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[alsa-devel] Speaker and Headphone Output problem with ALC1200
by Thomas Pietrzak 20 Apr '09

20 Apr '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a MSI GX720-061 laptop, and I unfortunately experiment the "Speaker and Headphone Output" problem. The sound card is the following: $ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) $ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC1200 Analog [ALC1200 Analog] Sous-périphériques: 0/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC1200 Digital [ALC1200 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 Note that in Ubuntu Intrepid it was recognized as ALC888. However my problem remains in Jaunty. These are the alsa packages I have (Ubuntu Jaunty): $ dpkg -l |grep alsa ii alsa-base 1.0.18.dfsg-1ubuntu6 ALSA driver configuration files ii alsa-oss 1.0.17-1 ALSA wrapper for OSS applications ii alsa-tools 1.0.18-1ubuntu2 Console based ALSA utilities for specific ha ii alsa-tools-gui 1.0.18-1ubuntu2 GUI based ALSA utilities for specific hardwa ii alsa-utils 1.0.18-1ubuntu10 ALSA utilities ii alsamixergui 0.9.0rc2-1-9 graphical soundcard mixer for ALSA soundcard ii alsaplayer-alsa 0.99.80-3ubuntu1 PCM player designed for ALSA (ALSA output mo ii alsaplayer-common 0.99.80-3ubuntu1 PCM player designed for ALSA (common files) ii alsaplayer-gtk 0.99.80-3ubuntu1 PCM player designed for ALSA (GTK version) ii bluez-alsa 4.32-0ubuntu3 Bluetooth audio support ii gnome-alsamixer 0.9.7~cvs.20060916.ds.1-2 ALSA sound mixer for GNOME ii gstreamer0.10-alsa 0.10.22-4 GStreamer plugin for ALSA ii libesd-alsa0 0.2.40-0ubuntu3 Enlightened Sound Daemon (ALSA) - Shared lib ii libpt-1.10.10-plugins-alsa 1.10.10-2ubuntu3 Portable Windows Library Audio Plugin for th ii libpt2.4.2-plugins-alsa 2.4.2-2ubuntu1 PTLib audio plugin for the ALSA Interface rc libsdl1.2debian-alsa 1.2.13-2ubuntu1 Simple DirectMedia Layer (with X11 and ALSA This is my kernel: $ uname -a Linux rigel 2.6.28-11-generic #38-Ubuntu SMP Fri Mar 27 10:01:17 UTC 2009 x86_64 GNU/Linux My problem is that I can't have both 7.1 sound and headphones working. If I add this line to /etc/modprobe.d/alsa-base.conf: options snd-hda-intel model=6stack-dig position_fix=0 I have 7.1 sound. But if I plug my headphones, I have sound on both headphones and PC speakers. The only way to mute the speakers and still have sound in the headphones is to uncheck "headphones" in the settings tab of the gnome mixer, and to mute "center" and "LFE". I tried the option probe_mask=1 that is recommended for ALC1200, but it didn't changed anything. I found a solution by googleing. It suggested to add this instead: options snd-hda-intel model=targa-2ch-dig options snd-hda-intel enable_msi=1 In this case I only have headphones when I plug my headphones. However I don't have 7.1 sound anymore, but only stereo. Can someone help me to make my headphones work normally with 7.1 sound? Please feel free to ask me any information you need. Many thanks, - --- Tom -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAknsLSoACgkQngTxdcSw9xxLXACcCX/k6FOFDoU7BWNUUj9UlDTA Sp0AnjjbC42TpssAREpB7nmaVOYORDQe =STMu -----END PGP SIGNATURE----- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Mon Apr 20 07:54:18 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!Kernel Information !!------------------ Kernel release: 2.6.28-11-generic Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.18rc3 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf8ef8000 irq 22 !!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: 1462:7220 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-hda-intel: model=6stack-dig probe_mask=1 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 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 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 : 6stack-dig,<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: Realtek ALC1200 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x14627220 Revision Id: 0x100101 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=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo 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 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x1b 0x1b] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [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 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x09 0x09] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital 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 0x10 Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01214410: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger 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 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x99130120: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01012050: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19860: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30961: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x6, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181306f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x6, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99130130: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN 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 Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x01454140: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Green DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN 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 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 8 Apr 20 09:10 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 7 Apr 20 09:13 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 6 Apr 20 09:51 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Apr 20 09:10 /dev/snd/pcmC0D1c crw-rw----+ 1 root audio 116, 4 Apr 20 09:10 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 3 Apr 20 09:10 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 2 Apr 20 09:10 /dev/snd/timer !!ALSA configuration files !!------------------------ !!User specific config file (~/.asoundrc) # ALSA library configuration file # Include settings that are under the control of asoundconf(1). # (To disable these settings, comment out this line.) </home/tom/.asoundrc.asoundconf> !!asoundconf-generated config file # ALSA library configuration file managed by asoundconf(1). # # MANUAL CHANGES TO THIS FILE WILL BE OVERWRITTEN! # # Manual changes to the ALSA library configuration should be implemented # by editing the ~/.asoundrc file, not by editing this file. !defaults.pcm.card Intel defaults.ctl.card Intel defaults.pcm.device 0 defaults.pcm.subdevice -1 defaults.pcm.nonblock 1 defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 defaults.pcm.dmix.max_periods 0 defaults.pcm.dmix.rate 48000 defaults.pcm.dmix.format "unchanged" defaults.pcm.dmix.card defaults.pcm.card defaults.pcm.dmix.device defaults.pcm.device defaults.pcm.dsnoop.card defaults.pcm.card defaults.pcm.dsnoop.device defaults.pcm.device defaults.pcm.front.card defaults.pcm.card defaults.pcm.front.device defaults.pcm.device defaults.pcm.rear.card defaults.pcm.card defaults.pcm.rear.device defaults.pcm.device defaults.pcm.center_lfe.card defaults.pcm.card defaults.pcm.center_lfe.device defaults.pcm.device defaults.pcm.side.card defaults.pcm.card defaults.pcm.side.device defaults.pcm.device defaults.pcm.surround40.card defaults.pcm.card defaults.pcm.surround40.device defaults.pcm.device defaults.pcm.surround41.card defaults.pcm.card defaults.pcm.surround41.device defaults.pcm.device defaults.pcm.surround50.card defaults.pcm.card defaults.pcm.surround50.device defaults.pcm.device defaults.pcm.surround51.card defaults.pcm.card defaults.pcm.surround51.device defaults.pcm.device defaults.pcm.surround71.card defaults.pcm.card defaults.pcm.surround71.device defaults.pcm.device defaults.pcm.iec958.card defaults.pcm.card defaults.pcm.iec958.device defaults.pcm.device defaults.pcm.modem.card defaults.pcm.card defaults.pcm.modem.device defaults.pcm.device defaults.pcm.file_format "raw" defaults.pcm.file_truncate true defaults.rawmidi.card 0 defaults.rawmidi.device 0 defaults.rawmidi.subdevice -1 defaults.hwdep.card 0 defaults.hwdep.device 0 defaults.timer.class 2 defaults.timer.sclass 0 defaults.timer.card 0 defaults.timer.device 0 defaults.timer.subdevice 0 defaults.namehint.showall off defaults.namehint.basic on defaults.namehint.extended off !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xf8ef8000 irq 22' Mixer name : 'Realtek ALC1200' Components : 'HDA:10ec0888,14627220,00100101' Controls : 40 Simple ctrls : 22 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 24 [77%] [-10.50dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [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 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Side',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 23 [74%] [0.00dB] [on] Front Right: Playback 23 [74%] [0.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined Playback channels: Mono Capture channels: Mono Mono: Playback [off] Capture [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 9 [29%] [-21.00dB] [on] Front Right: Playback 9 [29%] [-21.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 27 [87%] [24.00dB] [on] Front Right: Capture 27 [87%] [24.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-16.50dB] [off] Front Right: Capture 0 [0%] [-16.50dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '6ch' '8ch' Item0: '8ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 31 value.1 31 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 31 value.1 31 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Surround Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Center Playback Volume' value 31 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'LFE Playback Volume' value 31 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Center Playback Switch' value true } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'LFE Playback Switch' value true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 31 value.1 31 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Side Playback Switch' value.0 true value.1 true } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'CD Playback Volume' value.0 23 value.1 23 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'CD Playback Switch' value.0 true value.1 true } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.15 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 true value.1 true } control.16 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mic Boost' value.0 0 value.1 0 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Front Mic Boost' value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'PC Speaker Playback Volume' value.0 9 value.1 9 } control.23 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'PC Speaker Playback Switch' value.0 true value.1 true } control.24 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 27 value.1 27 } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.26 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.28 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' comment.item.2 Line comment.item.3 CD iface MIXER name 'Input Source' value Mic } control.29 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' comment.item.2 Line comment.item.3 CD iface MIXER name 'Input Source' index 1 value Mic } control.30 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 '6ch' comment.item.1 '8ch' iface MIXER name 'Channel Mode' value '8ch' } control.31 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.32 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.33 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.34 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.36 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Capture Switch' value false } control.37 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Capture Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.38 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 24 } control.39 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.40 { 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 binfmt_misc ppdev bridge stp bnep vboxnetflt vboxdrv input_polldev sbp2 lp parport arc4 ecb snd_hda_intel iwlagn snd_pcm_oss iwlcore snd_mixer_oss led_class snd_pcm uvcvideo mac80211 snd_timer joydev compat_ioctl32 snd videodev sdhci_pci soundcore video psmouse v4l1_compat sdhci snd_page_alloc iTCO_wdt iTCO_vendor_support output pcspkr serio_raw wacom intel_agp cfg80211 nvidia usbhid ohci1394 ieee1394 r8169 mii fbcon tileblit font bitblit softcursor

1 0

[alsa-devel] [PATCH 1/1] ASoC: TWL4030: Add support Voice DAI
by Joonyoung Shim 20 Apr '09

20 Apr '09

Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz (voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. If the system master clock is not 26MHz, the voice PCM interface is not available. The PCM voice interface has two modes - PCM mode1 : This uses the rising edge of the clock signal - PCM mode2 : This uses the falling edge of the clock signal PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 135 +++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 16 +++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 150 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index db7fecf..13461d1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1437,6 +1437,107 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) + return -EPERM; + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_DSP_B: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1448,8 +1549,14 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { - .name = "twl4030", +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", .channels_min = 2, @@ -1463,6 +1570,22 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1503,8 +1626,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1598,13 +1721,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..305b8ba 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -171,6 +171,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +247,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- 1.5.6.3

3 7

Re: [alsa-devel] [PATCH v2 2/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by naresh 20 Apr '09

20 Apr '09

Dave, Thanks for the review. I will re-submit the patch after fixing the comments. Regards, Naresh > >On Thursday 16 April 2009, Medisetty, Naresh wrote: >> arch/arm/mach-davinci/dm355.c | 29 ++++++++++++- >> arch/arm/mach-davinci/dm644x.c | 27 +++++++++++- >> arch/arm/mach-davinci/dm646x.c | 67 >++++++++++++++++++++++++++ >> arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- > >Note the merge dependency: most of those aren't yet in mainline. >I think Kevin sent the dm644x.c changes to the ARM list for their >first review cycle recently. > >My comments below are for the dm355.c changes, but they apply just >as much to the other two SoC-specific setup files. > > >> +static struct resource dm355_evm_snd_resources[] = { >> + { >> + .start = DAVINCI_ASP1_BASE, >> + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, >> + .flags = IORESOURCE_MEM, >> + }, >> +}; >> + >> +static struct evm_snd_platform_data dm355_evm_snd_data = { >> + .clk_name = "asp1", >> + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, >> + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, >> +}; > >If you're going to move to proper driver model support >(good!) ... the DMA channels should be IORESOURCE_DMA >resources, and the clock should be associated directly >with that platform device. > >That might seem to leave that current platform data >struct as useless ... not so! It should tell about how >the audio is configured: whether RX and TX are both >wired, if they're stereo or what, how *audio* clocks >are set up, and so forth. All that is specific to the >specific *board* in use, not to that SoC, so it should >be provided by board setup code. A lot of it is now >encoded in the sound/soc/davinci/davinci-evm.c file. > >Example: dm6446 evm supports six different audio clocks, >but right now one choice is fixed at board setup time. > >ASoC seems to have ways to let drivers make that choice. >That could be handled by board-specific callbacks and data >tables, and would be good to package in platform data. >A different dm6446 board may use a different audio clock >setup, with different choices. > > >> + >> +static struct platform_device dm355_asoc_device = { >> + .name = "davinci-asoc", >> + .id = -1, >> + .dev.platform_data = &dm355_evm_snd_data, >> + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), >> + .resource = dm355_evm_snd_resources, > >Looks like the alignment goofage there (for "=") is more than >what can be explained by patch mangling from email... > > >> +}; > > >> @@ -555,6 +579,7 @@ static int __init dm355_init_devices(void) >> >> davinci_cfg_reg(DM355_INT_EDMA_CC); >> platform_device_register(&dm355_edma_device); >> + platform_device_register(&dm355_asoc_device); > >This does *NOT* belong in the generic dm355 setup code like that. >Not every DM355 board will even have audio! Or have it wired >exactly like Spectrum's EVM boards. > >It'd be appropriate to add dm355_init_asoc(asp_num, platform_data) >or something similar, so that different boards wouldn't need to >repeat the won't-change parts. That routine might need to set up >some pin, IRQ, and EDMA muxing. Boards with no audio support would >not call that routine. Ones with two audio interfaces might need >to call it once for each interface. > > >> return 0; >> } >> postcore_initcall(dm355_init_devices); > > >Similar comments for the two other SoC chips touched by this patch; >although obviously the dm6446 wouldn't need an asp_num, etc. > >- dave

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[alsa-devel] Via 1708S no S/PDIF out
by Gus Zernial 19 Apr '09

19 Apr '09

I posted this on alsa-user and got no response. I have an ASUS M4A78T-E motherboard system, which has 7.1 channel HD Audio out using Via VT1708S chip, (plus ATI HDMI out). I'm running Kubuntu Jaunty 9.04 with a custom 2.6.28.9 kernel. I am getting sound out the Line Out/Speaker Out jacks, but no sound through the Toslink S/PDIF Out jack. The S/PDIF out channel does not show in aplay -L command output, and there is no S/PDIF out slider in amixer. This is a dual-boot system and everything, including S/PDIF out, works fine with WinXP. In linux, I'm running alsa version 1.0.19. I can post more information if needed, but I think I have the right modules loaded, and I've tried various "model=" options for snd_hda_intel, so far none provide S/PDIF out. My questions are .... can someone tell me if S/PDIF out is supposed to work with this audio chip, and if so what do I need to do configuration-wise to get it to work? Thankyou ...

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[alsa-devel] [PATCH] ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
by Philipp Zabel 19 Apr '09

19 Apr '09

Those macros are just screwed as soon as CONFIG_PXA25x is enabled. This patch - changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device - adds a corresponding ssp_get_scr function. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- To me, using ssp_dev seems to be cleaner, as all the places where ssp_set_scr is called, we already have an ssp_dev *ssp = priv->dev.ssp set up, which allows us to call ssp_set_scr(ssp, ...) instead of ssp_set_scr(&priv->dev, ...). Same for ssp_get_scr. --- sound/soc/pxa/pxa-ssp.c | 36 ++++++++++++++++++++++++++++-------- 1 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..d979c5c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. -- 1.6.2.3

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[alsa-devel] [PATCH] General fix for Palm27x aSoC driver
by Marek Vasut 19 Apr '09

19 Apr '09

Hi, Firstly, this patch makes the palm27x asoc driver a little more sane. Also, since all affected devices use GPIO95 as AC97_nRESET, this patch sets that properly. Affected are PalmT5, TX and LifeDrive. Please consider applying, it's against Russells tree. If Eric wants to merge it, I can remake it. Thanks.

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[alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Gustavo Vieira 19 Apr '09

19 Apr '09

Hi all, I've got sound problems in my HP Pavilion dv4-1125br. The sound card is a Intel 82801I with the IDT 92HD71B7X codec. Sound works ok, including the internal mic and external mic jack, but I still have a couple of problems. The main problem is that I don't have jack sense in the headphone jack. Headphones do not work and I'm unable even to make them work manually through the mixer. This machine got two headphone jacks in front and none of them works. I've tested it since alsa-driver 1.0.17 and it has never worked. The other problem is that the speaker volume is low. Even if I increase all available mixer controls (PCM and Master), the volume is still tiny. This is a regression. In alsa-driver 1.0.17 (Fedora 10, kernel 2.6.27.21) the volume is ok, the problem appeared sometime after that and alsa-driver 1.0.18a (Fedora 11 beta, kernel 2.6.29.1). One thing I noticed between these versions is the disappearance of the Front mixer control present in 1.0.17. Maybe that is the cause of the problem. I've tried all models available to my codec and none of them solved the issue. Actually, all of them seemed to created other problems. The no model version seems the best right now. I'm attaching the alsa-info output for the Fedora 10 and Fedora 11 beta configurations. I've tried to download and compile the alsa-driver snapshot, but the compilation failed early on with an a patch failure in my kernel 2.6.29.1 system. Abraços, Gustavo

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[alsa-devel] how to find the last plugged in USB headset ?
by reetesh chandran 18 Apr '09

18 Apr '09

Hi all, Is there a way to find the card number associated with the USB headset that was plugged in last. I am listening to udev events and I can find the card number by comparing the VendorId product Id combination. But if there are multiple devices of same VendorId and productId, how do I find the card number associated with the device that was plugged in the last ? Thanks in advance ? Reetesh

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[alsa-devel] SoC pxa2xx-ac97 + wm9705 + touchscreen suspend/resume
by Russell King - ARM Linux 18 Apr '09

18 Apr '09

Trying Linus' latest (as of this morning) out on a PXA255 platform with a WM9705 codec reveals a number of problems: 1. when suspend occurs, we turn the AC97 link off by setting the GCR_ACLINK_OFF bit, and stopping the functional units clock. On resume, we reconfigure the pins and re-enable the clock. Unfortunately, this results in GCR being left with zeros, and GSR remains all zeros, and the codec is inaccessible, with lots of warnings from the read/write functions scrolling by: pxa2xx_ac97_write: write error (ac97_reg=118 GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=118 GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=118 GSR=0x0) ... Setting GCR to '2' (to release cold reset) using devmem2 starts things moving again, shutting up this warning. 2. maybe as a result of the above problem, the wm9705 touchscreen driver doesn't reinitialize, causing loss of touchscreen. Unbinding and re-binding the driver restores the touchscreen, but with a very long lag between touching the screen and it being registered. I bring (2) up because I notice that the resume actions in (1) are deferred. Given that codecs have shared functions (such as touchscreens) need to access the codec from their own resume functions, how can this deferral be safe? Liam - this problem has been noticed on a NetbookPro (I'm sure you remember that platform...)

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Re: [alsa-devel] hda Regression: jack doesn't mute speaker
by Takashi Iwai 17 Apr '09

17 Apr '09

At Thu, 16 Apr 2009 09:23:25 +0200, Ortwin Glück wrote: > > Takashi Iwai wrote: > >>>> Also the "Front" > >>>> and "Surround" outputs are now both labeled "Speaker" (see attached images of kmix). > >>> What I/O are they actually corresponding to? Are they speaker outputs > >>> or line-outs, or anything else? > >> They correspond to the built-in speakers. > > > > So they are really front/surround 4.0-speaker outputs? > > Probably. They seem to control two distinct speakers inside the laptop. More I > don't know. In OS-X there is only a single volume control (a sacrifice to > usability I think. who would want to control individual volumes of these on a > laptop anyway). The problem is likely because hp_detect field isn't set on your machine because the headphone shares the same audio-out widget with the speaker out. Could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ce30b45..917bc5d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3076,6 +3076,11 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, unsigned int wid_caps; for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } nid = dac_nids[i]; if (!nid) continue; @@ -3119,11 +3124,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; - if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { - wid_caps = get_wcaps(codec, pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - } } } return 0;

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