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[alsa-devel] [PATCH 1/1] ALSA: usb-audio: dynamic detection of MIDI interfaces in uaxx-quirk
by pedro.lopez.cabanillas＠gmail.com 08 Oct '08

08 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> The MIDI interfaces have to be detected dynamically for Edirol devices ua-700, ua-25 and ua4-fx. This patch reverses the wrong changes made by my other patch in uaxx-quirk. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6e70ba4..bbd70d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3023,6 +3023,31 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); + if (altsd->bNumEndpoints == 2) { + static const struct snd_usb_midi_endpoint_info ua700_ep = { + .out_cables = 0x0003, + .in_cables = 0x0003 + }; + static const struct snd_usb_audio_quirk ua700_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &ua700_ep + }; + static const struct snd_usb_midi_endpoint_info uaxx_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk uaxx_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &uaxx_ep + }; + if (chip->usb_id == USB_ID(0x0582, 0x002b)) + return snd_usb_create_midi_interface(chip, iface, + &ua700_quirk); + else + return snd_usb_create_midi_interface(chip, iface, + &uaxx_quirk); + } + if (altsd->bNumEndpoints != 1) return -ENXIO; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 85532cf..69689e7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -870,11 +870,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1216,11 +1212,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1375,11 +1367,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 -- 1.5.2.4

2 1

[alsa-devel] ASoC WM8510 Mic PGA problem
by ext-jukka.hynninen＠vaisala.com 08 Oct '08

08 Oct '08

Hi all, I'm facing trouble with programmable mic gain stage and WM8510 ASoC driver. First of all, the DAPM control "Mic PGA Switch" seems to be operating just the opposite way the codec datasheet suggests. I inverted the control with expected result, PGA now almost works. When I start capturing, the "Mic PGA" control is instantly turned off again. If I manually enable the control while the capture is running, it works just fine. Any ideas? Am I missing some DAPM routings? -Jukka

3 11

Re: [alsa-devel] [PATCH] Sigmatel stac92hd71b7 low HP volume
by Takashi Iwai 08 Oct '08

08 Oct '08

[please don't drop Cc to alsa-devel ML] At Mon, 6 Oct 2008 12:48:43 +0200, Klaas De Craemer wrote: > > Hello, > > I'm sure that the audioquality will degrade somewhat because of the additional amplification, > but before, the HP-out was not usable at all. I had to use an external ampli just to use my > headphones, even after turning all volume sliders to 100%. Then again, when removing the HP, > the volume though the internal speakers was way too loud... > > I'm not sure what you mean with the output pin and headphone control, but I have uploaded the > output of the codegraph tool before and after my modification here: > http://users.vtk.be/~s0159120/codec_original.svg > http://users.vtk.be/~s0159120/codec_final.svg > > Below is the output of alsa-info with the patched module > Klaas Thanks. So, if I understand correctly, when the pin 0x0a is connected directly from DAC0 0x10, the output level is too low even if you set the all volumes maximal. In the alsa-info.sh output, I see the following: > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] which means that you decrease the output level intentionally. Is the HP output still too low even if you set to the higher value? Takashi > > ----- > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.51 > !!################################ > > !!Script ran on: Mon Oct 6 12:42:48 CEST 2008 > > > !!Linux Distribution > !!------------------ > > Debian GNU/Linux lenny/sid \n \l > > > !!Kernel Information > !!------------------ > > Kernel release: 2.6.26-5.slh.6-sidux-686 > Operating System: GNU/Linux > Architecture: i686 > Processor: unknown > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.16 > Library version: > Utilities version: 1.0.16 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xf6fdc000 irq 21 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > > > !!Advanced information - PCI Vendor/Device/Susbsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:293e (rev 03) > Subsystem: 1028:024f > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-pcsp: index=-2 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : 0 > id : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > single_cmd : N > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: IDT 92HD71B7X > Address: 0 > Vendor Id: 0x111d76b2 > Subsystem Id: 0x1028024f > Revision Id: 0x100302 > No Modem Function Group found > Default PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 > GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 > Node 0x0a [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x081c: OUT HP Detect > Pin Default 0x0421101f: [Jack] HP Out at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=30, enabled=1 > Connection: 3 > 0x10 0x11 0x17* > Node 0x0b [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x04a11221: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0c [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x0e [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x23a1902e: [Jack] Mic at Sep Left > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0xe > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x23014250: [Jack] Line Out at Sep Left > Conn = 1/8, Color = Green > DefAssociation = 0x5, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x3d 0x3d] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x12 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x1c > Processing caps: benign=0, ncoeff=0 > Node 0x13 [Audio Input] wcaps 0x1d0541: Stereo > Converter: stream=0, channel=0 > SDI-Select: 0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x1d > Processing caps: benign=0, ncoeff=0 > Node 0x14 [Pin Complex] wcaps 0x400100: Mono > Pincap 0x0810: OUT > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x16 > Node 0x15 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x10* 0x11 0x17 > Node 0x16 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x15 > Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x1f 0x1f] [0x1f 0x1f] [0x97 0x97] [0x97 0x97] [0x97 0x97] > Connection: 5 > 0x10 0x11 0x27 0x1a 0x1b > Node 0x18 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x90a000f0: [Fixed] Mic at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x19 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x01 0x01] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x00 0x00] > Connection: 4 > 0x1a* 0x17 0x18 0x19 > Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x80 0x80] > Connection: 4 > 0x1b* 0x17 0x18 0x19 > Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x24 > Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital > Pincap 0x0810010: OUT EAPD > EAPD 0x0: > Pin Default 0x40f000f6: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x6 > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Connection: 2 > 0x24* 0x25 > Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x40f000f7: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x7 > Pin-ctls: 0x00: > Connection: 1 > 0x25 > Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x22 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x24 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x21* 0x1c 0x1d > Node 0x25 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x22* 0x1c 0x1d > Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > Node 0x27 [Pin Complex] wcaps 0x400000: Mono > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Node 0x28 [Volume Knob Widget] wcaps 0x600000: Mono > Volume-Knob: delta=1, steps=127, direct=1, val=127 > Connection: 2 > 0x10* 0x11 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw---- 1 root audio 116, 7 okt 6 12:40 /dev/snd/controlC0 > crw-rw---- 1 root audio 116, 6 okt 6 12:40 /dev/snd/hwC0D0 > crw-rw---- 1 root audio 116, 5 okt 6 12:40 /dev/snd/pcmC0D0c > crw-rw---- 1 root audio 116, 4 okt 6 12:41 /dev/snd/pcmC0D0p > crw-rw---- 1 root audio 116, 3 okt 6 11:41 /dev/snd/seq > crw-rw---- 1 root audio 116, 2 okt 6 11:41 /dev/snd/timer > > > !!Aplay/Arecord output > !!------------ > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 2/2 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xf6fdc000 irq 21' > Mixer name : 'IDT 92HD71B7X' > Components : 'HDA:111d76b2' > Controls : 19 > Simple ctrls : 14 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 127 > Mono: Playback 94 [74%] [-24.75dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Front',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [on] > Front Right: Capture 0 [0%] [0.00dB] [on] > Simple mixer control 'Capture Mux',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 1 [33%] [10.00dB] > Front Right: 1 [33%] [10.00dB] > Simple mixer control 'Capture Mux',1 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'Capture',1 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [off] > Front Right: Capture 0 [0%] [0.00dB] [off] > Simple mixer control 'Analog Loopback',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Analog Loopback',1 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Digital Input Source',0 > Capabilities: enum > Items: 'Analog Inputs' 'Digital Mic 1' > Item0: 'Analog Inputs' > Simple mixer control 'Digital Mic 1',0 > Capabilities: cvolume > Capture channels: Front Left - Front Right > Limits: Capture 0 - 3 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > Simple mixer control 'Input Source',1 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.2 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.3 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > value false > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > index 1 > value false > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > value.0 0 > value.1 0 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > index 1 > value.0 0 > value.1 0 > } > control.9 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.10 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > index 1 > value.0 0 > value.1 0 > } > control.11 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Front Playback Volume' > value.0 127 > value.1 127 > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.13 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Headphone Playback Volume' > value.0 127 > value.1 127 > } > control.14 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.15 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Digital Mic 1 Capture Volume' > value.0 0 > value.1 0 > } > control.16 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 'Analog Inputs' > comment.item.1 'Digital Mic 1' > iface MIXER > name 'Digital Input Source' > value 'Analog Inputs' > } > control.17 { > comment.access 'read write' > comment.type INTEGER > comment.count 1 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Master Playback Volume' > value 94 > } > control.18 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Master Playback Switch' > value true > } > control.19 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > comment.dbmin -5100 > comment.dbmax 0 > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > snd_hda_intel > snd_hwdep > nls_cp437 > vfat > fat > loop > isofs > udf > nvidia > ppdev > lp > parport > acpi_cpufreq > cpufreq_performance > cpufreq_stats > cpufreq_ondemand > cpufreq_powersave > cpufreq_conservative > freq_table > ipv6 > af_packet > hfsplus > nls_utf8 > ntfs > nls_base > fuse > snd_pcm_oss > snd_pcm > snd_page_alloc > snd_mixer_oss > pcmcia > firmware_class > snd_seq_dummy > rtc_cmos > snd_seq_oss > rtc_core > rtc_lib > dcdbas > snd_seq_midi_event > snd_seq > snd_timer > snd_seq_device > serio_raw > psmouse > i2c_i801 > snd > pcspkr > evdev > i2c_core > yenta_socket > rsrc_nonstatic > pcmcia_core > soundcore > video > output > wmi > ac > button > battery > intel_agp > ext3 > jbd > mbcache > dm_mirror > dm_log > dm_snapshot > dm_mod > firewire_sbp2 > sg > sr_mod > cdrom > sd_mod > usbhid > hid > ff_memless > ahci > libata > scsi_mod > dock > ricoh_mmc > sdhci > mmc_core > firewire_ohci > firewire_core > crc_itu_t > ehci_hcd > uhci_hcd > usbcore > e1000e > thermal > processor > fan > > > On Monday 06 October 2008 11:44:33 you wrote: > > At Sun, 5 Oct 2008 22:10:46 +0200, > > > > Klaas De Craemer wrote: > > > Hello, > > > > > > A few days ago, I posted a patch to the linux kernel list, but afterwards > > > I realised that it is more appropriate for the alsa list. So I repost it > > > here. > > > > > > Below is my patch for an issue on Dell E6500 notebooks with the Sigmatel > > > stac92hd71b7 codec. In the default snd-hda-intel configuration, the sound > > > going to the headphone jack goes comes directly from the DAC, resulting > > > in very low volume output. The patch changes the codec's widget > > > connections so that the internal mixer is put in between and sets its > > > input amplifiers to the maximum value. The output to the internal speaker > > > port has been left untouched. > > > I have tested this and it gives a much better output. > > > > Thanks for the patch. > > > > I wonder whether this really improves the quality. > > Basically what you does is to set the volume of the input mixer over > > 0dB, and this is usually bad when really high signals come in. > > > > Just to be sure, is the output pin set to the headphone pin control? > > You can give alsa-info.sh output (run with --no-upload option) for > > further analysis. The script is found in > > http://www.alsa-project.org/alsa-info.sh > > > > > > thanks, > > > > Takashi > >

2 6

[alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by Sebastian Schneider 08 Oct '08

08 Oct '08

What about we start porting the driver on our own? -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

1 0

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with debugfs filesystem
by Troy Kisky 08 Oct '08

08 Oct '08

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f5a6dbb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -516,6 +516,10 @@ struct snd_soc_device { struct delayed_work delayed_work; struct work_struct deferred_resume_work; void *codec_data; +#ifdef CONFIG_SND_SOC_DEBUG_FS + struct dentry *debugfs_root; + struct dentry *debugfs_codec_reg; +#endif }; /* runtime channel data */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..aad7e0b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,14 @@ if SND_SOC config SND_SOC_AC97_BUS bool +config SND_SOC_DEBUG_FS + boolean "changing codec registers through debugfs (DEVELOPMENT)" + depends on DEBUG_FS + help + The ability to directly write codec registers through + /sys/kernel/debug/xx/codec_reg is useful when developing + a new codec. + # All the supported Soc's source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..46827cb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -962,10 +962,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +1000,113 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_SND_SOC_DEBUG_FS +static int code_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t code_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t res; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + + res = soc_codec_reg_show(devdata, buf); + res = simple_read_from_buffer(user_buf, count, ppos, buf, res); + kfree(buf); + return res; +} + +static ssize_t code_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = code_reg_open_file, + .read = code_reg_read_file, + .write = code_reg_write_file, +}; + +static void soc_init_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *debugfs_codec_reg; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + debugfs_codec_reg = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!debugfs_codec_reg) + goto exit2; + + socdev->debugfs_root = root; + socdev->debugfs_codec_reg = debugfs_codec_reg; + return; +exit2: + debugfs_remove(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove(socdev->debugfs_codec_reg); + debugfs_remove(socdev->debugfs_root); + socdev->debugfs_codec_reg = NULL; + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1217,6 +1320,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1254,6 +1358,7 @@ free_card: if (codec->card) snd_card_free(codec->card); device_remove_file(socdev->dev, &dev_attr_codec_reg); + soc_cleanup_debugfs(socdev); mutex_unlock(&codec->mutex); } EXPORT_SYMBOL_GPL(snd_soc_free_pcms); -- 1.5.4.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 07 Oct '08

07 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..1077e9d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,8 +3016,8 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); @@ -3031,20 +3031,20 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &ua700_ep }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { + static const struct snd_usb_midi_endpoint_info uaxx_ep = { .out_cables = 0x0001, .in_cables = 0x0001 }; - static const struct snd_usb_audio_quirk ua25_quirk = { + static const struct snd_usb_audio_quirk uaxx_quirk = { .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep + .data = &uaxx_ep }; if (chip->usb_id == USB_ID(0x0582, 0x002b)) return snd_usb_create_midi_interface(chip, iface, &ua700_quirk); else return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); + &uaxx_quirk); } if (altsd->bNumEndpoints != 1) @@ -3376,9 +3376,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..815bf36 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1197,15 +1197,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1338,6 +1338,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

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Re: [alsa-devel] 2 Au1xxx (Alchemy) AC97 drivers
by Manuel Lauss 07 Oct '08

07 Oct '08

On Tue, Oct 07, 2008 at 08:28:26AM -0500, Kevin Hickey wrote: > There appear to be 2 AC97 drivers for Alchemy in the latest Linux tree. > One is in sound/mips and the other is in sound/soc. Are both going to > be maintained going forward? If not, which one will be? The one in sound/soc is for Au1200/Au1550; I think the sound/mips one only works on Au1000/Au1500. Manuel Lauss -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [topic/asoc][PATCH 0/2] minor clean-ups and aic3x naming change
by Jarkko Nikula 07 Oct '08

07 Oct '08

These two patches are generated agains Takashi's topic/asoc branch. First one will remove unused AUDIO_NAME define from codec drivers. IMO it didn't look so readable to use this define so I end up removing it. Second one will use uniform naming like other tlv320aic drivers for tlv320aic3x. -- Jarkko

3 4

[alsa-devel] size of snd_hda_intel
by Jon Smirl 07 Oct '08

07 Oct '08

>ismod snd_hda_intel 435,772 435,772 bytes seems awfully large for an audio driver. The large size seems to be due to having support for multiple codecs in the driver. Could a generic hda driver load the correct codec dynamically instead of building them all in statically? I have a Realtek codec. -- Jon Smirl jonsmirl(a)gmail.com

5 8

[alsa-devel] Fwd: AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 07 Oct '08

07 Oct '08

---------- Forwarded message ---------- From: Jesslyn Abdul Salam <jesslyn.abdulsalam(a)gmail.com> Date: Tue, Oct 7, 2008 at 5:36 PM Subject: Re: [alsa-devel] AIC23 SOC Driver on OSK5912 To: Arun KS <arunks(a)mistralsolutions.com> Yes, it is there. but it shows as "tlv320aic23b" instead of "tlv320aic" Will that be the reason ???? On Tue, Oct 7, 2008 at 5:32 PM, Arun KS <arunks(a)mistralsolutions.com> wrote: > On Tue, Oct 7, 2008 at 5:24 PM, Mark Brown <broonie(a)sirena.org.uk> wrote: > > On Tue, Oct 07, 2008 at 05:18:55PM +0530, Jesslyn Abdul Salam wrote: > > > >> I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for the > past > >> few days. I got the driver patch from the linux-omap/alsa-devel mailing > >> lists. The driver got compiled along just fine. But when the kernel > boots, > >> no sound cards have been listed. > > > > CCing in Arun KS <arunks(a)mistralsolutions.com> who contributed that > > support. > > > >> Advanced Linux Sound Architecture Driver Version 1.0.17. > > > >> ASoC version > >> 0.13.2 > >> AIC23 Audio Codec > >> 0.1 > >> MCLK = 12000000 [12000000], usecount = > >> 0 > >> ALSA device > >> list: > >> No soundcards > >> found. > > > >> Any one have tested this before ? > > Can you check i2c? > > # cat /sys/bus/i2c/devices/1-001b/name > tlv320aic23 > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

2 2