV_SHUTDOWN, 0x00}, -}; - -static const unsigned int max98090_hp_tlv[] = { - TLV_DB_RANGE_HEAD(5), - 0x0, 0x6, TLV_DB_SCALE_ITEM(-6700, 400, 0), - 0x7, 0xE, TLV_DB_SCALE_ITEM(-4000, 300, 0), - 0xF, 0x15, TLV_DB_SCALE_ITEM(-1700, 200, 0), - 0x16, 0x1B, TLV_DB_SCALE_ITEM(-400, 100, 0), - 0x1C, 0x1F, TLV_DB_SCALE_ITEM(150, 50, 0), -}; - -static struct snd_kcontrol_new max98090_snd_controls[] = { - SOC_DOUBLE_R_TLV("Headphone Volume", MAX98090_0x2C_L_HP_VOL, - MAX98090_0x2D_R_HP_VOL, 0, 31, 0, max98090_hp_tlv), -}; - -/* Left HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x29_L_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x29_L_HP_MIX, 0, 1, 0), -}; - -/* Right HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x2A_R_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x2A_R_HP_MIX, 0, 1, 0), -}; - -static struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - /* Output */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* PGA */ - SND_SOC_DAPM_PGA("HPL Out", MAX98090_0x3F_OUT_ENABLE, 7, 0, NULL, 0), - SND_SOC_DAPM_PGA("HPR Out", MAX98090_0x3F_OUT_ENABLE, 6, 0, NULL, 0), - - /* Mixer */ - SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, - max98090_left_hp_mixer_controls, - ARRAY_SIZE(max98090_left_hp_mixer_controls)), - - SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, - max98090_right_hp_mixer_controls, - ARRAY_SIZE(max98090_right_hp_mixer_controls)), - - /* DAC */ - SND_SOC_DAPM_DAC("DACL", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 1, 0), -}; - -static struct snd_soc_dapm_route max98090_audio_map[] = { - /* Output */ - {"HPL", NULL, "HPL Out"}, - {"HPR", NULL, "HPR Out"}, - - /* PGA */ - {"HPL Out", NULL, "HPL Mixer"}, - {"HPR Out", NULL, "HPR Mixer"}, - - /* Mixer*/ - {"HPL Mixer", "DACR Switch", "DACR"}, - {"HPL Mixer", "DACL Switch", "DACL"}, - - {"HPR Mixer", "DACR Switch", "DACR"}, - {"HPR Mixer", "DACL Switch", "DACL"}, -}; - -static bool max98090_volatile(struct device *dev, unsigned int reg) -{ - if ((reg == MAX98090_0x01_INT_STS) || - (reg == MAX98090_0x02_JACK_STS) || - (reg > MAX98090_REG_MAX_CACHED)) - return true; - - return false; -} - -static int max98090_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - switch (params_rate(params)) { - case 96000: - val = 1 << 5; - break; - case 32000: - val = 1 << 4; - break; - case 48000: - val = 1 << 3; - break; - case 44100: - val = 1 << 2; - break; - case 16000: - val = 1 << 1; - break; - case 8000: - val = 1 << 0; - break; - default: - dev_err(codec->dev, "unsupported rate\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x05_SAMPLE_RATE, 0x03F, val); - - return 0; -} - -static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, 0); - - switch (freq) { - case 26000000: - val = 1 << 7; - break; - case 19200000: - val = 1 << 6; - break; - case 13000000: - val = 1 << 5; - break; - case 12288000: - val = 1 << 4; - break; - case 12000000: - val = 1 << 3; - break; - case 11289600: - val = 1 << 2; - break; - default: - dev_err(codec->dev, "Invalid master clock frequency\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x04_SYS_CLK, 0xFD, val); - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - dev_dbg(dai->dev, "sysclk is %uHz\n", freq); - - return 0; -} - -static int max98090_dai_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct snd_soc_codec *codec = dai->codec; - int is_master; - u8 val; - - /* master/slave mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - is_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - is_master = 0; - break; - default: - dev_err(codec->dev, "unsupported clock\n"); - return -EINVAL; - } - - /* format */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_I2S: - val = (is_master) ? MAX98090_I2S_M : MAX98090_I2S_S; - break; - case SND_SOC_DAIFMT_RIGHT_J: - val = (is_master) ? MAX98090_RJ_M : MAX98090_RJ_S; - break; - case SND_SOC_DAIFMT_LEFT_J: - val = (is_master) ? MAX98090_LJ_M : MAX98090_LJ_S; - break; - default: - dev_err(codec->dev, "unsupported format\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x06_DAI_IF, - MAX98090_DAI_IF_MASK, val); - - return 0; -} - -#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 -#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops max98090_dai_ops = { - .set_sysclk = max98090_dai_set_sysclk, - .set_fmt = max98090_dai_set_fmt, - .hw_params = max98090_dai_hw_params, -}; - -static struct snd_soc_dai_driver max98090_dai = { - .name = "max98090-Hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = MAX98090_RATES, - .formats = MAX98090_FORMATS, - }, - .ops = &max98090_dai_ops, -}; - -static int max98090_probe(struct snd_soc_codec *codec) -{ - struct max98090_priv *priv = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; - int ret; - - codec->control_data = priv->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - /* Device active */ - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - return 0; -} - -static int max98090_remove(struct snd_soc_codec *codec) -{ - return 0; -} - -static struct snd_soc_codec_driver soc_codec_dev_max98090 = { - .probe = max98090_probe, - .remove = max98090_remove, - .controls = max98090_snd_controls, - .num_controls = ARRAY_SIZE(max98090_snd_controls), - .dapm_widgets = max98090_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(max98090_dapm_widgets), - .dapm_routes = max98090_audio_map, - .num_dapm_routes = ARRAY_SIZE(max98090_audio_map), -}; - -static const struct regmap_config max98090_regmap = { - .reg_bits = 8, - .val_bits = 8, - .max_register = MAX98090_REG_END, - .volatile_reg = max98090_volatile, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = max98090_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(max98090_reg_defaults), -}; - -static int max98090_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct max98090_priv *priv; - struct device *dev = &i2c->dev; - unsigned int val; - int ret; - - priv = devm_kzalloc(dev, sizeof(struct max98090_priv), - GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); - if (IS_ERR(priv->regmap)) { - ret = PTR_ERR(priv->regmap); - dev_err(dev, "Failed to init regmap: %d\n", ret); - return ret; - } - - i2c_set_clientdata(i2c, priv); - - ret = regmap_read(priv->regmap, MAX98090_0xFF_REV_ID, &val); - if (ret < 0) { - dev_err(dev, "Failed to read device revision: %d\n", ret); - return ret; - } - dev_info(dev, "revision 0x%02x\n", val); - - ret = snd_soc_register_codec(dev, - &soc_codec_dev_max98090, - &max98090_dai, 1); - - return ret; -} - -static int max98090_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id max98090_i2c_id[] = { - { "max98090", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); - -static struct i2c_driver max98090_i2c_driver = { - .driver = { - .name = "max98090", - .owner = THIS_MODULE, - }, - .probe = max98090_i2c_probe, - .remove = max98090_i2c_remove, - .id_table = max98090_i2c_id, -}; -module_i2c_driver(max98090_i2c_driver); - -MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); -MODULE_AUTHOR("Peter Hsiang, Kuninori Morimoto"); -MODULE_LICENSE("GPL"); -- 1.7.0.4

2 7

[alsa-devel] [PATCH] ASoC: soc-compress: Add support for not memory mapped DSPs
by Charles Keepax 05 Feb '13

05 Feb '13

The ASoC compressed API did not implement the copy callback in its compressed ops which is required for DSPs that are not memory mapped. This patch adds a second set of compressed ops which does implement the copy callback and uses that when copy is defined in the platform compressed ops, ie. when the DSP is not memory mapped. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 36 +++++++++++++++++++++++++++++++++++- 1 files changed, 35 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..f9fb112 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,6 +302,22 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int soc_compr_copy(struct snd_compr_stream *cstream, + const char __user *buf, size_t count) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + ret = platform->driver->compr_ops->copy(cstream, buf, count); + + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, @@ -315,10 +331,25 @@ static struct snd_compr_ops soc_compr_ops = { .get_codec_caps = soc_compr_get_codec_caps }; +/* ASoC Compress operations for non memory mapped DSPs */ +static struct snd_compr_ops soc_compr_nomap_ops = { + .open = soc_compr_open, + .free = soc_compr_free, + .set_params = soc_compr_set_params, + .get_params = soc_compr_get_params, + .trigger = soc_compr_trigger, + .pointer = soc_compr_pointer, + .copy = soc_compr_copy, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + /* create a new compress */ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_compr *compr; @@ -335,7 +366,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -ENOMEM; } - compr->ops = &soc_compr_ops; + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + compr->ops = &soc_compr_nomap_ops; + else + compr->ops = &soc_compr_ops; mutex_init(&compr->lock); ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { -- 1.7.2.5

3 11

[alsa-devel] [PATCH] ASoC: omap-pcm: No need to set constraint at open time
by Peter Ujfalusi 05 Feb '13

05 Feb '13

The same constraint is going to be set in the snd_dmaengine_pcm_open() function, so there is no need to set it here as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-pcm.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 47bdbd4..c722c2e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -174,23 +174,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) static int omap_pcm_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_pcm_dma_data *dma_data; - int ret; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); - /* Ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ret = snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, - &dma_data->dma_req); - return ret; + + return snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, + &dma_data->dma_req); } static int omap_pcm_close(struct snd_pcm_substream *substream) -- 1.8.1.1

3 2

[alsa-devel] Join my network on LinkedIn
by Barry Song via LinkedIn 05 Feb '13

05 Feb '13

LinkedIn ------------ Barry Song requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Barry Accept invitation from Barry Song http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Barry Song http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/rso/196461153/a2mu/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix phantom jacks on VT1708
by David Henningsson 05 Feb '13

05 Feb '13

The VT1708 has no unsol event capability, and polling is set using the "Jack Detect" alsamixer control. In order not to create phantom Jack controls, temporary enable jackpoll during build_controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) Sorry, the previous version of this patch the "int err" got lost. diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9641c0e..e934c49 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -661,6 +661,18 @@ static int via_init(struct hda_codec *codec) return 0; } +static int vt1708_build_controls(struct hda_codec *codec) +{ + /* In order not to create "Phantom Jack" controls, + temporary enable jackpoll */ + int err; + int old_interval = codec->jackpoll_interval; + codec->jackpoll_interval = msecs_to_jiffies(100); + err = via_build_controls(codec); + codec->jackpoll_interval = old_interval; + return err; +} + static int vt1708_build_pcms(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -723,6 +735,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; codec->patch_ops = via_patch_ops; + codec->patch_ops.build_controls = vt1708_build_controls; codec->patch_ops.build_pcms = vt1708_build_pcms; /* clear jackpoll_interval again; it's set dynamically */ -- 1.7.9.5

2 2

[alsa-devel] How exactly snd_pcm_hw_rule_add() should to be used?
by Peter Ujfalusi 05 Feb '13

05 Feb '13

Hi, I got confused (again) regarding to how to use correctly the snd_pcm_hw_rule_add(). Looking at the documentation [1] and to the code in kernel seams to be suggesting the opposite. [A] using the 5.6. example from [1]: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, -1); The [1] tells me that the SNDRV_PCM_HW_PARAM_FORMAT will be refined based on the SNDRV_PCM_HW_PARAM_CHANNELS. In 5.7. of [1] the reverse rule is placed: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_CHANNELS, -1); Which according to the example will refine the SNDRV_PCM_HW_PARAM_CHANNELS based on the SNDRV_PCM_HW_PARAM_FORMAT. However when looking at the code in the kernel the opposite seams to be what is actually happening. [B] using the same 5.6. example from [1]: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, -1); It looks like the SNDRV_PCM_HW_PARAM_CHANNELS should be refined based on the SNDRV_PCM_HW_PARAM_FORMAT. Which one is the correct interpretation? The reason I'm asking is that I need to have a rule which would place constraint (step) on the BUFFER_SIZE based on the used RATE, but I can not use a single snd_pcm_hw_constraint_step(,,BUFFER_TIME,) because of HW limitation. Thank you, Péter [1] http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s09.html

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix phantom jacks on VT1708
by David Henningsson 05 Feb '13

05 Feb '13

The VT1708 has no unsol event capability, and polling is set using the "Jack Detect" alsamixer control. In order not to create phantom Jack controls, temporary enable jackpoll during build_controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9641c0e..d46c448 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -661,6 +661,17 @@ static int via_init(struct hda_codec *codec) return 0; } +static int vt1708_build_controls(struct hda_codec *codec) +{ + /* In order not to create "Phantom Jack" controls, + temporary enable jackpoll */ + int old_interval = codec->jackpoll_interval; + codec->jackpoll_interval = msecs_to_jiffies(100); + err = via_build_controls(codec); + codec->jackpoll_interval = old_interval; + return err; +} + static int vt1708_build_pcms(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -723,6 +734,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; codec->patch_ops = via_patch_ops; + codec->patch_ops.build_controls = vt1708_build_controls; codec->patch_ops.build_pcms = vt1708_build_pcms; /* clear jackpoll_interval again; it's set dynamically */ -- 1.7.9.5

1 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Use asynchronous I/O to write firmware and coefficients
by Mark Brown 04 Feb '13

04 Feb '13

Allow the regmap API to use asynchronous I/O where supported to minimise the delay between transfers, reducing firmware download times. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 82 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 66 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5487a94..be45e2b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/firmware.h> +#include <linux/list.h> #include <linux/pm.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -153,6 +154,43 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +struct wm_adsp_buf { + struct list_head list; + void *buf; +}; + +static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len, + struct list_head *list) +{ + struct wm_adsp_buf *buf = kzalloc(sizeof(*buf), GFP_KERNEL); + + if (buf == NULL) + return NULL; + + buf->buf = kmemdup(src, len, GFP_KERNEL | GFP_DMA); + if (!buf->buf) { + kfree(buf); + return NULL; + } + + if (list) + list_add_tail(&buf->list, list); + + return buf; +} + +static void wm_adsp_buf_free(struct list_head *list) +{ + while (!list_empty(list)) { + struct wm_adsp_buf *buf = list_first_entry(list, + struct wm_adsp_buf, + list); + list_del(&buf->list); + kfree(buf->buf); + kfree(buf); + } +} + #define WM_ADSP_NUM_FW 4 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { @@ -254,6 +292,7 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, static int wm_adsp_load(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); const struct firmware *firmware; struct regmap *regmap = dsp->regmap; unsigned int pos = 0; @@ -265,7 +304,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; - void *buf; + struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -420,18 +459,16 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(region->data, le32_to_cpu(region->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(region->data, + le32_to_cpu(region->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; } - ret = regmap_raw_write(regmap, reg, buf, - le32_to_cpu(region->len)); - - kfree(buf); - + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(region->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -445,12 +482,20 @@ static int wm_adsp_load(struct wm_adsp *dsp) pos += le32_to_cpu(region->len) + sizeof(*region); regions++; } - + + ret = regmap_async_complete(regmap); + if (ret != 0) { + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + goto out_fw; + } + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); out_fw: + regmap_async_complete(regmap); + wm_adsp_buf_free(&buf_list); release_firmware(firmware); out: kfree(file); @@ -655,6 +700,7 @@ out: static int wm_adsp_load_coeff(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); struct regmap *regmap = dsp->regmap; struct wmfw_coeff_hdr *hdr; struct wmfw_coeff_item *blk; @@ -664,7 +710,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; - void *buf; + struct wm_adsp_buf *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -776,8 +822,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(blk->data, le32_to_cpu(blk->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(blk->data, + le32_to_cpu(blk->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; @@ -786,27 +833,30 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", file, blocks, le32_to_cpu(blk->len), reg); - ret = regmap_raw_write(regmap, reg, blk->data, - le32_to_cpu(blk->len)); + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } - - kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); blocks++; } + ret = regmap_async_complete(regmap); + if (ret != 0) + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); out_fw: release_firmware(firmware); + wm_adsp_buf_free(&buf_list); out: kfree(file); return 0; -- 1.7.10.4

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[alsa-devel] [PATCH 0/4] ASoC: atmel: fixes for working as module
by Bo Shen 04 Feb '13

04 Feb '13

The following patches fixes the audio part when working as module Bo Shen (4): ASoC: atmel_pcm: make it buildable as module ASoC: atmel_ssc_dai: remove error set private data ASoC: atmel_ssc_dai: correct sequence when unload ASoC: sam9g20_wm8731: disable clock and correct sequence when unload sound/soc/atmel/atmel-pcm.h | 6 ++++-- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- sound/soc/atmel/sam9g20_wm8731.c | 6 +++--- 3 files changed, 8 insertions(+), 16 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 04 Feb '13

04 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..26ea6ae 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 0, + WM2200_MBIAS_LVL_1V8 = 1, + WM2200_MBIAS_LVL_1V9 = 2, + WM2200_MBIAS_LVL_2V0 = 3, + WM2200_MBIAS_LVL_2V2 = 4, + WM2200_MBIAS_LVL_2V4 = 5, + WM2200_MBIAS_LVL_2V5 = 6, + WM2200_MBIAS_LVL_2V6 = 7, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..e458252 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = wm2200->pdata.micbias[i].mb_lvl + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

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[alsa-devel] CA0132 DSP loader code merged to for-next branch
by Takashi Iwai 04 Feb '13

04 Feb '13

Hi Ian, FYI, right now I merged the pending CA0132 DSP stuff into for-next branch, targeted for 3.9 kernel. I changed the Kconfig to CONFIG_SND_HDA_CODEC_CA0132_DSP and leave it as default NO, in order to be compatible with the older kernels. Let's see whether the things go well with it. thanks, Takashi

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[alsa-devel] [RESEND PATCH v3 0/5] ARM: OMAP4+: HDMI: Update platform devices for audio
by Ricardo Neri 04 Feb '13

04 Feb '13

Hi Mark, Tomi, Liam, Tony, I am resending this series as Mark and possibly other people missed it. This set aims to be the version 3 of my previous submission[1] and aims to address the comments that Mark and Tomi kindly provided on such submission. The creation of the platform device for the HDMI audio interface from within the OMAPDSS HDMI driver that was previously submitted[2] is resubmitted to be complemented with code to relocate to arch/arm/mach-omap2/display.c the creation of the platform devices for the HDMI ASoC codec and card drivers. This series does not break the HDMI audio functionality in any patch. Also, the names of the platform devices are changed to give them more logical and more descriptive names. As the commit commit 14840b9a83c6a56629db2ba0ec247503e975f143 Author: Ricardo Neri <ricardo.neri(a)ti.com> Date: Tue Nov 6 00:19:17 2012 -0600 OMAPDSS: HDMI: Create platform device for audio support is reverted in Tomi's git://gitorious.org/linux-omap-dss2/linux.git master branch, this series applies cleanly. Changes from v1: *Put in a single series all the patches related to platform device updates. *Now HDMI audio works correctly in every patch. *Remove reference to the TPD12S015 HDMI companion chip as the ASoC drivers are not aware of this and other chips could be used in the future. Changes from v2: *Split in two patches the renaming and the relocation of the platform devices. *Create a separate patch to pass only the address offset of the DMA data port to the audio interface platform device. *Keep the name hdmi-audio-codec to not refer to explicitly to OMAP. The codec can be made generic in a different patch series submitted to alsa-devel. BR, Ricardo [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg80785.html [2]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html Ricardo Neri (5): ARM: OMAP4+: HDMI: Rename platform devices for ASoC drivers ARM: OMAP4: Assign IDs to DSS HDMI devices ARM4: OMAP4+: HDMI: Relocate devices for audio codec and card ARM: OMAP4+: HDMI: Relocate the device for audio interface ARM: OMAP4+: HDMI: Refine the DMA port resource for audio arch/arm/mach-omap2/board-4430sdp.c | 9 ++--- arch/arm/mach-omap2/board-omap4panda.c | 9 ++--- arch/arm/mach-omap2/devices.c | 31 ---------------- arch/arm/mach-omap2/display.c | 31 ++++++++++++++++ drivers/video/omap2/dss/hdmi.c | 62 ++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 4 +-- sound/soc/omap/omap-hdmi.c | 5 ++- sound/soc/omap/omap-hdmi.h | 2 -- 8 files changed, 103 insertions(+), 50 deletions(-) -- 1.7.10.4

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[alsa-devel] alsa usb-audio sound corruption caused by cached vmalloc mapping
by Matthieu CASTET 04 Feb '13

04 Feb '13

Hi, on a armv5 soc, we have audio corruption while using a usb-audio sound card. After analysis, it comes from alsa vmalloc mapping. in the current kernel vmalloc mapping used by alsa are cached. This cause coherency problem on armv5 architecture because the cache are VIVT : - alsa kernel driver map a buffer with vmalloc. The memory attribute is cached - userspace map vmalloc buffer. The memory attribute is cached But because the cache are VIVT, these two mapping don't share the same cache line and won't see the same data. This corruption can be hidden by context switch because cache are flushed on context switch. On x86 or armv7, the cache are PIPT and there is no such problem. There was a commit "pcm - Call pgprot_noncached() for vmalloc'ed buffers" (c32d977b8157bf67cdf47729ce7dd054a26eb534), that made the userspace mapping not cached, but this caused issues because the kernel vmalloc mapping was still cached. Instead of revering this commit, we could have made the kernel vmalloc noncached (see attached patch). It fix the issue on armv5. But on some architecture (armv7 for example) you can't have mapping for the same physical memory cached (normal cached) and noncached (device memory). So the vmalloc and userspace mapping will conflict with the kernel direct RAM mapping. A workaround could be to use writecombine protection instead of noncached. This work because writecombine is mapped on armv7 as L_PTE_MT_BUFFERABLE (normal uncached) that is compatible with cached memory (the same trick is used for dmacoherent). But that's architecture specific may not work on others architectures. What's the best way the fix this issue on all architecture ? Matthieu

2 1

[alsa-devel] alsa bt878 driver
by grigore calugar 04 Feb '13

04 Feb '13

Please add support in "snd-bt87x" for this card: 137 -> Conceptronic CTVFMi v2 [036e:109e] Best regards.

2 1