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

[alsa-devel] microphone recording got lower or higher tune than standard - 1
by 刘伟丰 27 Dec '13

27 Dec '13

Hello, you all! sorry I have to repost the message by spliting in three ones as the original size is a bit too big to be passed. I am facing a strange problem on our arm-based box runnig ubuntu. firstly the following tests worked fine: arecord -f S16_LE -D plughw:0,0 -r 8000 | aplay -f S16_LE -D plughw:0,0 -r 8000 arecord -f S16_LE -D plughw:0,0 -r 16000 | aplay -f S16_LE -D plughw:0,0 -r 16000 an so on up to 32k,44k,48k., and for c1, c2 but using Mumble's Audio Tuninf Wizard to test speaker, I could only hear my voice with very lower tune than my normal, and when I use Linphone to do the same test I heard my voice with very higher tune than standard. in Mumble's test I used the following setting: Audio Tuninf Wizard: input Device This is the device your microphone is connectedto. System: ALSA Device: [hw:CARD=AMLCS42L52,DEV=] AML-CS42L52, Direct hardware device without any conversions or [hw:CARD=AMLCS42L52,DEV=] AML-CS42L52, Direct hardware device with all software conversions if I select any Pulse options I could hear nothing except choppy sounds. I am puzzled if the problem is in kernel driver or alsa or pulseaudio? any advice appreciated. #these are my alsa-info: ---------------------------------------------------------------------------------------------- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Fri Dec 27 02:58:19 UTC 2013 !!Linux Distribution !!------------------ !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: Firmware Version: !!Kernel Information !!------------------ Kernel release: 3.0.50-svn2 Operating System: GNU/Linux Architecture: armv7l Processor: armv7l SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.24 Library version: Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [AMLCS42L52 ]: SOC-Audio - AML-CS42L52 AML-CS42L52 !!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-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!--------------------------- !!ALSA Device nodes !!----------------- crw-rw---T+ 1 root audio 116, 0 Dec 27 10:57 /dev/snd/controlC0 crw-rw---T+ 1 root audio 116, 24 Dec 27 10:57 /dev/snd/pcmC0D0c crw-rw---T+ 1 root audio 116, 16 Dec 27 10:57 /dev/snd/pcmC0D0p crw-rw---T+ 1 root audio 116, 33 Dec 27 10:57 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Dec 27 10:57 . drwxr-xr-x 3 root root 140 Dec 27 10:57 .. lrwxrwxrwx 1 root root 12 Dec 27 10:57 platform-soc-audio -> ../controlC0 !!ALSA configuration files !!------------------------ !!User specific config file (~/.asoundrc) pcm.convert_mic { type route slave { pcm "plughw:0" channels 2 } ttable { 0 { 1 1.0 1 1.0 } } } pcm.!default { type asym playback { pcm "plughw:0" } capture { pcm "convert_mic" } } !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: AMLCS42L52 [AML-CS42L52], device 0: CS42L52 PCM cs42l52-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: AMLCS42L52 [AML-CS42L52], device 0: CS42L52 PCM cs42l52-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [AMLCS42L52] Card hw:0 'AMLCS42L52'/'AML-CS42L52' Mixer name : '' Components : '' Controls : 79 Simple ctrls : 76 Simple mixer control 'Master',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 176 Front Left: 0 [0%] [-102.00dB] Front Right: 0 [0%] [-102.00dB] Simple mixer control 'Headphone',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 152 Front Left: 0 [0%] [-99999.99dB] Front Right: 0 [0%] [-99999.99dB] Simple mixer control 'Headphone Analog Gain',0 Capabilities: enum Items: '0.3959' '0.4571' '0.5111' '0.6047' '0.7099' '0.8399' '1.000' '1.1430' Item0: '1.000' Simple mixer control 'Speaker',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 254 Front Left: 127 [50%] [-38.50dB] Front Right: 127 [50%] [-38.50dB] Simple mixer control 'Tone Control',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Bass Gain',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] [-94.50dB] Simple mixer control 'Treble Gain',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] [-94.50dB] Simple mixer control 'PCM Left Swap',0 Capabilities: enum Items: 'Left' 'LR 2' 'Right' Item0: 'Left' Simple mixer control 'PCM Mixer',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - -102 Front Left: 1 [-1%] [-101.50dB] Playback [on] Front Right: 1 [-1%] [-101.50dB] Playback [on] Simple mixer control 'PCM Right Swap',0 Capabilities: enum Items: 'Right' 'LR 2' 'Left' Item0: 'Right' Simple mixer control 'Beep',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 31 Mono: 31 [100%] [-86.50dB] Simple mixer control 'Beep Bass Corner Freq',0 Capabilities: enum Items: '50 Hz' '100 Hz' '200 Hz' '250 Hz' Item0: '50 Hz' Simple mixer control 'Beep Config',0 Capabilities: enum Items: 'Off' 'Single' 'Multiple' 'Continuous' Item0: 'Off' Simple mixer control 'Beep Mixer',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep Pitch',0 Capabilities: enum Items: 'C4' 'C5' 'D5' 'E5' 'F5' 'G5' 'A5' 'B5' 'C6' 'D6' 'E6' 'F6' 'G6' 'A6' 'B6' 'C7' Item0: 'C4' Simple mixer control 'Beep Treble Corner Freq',0 Capabilities: enum Items: '5 kHz' '7 kHz' '10 kHz' ' 15 kHz' Item0: '5 kHz' Simple mixer control 'Beep off Time',0 Capabilities: enum Items: '1.23 s' '2.58 s' '3.90 s' '5.20 s' '6.60 s' '8.05 s' '9.35 s' '10.80 s' Item0: '1.23 s' Simple mixer control 'Beep on Time',0 Capabilities: enum Items: '86 ms' '430 ms' '780 ms' '1.20 s' '1.50 s' '1.80 s' '2.20 s' '2.50 s' '2.80 s' '3.20 s' '3.50 s' '3.80 s' '4.20 s' '4.50 s' '4.80 s' '5.20 s' Item0: '86 ms' Simple mixer control 'ADC',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 32 Front Left: 8 [25%] [-88.00dB] Playback [off] Front Right: 8 [25%] [-88.00dB] Playback [off] Simple mixer control 'ADC Left Mux',0 Capabilities: enum Items: 'Input1A' 'Input2A' 'Input3A' 'Input4A' 'PGA Input Left' Item0: 'PGA Input Left' Simple mixer control 'ADC Left Swap',0 Capabilities: enum Items: 'Left' 'LR 2' 'Right' Item0: 'Left' Simple mixer control 'ADC Mixer',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - -102 Front Left: 1 [-1%] [-95.00dB] Playback [off] Front Right: 1 [-1%] [-95.00dB] Playback [off] Simple mixer control 'ADC Right Mux',0 Capabilities: enum Items: 'Input1B' 'Input2B' 'Input3B' 'Input4B' 'PGA Input Right' Item0: 'PGA Input Right' Simple mixer control 'ADC Right Swap',0 Capabilities: enum Items: 'Right' 'LR 2' 'Left' Item0: 'Right' Simple mixer control 'ALC',0 Capabilities: cswitch penum Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'ALC Attack Rate',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 63 Mono: 0 [0%] [-30.00dB] Simple mixer control 'ALC Max Threshold',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 1 [14%] [-24.00dB] Simple mixer control 'ALC Min Threshold',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 7 [100%] [0.00dB] Simple mixer control 'ALC Release Rate',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 63 Mono: 63 [100%] Simple mixer control 'ALC SR',0 Capabilities: cswitch penum Capture channels: Front Left - Front Right Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'ALC ZC',0 Capabilities: cswitch penum Capture channels: Front Left - Front Right Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Analog SR',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Analog ZC',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Batt Compensation',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Batt VP Monitor',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Batt VP ref',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control 'Bypass',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 120 Front Left: 40 [33%] [14.00dB] Front Right: 40 [33%] [14.00dB] Simple mixer control 'Bypass Left',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Bypass Mute',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Bypass Right',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Deemphasis',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Digital SR',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Digital ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'HP Left Amp',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'HP Right Amp',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'HPF',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Limiter',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Limiter Attack Rate',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 63 Mono: 0 [0%] [-30.00dB] Simple mixer control 'Limiter Cushion Threshold',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] [-30.00dB] Simple mixer control 'Limiter Max Threshold',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] [-30.00dB] Simple mixer control 'Limiter Release Rate',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 63 Mono: 63 [100%] Simple mixer control 'Limiter SR',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Limiter ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'MIC Bias Level',0 Capabilities: enum Items: '0.5 +VA' '0.6 +VA' '0.7 +VA' '0.8 +VA' '0.83 +VA' '0.91 +VA' Item0: '0.5 +VA' Simple mixer control 'MIC Gain',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 16 Front Left: 0 [0%] [16.00dB] Front Right: 0 [0%] [16.00dB] Simple mixer control 'MICA Mux',0 Capabilities: enum Items: 'Single' 'Differential' Item0: 'Single' Simple mixer control 'MICB Mux',0 Capabilities: enum Items: 'Single' 'Differential' Item0: 'Single' Simple mixer control 'NG Boost',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'NG Delay',0 Capabilities: enum Items: '50ms' '100ms' '150ms' '200ms' Item0: '50ms' Simple mixer control 'NG Enable',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'NG Threshold',0 Capabilities: enum Items: '-34dB' '-37dB' '-40dB' '-43dB' '-46dB' '-52dB' '-58dB' '-64dB' Item0: '-34dB' Simple mixer control 'NG Type',0 Capabilities: enum Items: 'Apply Specific' 'Apply All' Item0: 'Apply Specific' Simple mixer control 'Output Mux',0 Capabilities: enum Items: 'ADC' 'DSP' Item0: 'ADC' Simple mixer control 'PGA',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 8 Front Left: 0 [0%] [-6.00dB] Front Right: 0 [0%] [-6.00dB] Simple mixer control 'PGA AIN1L',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN1R',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN2L',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN2R',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN3L',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN3R',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN4L',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA AIN4R',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PGA MICA',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'PGA MICB',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'SPK Left Amp',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'SPK Right Amp',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] !!Alsactl output !!-------------- --startcollapse-- --endcollapse-- !!All Loaded Modules !!------------------ Module !!ALSA/HDA dmesg !!-------------- [ 0.607134@0] register tv module server ok [ 0.611486@0] HDMI DEBUG: amhdmitx_init [1771] [ 0.615831@0] HDMI DEBUG: amhdmitx_probe [1581] [ 0.620652@1] Set HDMI:Chip C [ 0.623486@1] HDMI DEBUG: HDMITX_M1B_Init [3329] [ 0.627882@1] HDMI DEBUG: HDMITX_M1B_Init [3332] [ 0.632497@1] HDMI DEBUG: hdmi_hw_init [1222] [ 0.632503@0] HDMI: get hdmi platform data [ 0.632508@0] HDMI 5V Power On [ 0.635069@0] ------------[ cut here ]------------ -- [ 0.786779@0] cfg80211: Calling CRDA to update world regulatory domain [ 0.787136@1] HDMI: get brd phy data [ 0.787142@1] hdmi phy setting [ 0.787211@1] HDMI: get brd phy data [ 0.787214@1] hdmi phy setting [ 0.788020@1] HDMI: reset intr mask [ 0.791243@1] Switched to NOHz mode on CPU #1 [ 0.806787@1] HDMI DEBUG: hdmi_task_handle [1216] [ 0.818857@0] Switched to NOHz mode on CPU #0 -- [ 6.413727@0] aml_hw_crypto initialization. [ 6.418996@0] ***Entered sound/soc/codecs/cs42l52.c:cs42l52_modinit [ 6.424078@0] ***Entered sound/soc/codecs/cs42l52.c:cs42l52_i2c_probe [ 6.430501@0] ***cs42l52 reset*** [ 6.464886@0] enter aml_dai_probe [ 6.465038@0] ***Entered sound/soc/aml/aml_m6_cs42l52.c:m6_codec_audio_probe [ 6.466738@0] ***Entered sound/soc/codecs/cs42l52.c:cs42l52_probe [ 6.495203@0] cs42l52 1-004a: Failed to add route MIC IN->Mic Bias -- [ 6.525362@0] asoc: cs42l52 <-> aml-dai0 mapping ok [ 6.532326@0] ALSA device list: [ 6.533331@0] #0: AML-CS42L52 -- [ 11.911917@0] [ 11.911952@0] HDMITX: HDCP disable [ 11.911982@0] HDMI: reset IP: 0x10 [ 11.911994@0] TV mode VGA_1280X720 selected. -- [ 11.912331@0] set ok [ 11.912345@0] HDMI: get current mode: VGA_1280x720 [ 11.912366@0] set mode VIC 4 (cd0,cs0,pm1,vd0,1) [ 11.912373@0] HDMI DEBUG: hdmi_hw_reset [1395] [ 11.912392@0] HDMI: get brd phy data [ 11.912398@0] hdmi phy setting [ 11.913597@0] HDMI: get brd phy data [ 11.913610@0] hdmi phy setting [ 11.913695@0] HDMI DEBUG: hdmitx_set_pll [2069] [ 11.913703@0] param->VIC:4 -- [ 11.917944@1] audio sample rate: 0 [ 11.917949@1] HDMI: reset audio N para [ 11.917969@1] PCM out to HDMI [ 11.921776@0] set_vout_mode[179] [ 11.945810@1] Time out: AIU_HDMI_CLK_DATA_CTRL [ 11.978499@1] Time out: AIU_HDMI_CLK_DATA_CTRL [ 11.978517@1] Enable audio spdif to HDMI [ 11.978659@1] HDMI: set audio param [ 12.134549@1] Ethernet Driver ioctl (8947) ---------------------------------------------------------------------------------------- and my /proc/asound/card0/xxx follows this mesage -1 with seq. message -2

1 0

[alsa-devel] need help with loopback driver
by Olivier Langlois 27 Dec '13

27 Dec '13

Hi, I am trying to split a pcm stream and send it to my sound card and send a downmixed/downsampled copy to the playback device of the loopback card. I am using speaker-test to test my config. the sound goes to the speakers and I am able to record the downmixed signal through the loopback capture device but I am seeing a bunch of error messages in the speaker-test output which makes me doubt that there is maybe something wrong in what I am doing. So I am asking help from the experts: My conf: pcm.loop_44100_16 { type plug slave { pcm "hw:1,0,0" format S16_LE rate 44100 } } pcm.loop_capture { type hw card 1 device 1 subdevice 0 } pcm.split { type plug slave.pcm { type multi slaves { a { channels 8 pcm "hdmi:0,0" } # hdmi output b { channels 2 pcm "loop_44100_16" } # loopback record } bindings { 0 { slave a channel 0 } # Front Left 1 { slave a channel 1 } # Front Right 2 { slave a channel 2 } # Rear Left 3 { slave a channel 3 } # Rear right 4 { slave a channel 4 } # center 5 { slave a channel 5 } # LFE 6 { slave a channel 6 } # Side Left 7 { slave a channel 7 } # Side right 8 { slave b channel 0 } # left 9 { slave b channel 1 } # right } } ttable [ [ 1 0 0 0 0 0 0 0 0.38 0 ] # Front left [ 0 1 0 0 0 0 0 0 0 0.38 ] # Front right [ 0 0 1 0 0 0 0 0 0.22 0 ] # Rear left [ 0 0 0 1 0 0 0 0 0 0.22 ] # Rear right [ 0 0 0 0 1 0 0 0 0.18 0.18 ] # center [ 0 0 0 0 0 1 0 0 0 0 ] # LFE [ 0 0 0 0 0 0 1 0 0.22 0 ] # Side left [ 0 0 0 0 0 0 0 1 0 0.22 ] # Side right ] } $ speaker-test -Dsplit -c8 -r192000 -F S32_LE speaker-test 1.0.27.2 Playback device is split Stream parameters are 192000Hz, S32_LE, 8 channels Using 16 octaves of pink noise Rate set to 192000Hz (requested 192000Hz) Buffer size range from 296 to 131072 Period size range from 148 to 65536 Using max buffer size 131072 Periods = 4 was set period_size = 65536 was set buffer_size = 131072 0 - Front Left Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy 4 - Center Write error: -32,Broken pipe Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy 1 - Front Right Write error: -32,Broken pipe Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy 7 - Side Right Write error: -32,Broken pipe Write error: -32,Broken pipe Can't recovery from underrun, prepare failed: Device or resource busy lano1106@whippet2 ~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 11: HDMI 5 [HDMI 5] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 1: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Thank you,

1 3

Re: [alsa-devel] snd_pcm_drain() always fails
by Giridhara 26 Dec '13

26 Dec '13

Hynek Hanke,,, <hanke <at> volny.cz> writes: > > > > Or, what is the correct way to wait until the rest of the sound is > > > played when I'm using the poll() approach? > > Set the blocking mode before drain() is called (and return the mode back > > after it, if you want). > > Not possible. I'm waiting for other events (not comming from ALSA) too in the > poll(). > > > The better way might be setting sw_params to wakeup app after all frames are > > played and use the standard poll() call to wait in app. > > How do I do this or where can I read about it? Still, I need to receive other events > too while waiting for the sound to stop playing. > > Thank you, > Hynek Hanke > > ------------------------------------------------------- > SF.Net email is sponsored by: Discover Easy Linux Migration Strategies > from IBM. Find simple to follow Roadmaps, straightforward articles, > informative Webcasts and more! Get everything you need to get up to > speed, fast. http://ads.osdn.com/?ad_id=7477&alloc_id=16492&op=click > Hello Team, Any body successfully run the audio in the non blocking mode. I too have the same issues. Please suggest me. Any sample programs are great help for me. Thanks and Regards, Giri

1 0

[alsa-devel] [PATCH 1/2] Add __compat_aligned_s64 and __compat_aligned_u64
by H.J. Lu 25 Dec '13

25 Dec '13

X32 uses the 32-bit compat layer for 32-bit kernel interface, which is also used by ia32. But long long is 4-byte aligned for ia32 and 8-byte aligned for x32. When the long long field in a system call struct has different offsets between ia32 and x32 due to alignment padding, like struct snd_ctl_elem_value in include/uapi/sound/asound.h, we need to either add a special case for x32 or we can use 4-byte aligned long long in the system call struct so that x32 can share the same compat system cal with ia32. This patch adds __compat_aligned_s64 and __compat_aligned_u64 to include/uapi/asm-generic/int-ll64.h. By default, they are the same as __s64/__u64. It does the following things: 1. Add default __compat_aligned_s64/__compat_aligned_u64 to include/uapi/asm-generic/int-ll64.h. 2. Add arch/x86/include/asm/types.h for kernel internal use, which is the same as before, i.e., <asm-generic/types.h>. 3. Change arch/x86/include/uapi/asm/types.h not to use the default include/uapi/asm-generic/int-ll64.h. Instead, it has a modified copy of include/uapi/asm-generic/int-ll64.h with 4-byte aligned __compat_aligned_s64 and __compat_aligned_u64 for x32 and ia32. When building x86 kernel, nothing is changed. The difference is x32 can use 4-byte aligned long long for system calls so that it can share the same 32-bit compat system call with ia32. Signed-off-by: H.J. Lu <hjl.tools(a)gmail.com> --- arch/x86/include/asm/types.h | 6 ++++++ arch/x86/include/uapi/asm/types.h | 35 ++++++++++++++++++++++++++++++++++- include/uapi/asm-generic/int-ll64.h | 3 +++ 3 files changed, 43 insertions(+), 1 deletion(-) create mode 100644 arch/x86/include/asm/types.h diff --git a/arch/x86/include/asm/types.h b/arch/x86/include/asm/types.h new file mode 100644 index 0000000..8e8c23f --- /dev/null +++ b/arch/x86/include/asm/types.h @@ -0,0 +1,6 @@ +#ifndef _ASM_X86_TYPES_H +#define _ASM_X86_TYPES_H + +#include <asm-generic/types.h> + +#endif /* _ASM_X86_TYPES_H */ diff --git a/arch/x86/include/uapi/asm/types.h b/arch/x86/include/uapi/asm/types.h index 8e8c23f..7dac28b 100644 --- a/arch/x86/include/uapi/asm/types.h +++ b/arch/x86/include/uapi/asm/types.h @@ -1,6 +1,39 @@ #ifndef _ASM_X86_TYPES_H #define _ASM_X86_TYPES_H -#include <asm-generic/types.h> +#include <asm/bitsperlong.h> + +#ifndef __ASSEMBLY__ +/* + * __xx is ok: it doesn't pollute the POSIX namespace. Use these in the + * header files exported to user space + */ + +typedef __signed__ char __s8; +typedef unsigned char __u8; + +typedef __signed__ short __s16; +typedef unsigned short __u16; + +typedef __signed__ int __s32; +typedef unsigned int __u32; + +#ifdef __GNUC__ +__extension__ typedef __signed__ long long __s64; +__extension__ typedef unsigned long long __u64; +#else +typedef __signed__ long long __s64; +typedef unsigned long long __u64; +#endif + +#if defined(__i386__) || defined(__ILP32__) +typedef s64 __attribute__((aligned(4))) __compat_aligned_s64; +typedef u64 __attribute__((aligned(4))) __compat_aligned_u64; +#else +typedef __s64 __compat_aligned_s64; +typedef __u64 __compat_aligned_u64; +#endif + +#endif /* __ASSEMBLY__ */ #endif /* _ASM_X86_TYPES_H */ diff --git a/include/uapi/asm-generic/int-ll64.h b/include/uapi/asm-generic/int-ll64.h index a8658b2..3306917 100644 --- a/include/uapi/asm-generic/int-ll64.h +++ b/include/uapi/asm-generic/int-ll64.h @@ -33,6 +33,9 @@ typedef __signed__ long long __s64; typedef unsigned long long __u64; #endif +typedef __s64 __compat_aligned_s64; +typedef __u64 __compat_aligned_u64; + #endif /* __ASSEMBLY__ */ -- 1.8.1.4

1 1

[alsa-devel] [PATCH 2/2] Use __compat_aligned_s64 in uapi <sound/asound.h>
by H.J. Lu 25 Dec '13

25 Dec '13

The integer64 field in struct snd_ctl_elem_value has different offsets, depending the alignment of long long. This patch replaces long long with the newly introduced __compat_aligned_s64 so that x32 and ia32 can use the same compat system call for struct snd_ctl_elem_value. Signed-off-by: H.J. Lu <hjl.tools(a)gmail.com> --- include/uapi/sound/asound.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index e3983d5..ea6489c 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -877,8 +877,8 @@ struct snd_ctl_elem_value { long *value_ptr; /* obsoleted */ } integer; union { - long long value[64]; - long long *value_ptr; /* obsoleted */ + __compat_aligned_s64 value[64]; + __compat_aligned_s64 *value_ptr; /* obsoleted */ } integer64; union { unsigned int item[128]; -- 1.8.1.4

1 0

Re: [alsa-devel] [PATCH 2/2] ALSA: hda - Fix microphone for Sony VAIO Pro 13 (Haswell model)
by Hendrik-Jan Heins 25 Dec '13

25 Dec '13

Dear Takashi, David, it looks like this quirk also appears on the 11 Inch model of the Sony Vaio pro ( SVP112A1CM ). However, the device addess is different. Instead of: SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), I see address 0x90b5 (no clue where I should find the 0x104d, can you please send me a pointer how to find this one?). Would it be possible to please also incorporate the fix for the 11 inch Vaio? Thanks, Hendrik-Jan Heins

1 0

[alsa-devel] [PATCH] ALSA: Add params_width() helpers
by Mark Brown 24 Dec '13

24 Dec '13

From: Mark Brown <broonie(a)linaro.org> Add helpers for obtaining the width of a format directly from params since this is expected to become a common operation in ASoC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/pcm_params.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 37ae12e..6b1c78f 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -354,4 +354,16 @@ params_period_bytes(const struct snd_pcm_hw_params *p) params_channels(p)) / 8; } +static inline int +params_width(const struct snd_pcm_hw_params *p) +{ + return snd_pcm_format_width(params_format(p)); +} + +static inline int +params_physical_width(const struct snd_pcm_hw_params *p) +{ + return snd_pcm_format_physical_width(params_format(p)); +} + #endif /* __SOUND_PCM_PARAMS_H */ -- 1.8.5.2

2 1

Re: [alsa-devel] linux-next: Tree for Dec 23 (sound/soc/soc-io.c)
by Randy Dunlap 24 Dec '13

24 Dec '13

On 12/22/13 21:19, Stephen Rothwell wrote: > Hi all, > > There will be no linux-next releases between Dec 25 and Jan 5 inclusive - > unless I get extremely bored. > > Changes since 20131220: > on x86_64: sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x244ca): undefined reference to `regmap_init_i2c' CONFIG_REGMAP=y CONFIG_REGMAP_I2C=m Full randconfig file is attached. -- ~Randy

1 0

[alsa-devel] [PATCH] ALSA: Add params_width() helper
by Mark Brown 23 Dec '13

23 Dec '13

From: Mark Brown <broonie(a)linaro.org> Add a helper for obtaining the width of a format directly from params since this is expected to become a common operation in ASoC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Just starting on the ASoC changes to use this; I'm not 100% sure on the lack of prefixing for the name but it's consistent with the other uses so... include/sound/pcm_params.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 37ae12e..47b03b4 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -354,4 +354,10 @@ params_period_bytes(const struct snd_pcm_hw_params *p) params_channels(p)) / 8; } +static inline unsigned int +params_width(const struct snd_pcm_hw_params *p) +{ + return snd_pcm_format_width(params_format(p)); +} + #endif /* __SOUND_PCM_PARAMS_H */ -- 1.8.5.2

2 1

[alsa-devel] ALSA: hiface: Fix typo in 352800 rate definition
by Michael Trimarchi 23 Dec '13

23 Dec '13

The Vaughan device support the 352800 rate and not the 352000 Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Antonio Ospite <ao2(a)amarulasolutions.com> Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/usb/hiface/pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index c21a3df..2c44139 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -110,7 +110,7 @@ static const struct snd_pcm_hardware pcm_hw = { #define HIFACE_RATE_96000 0x4a #define HIFACE_RATE_176400 0x40 #define HIFACE_RATE_192000 0x48 -#define HIFACE_RATE_352000 0x58 +#define HIFACE_RATE_352800 0x58 #define HIFACE_RATE_384000 0x68 static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) @@ -141,8 +141,8 @@ static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) case 192000: rate_value = HIFACE_RATE_192000; break; - case 352000: - rate_value = HIFACE_RATE_352000; + case 352800: + rate_value = HIFACE_RATE_352800; break; case 384000: rate_value = HIFACE_RATE_384000; -- 1.8.1.2 -- | Michael Nazzareno Trimarchi Amarula Solutions BV | | COO - Founder Cruquiuskade 47 | | +31(0)851119172 Amsterdam 1018 AM NL |

2 1

[alsa-devel] HD-Audio/HDMI: avoid missing start of audio stream on audio reconfiguation or after S3
by Lin, Mengdong 23 Dec '13

23 Dec '13

Hi Takashi, We hope to continue discussing the workaround to avoid HDMI data missing when audio configuration changes (eg. stereo ->5.1 channel) or after S3. The basic idea is to insert silent audio stream playback on audio configuration changes. Please see updates below ... > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Thursday, December 19, 2013 9:27 PM > To: Lin, Mengdong > Cc: Wu, Fengguang; Li, Jocelyn; Girdwood, Liam R; Chehab, John > Subject: Re: Could you share some advice to check missing start of audio > stream in HDMI audio playback? > > At Thu, 19 Dec 2013 12:45:03 +0000, > Lin, Mengdong wrote: > > > > Many thanks, Takashi and Fengguang! > > > > Windows team met same issue. They also confirmed the latency for > monitor/AV receiver to reconfigure audio setup is not avoidable. > > > > And their workaround is to send silent audio data after each stream > reconfiguration, before playing back the valid audio stream. > > > > Do you feel this workaround should be implemented in driver or in > user-space? > > Hard to say. An easier for us would be user-space, but it can be done in > the driver (e.g. in the prepare callback), too, although the implementation > would become tricky. We'd need a static silence page and play it in > hda_intel.c as a special requirement in the prepare, then set the BLD back > to the normal buffer at the end of azx_pcm_prepare(). A bit complex, > but certainly doable. For Intel display engine, including Haswell/Ivybridg/Sandibridge, HW can generate silent audio data on driver request during a link underrun. So for these platforms, we can - enable this "HW internal fabrication" feature in i915 driver. HW supported 32kHz, N*44,1kHz, N*48kHz. - and add specified sleep at the end of azx_pcm_prepare() or generic_hdmi_playback_pcm_prepare() This can avoid audio driver to prepare silent data and transmit over HD-A link. Is it doable? > > Doing this in may generate extra latency to open a stream. And the > length of the steam depends on experience. > > Yes, it's purely depending on the receiver hardware. Can we add a kernel parameter to specify the length of the silent stream? It could be 0 by default, meaning no silent stream insertion and thus no extra latency for playback. > > > And silent streams in different formats need to be prepared since there > can be wrapped compressed stream for playback. > > So I'd like such a workaround in audio middleware in user-space. > > For the compress audio, it's a bit different problem, indeed. We need to check whether HW supports compressed data format. But at least we can apply this feature on PCM format. Thanks Mengdong > > Takashi > > > Thanks > > Mengdong > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Thursday, December 19, 2013 7:55 PM > > > To: Wu, Fengguang > > > Cc: Lin, Mengdong; Li, Jocelyn; Girdwood, Liam R; Chehab, John > > > Subject: Re: Could you share some advice to check missing start of > > > audio stream in HDMI audio playback? > > > > > > At Thu, 19 Dec 2013 15:38:40 +0800, > > > Fengguang Wu wrote: > > > > > > > > Mengdong, > > > > > > > > On Thu, Dec 19, 2013 at 02:30:33PM +0800, Lin, Mengdong wrote: > > > > > Hi Takashi/Fengguang > > > > > > > > > > > > > > > > > > > > Sorry to interrupt. Would you please share some advice to check > > > > > a HDMI audio issue below? > > > > > > > > > > > > > > > > > > > > For HDMI audio playback under Linux on Haswell/Valleyview, start > > > > > of an audio stream (2~4 seconds) may be missing before we can > > > > > hear the > > > sound, in two cases: > > > > > > > > > > - 1^st playback after S3. First several seconds of audio > > > stream can be > > > > > missing for some monitor, while other monitor does not have this > > > issue. > > > > > > > > > > - Switching from playing stereo PCM to 5.1 channel > PCM. > > > About first 2 > > > > > seconds of the 5.1 channel PCM cannot be heard from the AV > receiver. > > > > > > > > > > > > > > > > > > > > It seems some monitor/AV receiver can drop audio data before > > > > > playing sound. We wonder if we could improve in audio/gfx driver. > > > > > > > > > > Have you ever met similar issue on previous platforms? > > > > > > > > > > > > > > > > > > > > We’ll be appreciated if you could share us some tips to check > > > > > this > > > issue. > > > > > > > > > > > > > I once fixed a similar issue, hope it's useful information for you. > > > > > > > > commit 5779191e0efd851fb0d54698c13cb4f5325caca6 > > > > Author: Wu Fengguang <fengguang.wu(a)intel.com> > > > > AuthorDate: Wed Nov 18 12:38:06 2009 +0800 > > > > Commit: Takashi Iwai <tiwai(a)suse.de> > > > > CommitDate: Wed Nov 18 07:46:19 2009 +0100 > > > > > > > > ALSA: intelhdmi - sticky stream id and format > > > > > > > > We tracked down the first-0.5s-hdmi-audio-samples-lost > problem > > > to the > > > > AC_VERB_SET_CHANNEL_STREAMID command. It is suspected > that > > > many HDMI > > > > sinks need some time to adapt to the new state. > > > > > > > > The workaround is to avoid changing stream id/format > whenever > > > possible. > > > > Proposed by David. > > > > > > > > Signed-off-by: David Härdeman <david(a)hardeman.nu> > > > > Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > Yeah, sounds like a similar problem. > > > > > > In general, such a kind of drop is due to the receiver trying to > > > synchronize the stream at the beginning of the playback. So, the > > > patch above essentially keeps the audio widget untouched as long as > > > it's left unchanged. But, after S3, this can't be avoided. Ditto > > > for changing from stereo to 5.1. It's reconfiguring the audio setup > > > bits, thus it has to be reset. That said, the problem is likely > unavoidable in both situations. > > > > > > Of course, my wile guess might be wrong. I'd apply happily any fix > > > patches :) > > > > > > > > > Takashi

1 0

[alsa-devel] [PATCH 0/6] ASoC: fsl_sai: Fix some minor issues
by Nicolin Chen 21 Dec '13

21 Dec '13

This series patches fix some minor issues in the new fsl_sai driver. @Mark Brown I's so busy these days that I couldn't send these, which should have been review comments, in time. So some of them might be too minor to become a meaningful patch but I still sent them to see if you would like to apply. Thank you. Nicolin Chen (6): ASoC: fsl_sai: Keep symmetry for clk_enable() and clk_disable() ASoC: fsl_sai: Use snd_pcm_format_width() ASoC: fsl_sai: Drop useless channels check in hw_params() ASoC: fsl_sai: Drop useless ret in startup() ASoC: fsl_sai: Make dev_err information neater ASoC: fsl_sai: Sort local variable in general way sound/soc/fsl/fsl_sai.c | 71 ++++++++++++++++--------------------------------- 1 file changed, 23 insertions(+), 48 deletions(-) -- 1.8.4

3 8

[alsa-devel] [PATCH] ASoC: fsl-sai: Use snd_soc_dai_init_dma_data()
by Xiubo Li 21 Dec '13

21 Dec '13

Makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e..9c89f97 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -377,8 +377,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - cpu_dai->playback_dma_data = &sai->dma_params_tx; - cpu_dai->capture_dma_data = &sai->dma_params_rx; + snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, + &sai->dma_params_rx); snd_soc_dai_set_drvdata(cpu_dai, sai); -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: fsl-sai: Use devm_snd_dmaengine_pcm_register()
by Xiubo Li 21 Dec '13

21 Dec '13

Makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 15 +-------------- 1 file changed, 1 insertion(+), 14 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e..39cb9ee 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -454,19 +454,8 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); - if (ret) - return ret; - - return 0; -} - -static int fsl_sai_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - - return 0; } static const struct of_device_id fsl_sai_ids[] = { @@ -476,8 +465,6 @@ static const struct of_device_id fsl_sai_ids[] = { static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, - .remove = fsl_sai_remove, - .driver = { .name = "fsl-sai", .owner = THIS_MODULE, -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: fsl-sai: Remove fsl_sai_remove()
by Xiubo Li 21 Dec '13

21 Dec '13

There is no need of this function and makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e..1868ec3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -385,19 +385,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) -{ - cpu_dai->playback_dma_data = NULL; - cpu_dai->capture_dma_data = NULL; - - snd_soc_dai_set_drvdata(cpu_dai, NULL); - - return 0; -} - static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, - .remove = fsl_sai_dai_remove, .playback = { .channels_min = 1, .channels_max = 2, -- 1.8.4

2 1

[alsa-devel] [PATCH 23/38] sound: soc: add missing put_device call
by Levente Kurusa 21 Dec '13

21 Dec '13

This is required so that we give up the last reference to the device. Signed-off-by: Levente Kurusa <levex(a)linux.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87..21caa09 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -536,6 +536,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) err = device_register(&codec->ac97->dev); if (err < 0) { dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); + put_device(&codec->ac97->dev); codec->ac97->dev.bus = NULL; return err; } -- 1.8.3.1

2 1

[alsa-devel] [PATCH 1/1] ASoC: pcm1792a: Remove redundant of_match_ptr helper
by Sachin Kamat 21 Dec '13

21 Dec '13

'pcm1792a_of_match' is always compiled in. Hence the helper macro is not needed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 7146653a8e16..f4d9b769bcda 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -244,7 +244,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(pcm1792a_of_match), + .of_match_table = pcm1792a_of_match, }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/1] ASoC: sam9x5_wm8731: Remove redundant of_match_ptr helper
by Sachin Kamat 21 Dec '13

21 Dec '13

'sam9x5_wm8731_of_match' is always compiled in. Hence the helper macro is not needed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/atmel/sam9x5_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 7d6a9055874b..59df542afc7d 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -195,7 +195,7 @@ static struct platform_driver sam9x5_wm8731_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, - .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), + .of_match_table = sam9x5_wm8731_of_match, }, .probe = sam9x5_wm8731_driver_probe, .remove = sam9x5_wm8731_driver_remove, -- 1.7.9.5

2 1

[alsa-devel] snd_hda_intel and Beats Audio
by Dan McGhee 21 Dec '13

21 Dec '13

I have an HP Envy m6 Sleekbook model k010 with Beats Audio. My system is LFS-7.4 and BLFS-7.4. I have subscribed to this list in order to get the sound on my laptop working and also, hopefully, to provide you with info on what might need to be done in the hda_intel driver. If this list is not the appropriate one in which to do this, I will be more than happy to unsubscribe to it and get on the "users" list. In the hope that this situation is acceptable for this list--and please let me know if it's not--let me continue. Except for video replay, sound and printing my build is complete. I do not have any other sound applications or "helpers" installed, particularly PulseAudio, except gstreamer which is for dependencies. I'm using xcfe4 desktop. In my troubleshooting and research I discovered "hda-jack-retask." I installed it and have been "poking around" for awhile unsuccessfully. I guess I just don't understand the "pin out." This particular laptop has two speakers and a woofer. I ran "alsa-info.sh" and here is the section on the pins: > /sys/class/sound/hwC0D0/init_pin_configs: > 0x03 0x185600f0 > 0x05 0x585600f0 > 0x07 0x585600f0 > 0x09 0x585600f0 > 0x0b 0x585600f0 > 0x0d 0x585600f0 and > /sys/class/sound/hwC1D0/init_pin_configs: > 0x0a 0x03a11020 > 0x0b 0x0321101f > 0x0c 0x40f000f0 > 0x0d 0x92170110 > 0x0e 0x40f000f0 > 0x0f 0x40f000f0 > 0x10 0x40f000f0 > 0x11 0xd5a30130 > 0x1f 0x40f000f0 If you need the results of the whole script, I will run it again and provide the link. I just don't understand the mapping for the pins to be able to "override" them in hda-jack-retask. Here is more, I hope helpful info. Output from "speaker-test"--whether run as root or an unprivileged user: > speaker-test 1.0.27.2 > > Playback device is default > Stream parameters are 48000Hz, S16_LE, 1 channels > Using 16 octaves of pink noise > Channels count (1) not available for playbacks: Invalid argument > Setting of hwparams failed: Invalid argument From "aplay -L > null > Discard all samples (playback) or generate zero samples (capture) > hdmi:CARD=HDMI,DEV=0 > HDA ATI HDMI, HDMI 0 > HDMI Audio Output > sysdefault:CARD=Generic > HD-Audio Generic, 92HD91BXX Analog > Default Audio Device > front:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > Front speakers > surround40:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 4.0 Surround output to Front and Rear speakers > surround41:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 4.1 Surround output to Front, Rear and Subwoofer speakers > surround50:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 5.0 Surround output to Front, Center and Rear speakers > surround51:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 5.1 Surround output to Front, Center, Rear and Subwoofer speakers > surround71:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers I will be grateful for any help. Also, once again, if this list is not the appropriate one, please let me know and I'll go to "users." Thanks, Dan

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable subwoofer on Dell Vostro 5460/5470
by David Henningsson 20 Dec '13

20 Dec '13

In this case, there are two DACs, and DAC 0x03 is mono. In order to make headphones and front speaker use DAC 0x02, and subwoofer use DAC 0x03, we artificially cut the connection from nodes 0x14 and 0x15 to node 0x03, so they can only use DAC 0x02. In addition, the 5460 and 5470 differs in the sense that 5470 also needs a headset mic patch, whereas 5460 has individual detection for headphone and headset mic. BugLink: https://bugs.launchpad.net/bugs/1211920 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I'm adding Kailang to cc on this one; hopefully he'll scream if this patch is doing anything dangerous, such as if there's a risk to blow up the subwoofer. sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++------ 1 file changed, 37 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dfe4080..16c0caa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3770,10 +3770,14 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, static void alc290_fixup_mono_speakers(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PRE_PROBE) - /* Remove DAC node 0x03, as it seems to be - giving mono output */ - snd_hda_override_wcaps(codec, 0x03, 0); + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* DAC node 0x03 is giving mono output. We therefore want to + make sure 0x14 (front speaker) and 0x15 (headphones) use the + stereo DAC, while leaving 0x17 (bass speaker) for node 0x03. */ + hda_nid_t conn1[2] = { 0x0c }; + snd_hda_override_conn_list(codec, 0x14, 1, conn1); + snd_hda_override_conn_list(codec, 0x15, 1, conn1); + } } #if IS_ENABLED(CONFIG_THINKPAD_ACPI) @@ -3913,6 +3917,9 @@ enum { ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, ALC290_FIXUP_MONO_SPEAKERS, + ALC290_FIXUP_MONO_SPEAKERS_HSJACK, + ALC290_FIXUP_SUBWOOFER, + ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, @@ -4235,9 +4242,31 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC290_FIXUP_SUBWOOFER_HSJACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x17, 0x90170112 }, /* subwoofer */ + { } + }, + .chained = true, + .chain_id = ALC290_FIXUP_MONO_SPEAKERS_HSJACK, + }, + [ALC290_FIXUP_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x17, 0x90170112 }, /* subwoofer */ + { } + }, + .chained = true, + .chain_id = ALC290_FIXUP_MONO_SPEAKERS, + }, [ALC290_FIXUP_MONO_SPEAKERS] = { .type = HDA_FIXUP_FUNC, .v.func = alc290_fixup_mono_speakers, + }, + [ALC290_FIXUP_MONO_SPEAKERS_HSJACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, .chained = true, .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, }, @@ -4282,6 +4311,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -4303,10 +4333,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Don't be too specific for conflicting boost ctl names
by Takashi Iwai 20 Dec '13

20 Dec '13

When a boost control influences on multiple input paths, we shouldn't pick up the name string specific to one input but rather choose a more generic name. A problem seen often is that a single mic boost controls both internal and external mics although the driver picks up the very first name randomly like "Internal Mic Boost". This should have been "Mic Boost", instead. This patch tries to correct that behavior: when a boost control is available, check whether it conflicts with other inputs. If it does, use a common string ("Mic", "Line") as long as possible, or take a generic name "Input". Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 44 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 40 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8cebdcfdcfdc..e3f934703aa2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3651,6 +3651,8 @@ static int parse_mic_boost(struct hda_codec *codec) struct hda_gen_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; struct hda_input_mux *imux = &spec->input_mux; + static const char *input_type_labels[3] = { "Mic", "Line", "Input" }; + int input_type_idxs[3] = {}; int i; if (!spec->num_adc_nids) @@ -3659,7 +3661,9 @@ static int parse_mic_boost(struct hda_codec *codec) for (i = 0; i < imux->num_items; i++) { struct nid_path *path; unsigned int val; - int idx; + int idx, type, j; + bool conflict; + const char *pfx; char boost_label[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; idx = imux->items[i].index; @@ -3678,11 +3682,43 @@ static int parse_mic_boost(struct hda_codec *codec) if (!val) continue; + /* check whether conflicting with other input paths */ + type = cfg->inputs[idx].type; + conflict = false; + for (j = i + 1; j < imux->num_items; j++) { + int idx_next = imux->items[i].index; + int type_next; + struct nid_path *path_next; + if (idx_next >= imux->num_items) + continue; + type_next = cfg->inputs[idx_next].type; + if (type_next > AUTO_PIN_LINE_IN) + continue; + path_next = get_input_path(codec, 0, j); + if (!path_next) + continue; + /* conflicting value? */ + if (look_for_boost_amp(codec, path_next) == val) { + conflict = true; + if (type != type_next) { + type = 2; /* generic input */ + break; + } + } + } + /* create a boost control */ + if (!conflict) { + pfx = spec->input_labels[idx]; + idx = spec->input_label_idxs[idx]; + } else { + pfx = input_type_labels[type]; + idx = input_type_idxs[type]++; + } + snprintf(boost_label, sizeof(boost_label), - "%s Boost Volume", spec->input_labels[idx]); - if (!add_control(spec, HDA_CTL_WIDGET_VOL, boost_label, - spec->input_label_idxs[idx], val)) + "%s Boost Volume", pfx); + if (!add_control(spec, HDA_CTL_WIDGET_VOL, boost_label, idx, val)) return -ENOMEM; path->ctls[NID_PATH_BOOST_CTL] = val; -- 1.8.5

3 7

[alsa-devel] [PATCH] ASoC: Add resource managed devm_snd_soc_register_codec()
by Xiubo Li 20 Dec '13

20 Dec '13

Adds a resource managed version of snd_soc_register_codec() which makes it possible to simplify the remove function as well as the error path for codec drivers. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-devres.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 41 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..60d6c67 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -378,6 +378,9 @@ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); +int devm_snd_soc_register_codec(struct device *dev, + const struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 7ac745d..3c8552f 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -125,3 +125,41 @@ int devm_snd_dmaengine_pcm_register(struct device *dev, EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); #endif + +static void devm_codec_release(struct device *dev, void *res) +{ + snd_soc_unregister_codec(*(struct device **)res); +} + +/** + * devm_snd_soc_register_codec - resource managed codec registration + * @codec_drv: codec to register + * @dai_drv: DAI to register + * @num_dai: Number of DAIs + * + * Register a codec with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_codec(struct device *dev, + const struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_codec_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_codec(dev, codec_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_codec); -- 1.8.4

2 1

Re: [alsa-devel] How to interrupt snd_pcm_writei in blocking mode?
by Giridhara 20 Dec '13

20 Dec '13

Takashi Iwai <tiwai <at> suse.de> writes: > > At Thu, 30 Jun 2005 20:42:28 +0200, > Hynek Hanke wrote: > > > > > > Hynek Hanke wrote: > > > > > I'm running it in two separate threads. One does all the preparations and calls > > > > > snd_pcm_writei() in blocking mode. The other one then decides it wants to stop > > > > > playing, so it calls snd_pcm_drop(). This causes the device to switch from > > > > > state RUNNING to state SETUP, but the snd_pcm_writei() hangs forever and never > > > > > returns. Why is this so? > > > Well, but pcm_lib.c::snd_pcm_lib_write1() should check the state SETUP > > > and break the loop immediately. Ditto for snd_pcm_lib_read1(). > > > The quick hack is below. Please give a shot. > > > > I tried to apply the patch to the 2.6.11 kernel but after > > compilation/instalation, there is no difference. It still hangs. Logfiles say > > it was in the state RUNNING just before snd_pcm_drop() was called and got into > > state SETUP immediatelly after. The logfile from the other thread shows > > snd_pcm_writei() never returns (not even after the sound would surely be > > completed). > > Hmm, then still something is missing. I'll check on it later. > > > So, what can I do? Do I really have to use the non-blocking mode or is this > > supposed to work (if I take care about all the other thread related issues > > myself)? My colleague is reporting it's working for him under some different > > configuration. > > The non-blocking mode and poll() should work (hopefully :) > If not, let me know. > > > Can I enable some debugging messages or something to help you to see what's > > wrong? > > Check the code flow in snd_pcm_lib_write1(). You can see the places > my patch tried to fix. Basically, the thread should be woken up and > get out of the loop. But according to your report, it didn't work as > expected by some reason... > > Takashi > > ------------------------------------------------------- > SF.Net email is sponsored by: Discover Easy Linux Migration Strategies > from IBM. Find simple to follow Roadmaps, straightforward articles, > informative Webcasts and more! Get everything you need to get up to > speed, fast. http://ads.osdn.com/?ad_id=7477&alloc_id=16492&op=click > Hello Takashi, We have the same problem. snd_pcm_writei blocks when we switch between playback and record continuously for several times. Is this issue is fixed in any kernel versions. I am running kernel version 2.6.31 Alsa version 1.0.18 is it a issue of buffers we allocate using snd_pcm_hw_params_set_period_size_near whats the way to calculate this value and how should be the ideal value. I appreciate any help/suggestions in this regard. Thanks and Regards, Giri

1 0

[alsa-devel] [PATCH v3 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 20 Dec '13

20 Dec '13

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx51 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx51 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. v3 adds another fixup for the interrupt stats. The interrupt handler currently writes back a subset of the interrupt flags to clear the interrupts in the status register. This is not implemented on every processor. For example imx27 does not allow writebacks, imx53 does not allow writeback of RFRC and TFRC. The interrupt stats were disabled for imx by default. 'Fix interrupt mapping and release' enables it for mx53 again and changes the writeback flags. I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v3: - Rebase on latest topic/fsl - Added patch to cleanup probe function error handling - Fix interrupt stats for imx53 - Replace imx50-ssi by imx51-ssi Markus Pargmann (9): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt mapping and release ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx51-ssi arch/arm/boot/dts/imx51.dtsi | 3 +- arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 539 +++++++++++++++++++++++++++-------------- 5 files changed, 387 insertions(+), 189 deletions(-) -- 1.8.5.1

4 17

[alsa-devel] [UPDATE RFC PATCH 15/15] ACPI/thinkpad: Fix wrong <acpi/acpi.h> inclusion in Thinkpad ACPI users.
by Lv Zheng 20 Dec '13

20 Dec '13

CONFIG_ACPI dependent code should include <linux/acpi.h> instead of directly including <acpi/acpi.h>. This patch cleans up such wrong inclusions for Thinkpad ACPI users. Cc: Henrique de Moraes Holschuh <ibm-acpi(a)hmh.eng.br> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Cc: ibm-acpi-devel(a)lists.sourceforge.net Cc: platform-driver-x86(a)vger.kernel.org Signed-off-by: Lv Zheng <lv.zheng(a)intel.com> --- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3fbf288..6df4c35 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3243,7 +3243,7 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> -#include <acpi/acpi.h> +#include <linux/acpi.h> static int (*led_set_func)(int, bool); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34de5dc..5dca90e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3726,7 +3726,7 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> -#include <acpi/acpi.h> +#include <linux/acpi.h> static int (*led_set_func)(int, bool); -- 1.7.10

2 1

[alsa-devel] HD audio problem
by Russ Whitaker 20 Dec '13

20 Dec '13

Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

1 0

[alsa-devel] [PATCH] ASoC: imx-spdif: Use module_init() to handle platform_device_register()
by Nicolin Chen 20 Dec '13

20 Dec '13

This is a quick fix for the below two issues when building spdif as modules. 1) When modprobing modules in order: snd-soc-fsl-spdif -> snd-soc-imx-spdif -> snd-soc-spdif-tx/rx, we will fail to create imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe it again due to inappropriate condition of platform_driver_unregister() in probe(). 2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok", doing rmmod the imx-spdif would cause kernel Oops: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90() sysfs: can not remove 'dapm_widget', no directory Modules linked in: snd_soc_imx_spdif(-) snd_soc_fsl_spdif imx_pcm_dma fuse bnep rfcomm bluetooth imx_sdma imx_thermal imx2_wdt snd_soc_spdif_tx hid_cypress [last unloaded: imx_pcm_dma] CPU: 0 PID: 1301 Comm: rmmod Not tainted 3.13.0-rc4+ #387 Backtrace: [<c0012640>] (dump_backtrace) from [<c00127dc>] (show_stack+0x18/0x1c) r6:00000393 r5:c016bfa4 r4:00000000 r3:00000000 [<c00127c4>] (show_stack) from [<c0683a18>] (dump_stack+0x70/0x90) [<c06839a8>] (dump_stack) from [<c00244e0>] (warn_slowpath_common+0x74/0x94) r4:dab89d30 r3:00000000 [<c002446c>] (warn_slowpath_common) from [<c00245a4>] (warn_slowpath_fmt+0x38/0x40) r8:c689ce00 r7:00000000 r6:00000000 r5:c077c04c r4:dba2a318 [<c0024570>] (warn_slowpath_fmt) from [<c016bfa4>] (sysfs_hash_and_remove+0x84/0x90) r3:c077c04c r2:c06b3c38 [<c016bf20>] (sysfs_hash_and_remove) from [<c016a00c>] (sysfs_remove_file_ns+0x18/0x1c) r7:da916900 r6:00000000 r5:db2c900c r4:dba2a318 [<c0169ff4>] (sysfs_remove_file_ns) from [<c03687d4>] (device_remove_file+0x20/0x24) [<c03687b4>] (device_remove_file) from [<c04f532c>] (snd_soc_dapm_free+0x1c/0x22c) [<c04f5310>] (snd_soc_dapm_free) from [<c04f0c98>] (soc_remove_codec+0x34/0x98) r10:dabae59c r9:00000000 r8:c689ce00 r7:da916900 r6:00000000 r5:db2c900c r4:dba2a200 r3:00000000 [<c04f0c64>] (soc_remove_codec) from [<c04f3188>] (soc_remove_dai_links.clone.28+0x3c0/0x3ec) r4:00000000 r3:00000000 [<c04f2dc8>] (soc_remove_dai_links.clone.28) from [<c04f3248>] (snd_soc_unregister_card+0x94/0xc8) r10:d8435000 r9:db924204 r8:d8435000 r7:d079d480 r6:00000001 r5:00000608 r4:dabae4b8 [<c04f31b4>] (snd_soc_unregister_card) from [<c04ffa70>] (devm_card_release+0x14/0x18) r6:00000003 r5:db924010 r4:dab89e60 r3:c04ffa5c [<c04ffa5c>] (devm_card_release) from [<c036ec74>] (release_nodes+0x190/0x1f8) [<c036eae4>] (release_nodes) from [<c036ed94>] (devres_release_all+0x38/0x54) r10:00000000 r9:dab88000 r8:c000eb44 r7:bef87600 r6:db924044 r5:bf131014 r4:db924010 [<c036ed5c>] (devres_release_all) from [<c036b7a8>] (__device_release_driver+0x80/0xd4) r4:db924010 r3:bf12f000 [<c036b728>] (__device_release_driver) from [<c036bf28>] (driver_detach+0xbc/0xc0) r5:bf131014 r4:db924010 [<c036be6c>] (driver_detach) from [<c036b490>] (bus_remove_driver+0x54/0x98) r6:dab89f3c r5:bf131058 r4:bf131014 r3:db0c8000 [<c036b43c>] (bus_remove_driver) from [<c036c588>] (driver_unregister+0x30/0x4c) r4:bf131014 r3:c68a7a80 [<c036c558>] (driver_unregister) from [<c036d21c>] (platform_driver_unregister+0x14/0x18) r4:00000000 r3:bf12f2d0 [<c036d208>] (platform_driver_unregister) from [<bf12f2e4>] (imx_spdif_driver_exit+0x14/0xd30 [snd_soc_imx_spdif]) [<bf12f2d0>] (imx_spdif_driver_exit [snd_soc_imx_spdif]) from [<c008d764>] (SyS_delete_module+0x140/0x190) [<c008d624>] (SyS_delete_module) from [<c000e980>] (ret_fast_syscall+0x0/0x48) r7:00000081 r6:000120a8 r5:bef87600 r4:00000880 ---[ end trace 09423e64ab60df46 ]--- Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = dab9c000 [00000008] *pgd=2062f831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] SMP ARM Modules linked in: snd_soc_imx_spdif(-) snd_soc_fsl_spdif imx_pcm_dma fuse bnep rfcomm bluetooth imx_sdma imx_thermal imx2_wdt snd_soc_spdif_tx hid_cypress [last unloaded: imx_pcm_dma] CPU: 0 PID: 1301 Comm: rmmod Tainted: G W 3.13.0-rc4+ #387 task: db0c8000 ti: dab88000 task.ti: dab88000 PC is at soc_remove_codec+0x78/0x98 LR is at trace_hardirqs_on+0x14/0x18 pc : [<c04f0cdc>] lr : [<c0066514>] psr: 600f0013 sp : dab89dc8 ip : 00000020 fp : dab89ddc r10: dabae59c r9 : 00000000 r8 : c689ce00 r7 : da916900 r6 : 00000000 r5 : db2c900c r4 : dba2a200 r3 : 00000000 r2 : 00100100 r1 : dabac810 r0 : dabae5f8 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 2ab9c059 DAC: 00000015 Process rmmod (pid: 1301, stack limit = 0xdab88240) Stack: (0xdab89dc8 to 0xdab8a000) 9dc0: 00000000 00000000 dab89e24 dab89de0 c04f3188 c04f0c70 9de0: d84355bc dabae5a0 00200200 00100100 00200200 00100100 d079d480 dabae4b8 9e00: 00000608 00000001 d079d480 d8435000 db924204 d8435000 dab89e44 dab89e28 9e20: c04f3248 c04f2dd4 c04ffa5c dab89e60 db924010 00000003 dab89e54 dab89e48 9e40: c04ffa70 c04f31c0 dab89e94 dab89e58 c036ec74 c04ffa68 dab89e94 a00f0013 9e60: dabae400 d079d480 00000002 db924010 bf131014 db924044 bef87600 c000eb44 9e80: dab88000 00000000 dab89eac dab89e98 c036ed94 c036eaf0 bf12f000 db924010 9ea0: dab89ec4 dab89eb0 c036b7a8 c036ed68 db924010 bf131014 dab89ee4 dab89ec8 9ec0: c036bf28 c036b734 db0c8000 bf131014 bf131058 dab89f3c dab89efc dab89ee8 9ee0: c036b490 c036be78 c68a7a80 bf131014 dab89f14 dab89f00 c036c588 c036b448 9f00: bf12f2d0 00000000 dab89f24 dab89f18 c036d21c c036c564 dab89f34 dab89f28 9f20: bf12f2e4 c036d214 dab89fa4 dab89f38 c008d764 bf12f2dc 00000000 5f646e73 9f40: 5f636f73 5f786d69 69647073 bef80066 dab89f74 dab89f60 c005f0d4 c0069020 9f60: 0077a008 00000000 00000880 bef87600 000120a8 00000081 bf131058 00000880 9f80: dab89f84 00000000 00000880 bef87600 000120a8 00000081 00000000 dab89fa8 9fa0: c000e980 c008d630 00000880 bef87600 bef87600 00000880 00009778 bef875f4 9fc0: 00000880 bef87600 000120a8 00000081 00000001 000120bc 00000001 00000000 9fe0: b6f03840 bef875fc 00008f75 b6f0384c 800f0010 bef87600 00000000 00000000 Backtrace: [<c04f0c64>] (soc_remove_codec) from [<c04f3188>] (soc_remove_dai_links.clone.28+0x3c0/0x3ec) r4:00000000 r3:00000000 [<c04f2dc8>] (soc_remove_dai_links.clone.28) from [<c04f3248>] (snd_soc_unregister_card+0x94/0xc8) r10:d8435000 r9:db924204 r8:d8435000 r7:d079d480 r6:00000001 r5:00000608 r4:dabae4b8 [<c04f31b4>] (snd_soc_unregister_card) from [<c04ffa70>] (devm_card_release+0x14/0x18) r6:00000003 r5:db924010 r4:dab89e60 r3:c04ffa5c [<c04ffa5c>] (devm_card_release) from [<c036ec74>] (release_nodes+0x190/0x1f8) [<c036eae4>] (release_nodes) from [<c036ed94>] (devres_release_all+0x38/0x54) r10:00000000 r9:dab88000 r8:c000eb44 r7:bef87600 r6:db924044 r5:bf131014 r4:db924010 [<c036ed5c>] (devres_release_all) from [<c036b7a8>] (__device_release_driver+0x80/0xd4) r4:db924010 r3:bf12f000 [<c036b728>] (__device_release_driver) from [<c036bf28>] (driver_detach+0xbc/0xc0) r5:bf131014 r4:db924010 [<c036be6c>] (driver_detach) from [<c036b490>] (bus_remove_driver+0x54/0x98) r6:dab89f3c r5:bf131058 r4:bf131014 r3:db0c8000 [<c036b43c>] (bus_remove_driver) from [<c036c588>] (driver_unregister+0x30/0x4c) r4:bf131014 r3:c68a7a80 [<c036c558>] (driver_unregister) from [<c036d21c>] (platform_driver_unregister+0x14/0x18) r4:00000000 r3:bf12f2d0 [<c036d208>] (platform_driver_unregister) from [<bf12f2e4>] (imx_spdif_driver_exit+0x14/0xd30 [snd_soc_imx_spdif]) [<bf12f2d0>] (imx_spdif_driver_exit [snd_soc_imx_spdif]) from [<c008d764>] (SyS_delete_module+0x140/0x190) [<c008d624>] (SyS_delete_module) from [<c000e980>] (ret_fast_syscall+0x0/0x48) r7:00000081 r6:000120a8 r5:bef87600 r4:00000880 Code: e594100c e5842068 e584306c e5913080 (e5930008) ---[ end trace 09423e64ab60df47 ]--- Ideally, we should bypass the device_unregister() if getting EPROBE_DEFER, and find the root cause of "can not remove 'dapm_widget', no directory". But the root cause seems to be beyond the imx-spdif driver level, which might be related to the disordered resource releasing of the whole link. Thus this patch can provide a quick fix, if acceptable, to these two bugs so that the driver can finely work when being built as modules. And it's also safe enough to avoid duplicate regisitering/unregistering since we don't handle them in the probe() and remove() any more. At the meantime, we can trace the root cause of the Oops without pressure. Test report: root@freescale ~$ modprobe snd-soc-spdif-rx imx-spdif sound-spdif.10: dir-hifi <-> 2004000.spdif mapping ok root@freescale ~$ rmmod snd-soc-imx-spdif root@freescale ~$ modprobe snd-soc-imx-spdif imx-spdif sound-spdif.10: dir-hifi <-> 2004000.spdif mapping ok root@freescale ~$ rmmod snd-soc-imx-spdif root@freescale ~$ rmmod snd-soc-fsl-spdif root@freescale ~$ modprobe snd-soc-imx-spdif imx-spdif sound-spdif.10: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.10: snd_soc_register_card failed: -517 platform sound-spdif.10: Driver imx-spdif requests probe deferral root@freescale ~$ modprobe snd-soc-fsl-spdif imx-spdif sound-spdif.10: dir-hifi <-> 2004000.spdif mapping ok root@freescale ~$ rmmod snd-soc-imx-spdif root@freescale ~$ rmmod snd-soc-spdif-rx root@freescale ~$ modprobe snd-soc-imx-spdif imx-spdif sound-spdif.10: ASoC: CODEC spdif-dir not registered imx-spdif sound-spdif.10: snd_soc_register_card failed: -517 platform sound-spdif.10: Driver imx-spdif requests probe deferral root@freescale ~$ modprobe snd-soc-spdif-rx imx-spdif sound-spdif.10: dir-hifi <-> 2004000.spdif mapping ok Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 84 ++++++++++++++++++++++++----------------------- 1 file changed, 43 insertions(+), 41 deletions(-) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 980dd1f..c4b9894 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -16,8 +16,6 @@ struct imx_spdif_data { struct snd_soc_dai_link dai[2]; struct snd_soc_card card; - struct platform_device *txdev; - struct platform_device *rxdev; }; static int imx_spdif_audio_probe(struct platform_device *pdev) @@ -47,13 +45,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->dai[num_links].cpu_of_node = spdif_np; data->dai[num_links].platform_of_node = spdif_np; num_links++; - - data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); - if (IS_ERR(data->txdev)) { - ret = PTR_ERR(data->txdev); - dev_err(&pdev->dev, "register dit failed: %d\n", ret); - goto end; - } } if (of_property_read_bool(np, "spdif-in")) { @@ -64,18 +55,11 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->dai[num_links].cpu_of_node = spdif_np; data->dai[num_links].platform_of_node = spdif_np; num_links++; - - data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); - if (IS_ERR(data->rxdev)) { - ret = PTR_ERR(data->rxdev); - dev_err(&pdev->dev, "register dir failed: %d\n", ret); - goto error_dit; - } } if (!num_links) { dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); - goto error_dir; + goto end; } data->card.dev = &pdev->dev; @@ -84,24 +68,16 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto error_dir; + goto end; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); - goto error_dir; + goto end; } platform_set_drvdata(pdev, data); - goto end; - -error_dir: - if (data->rxdev) - platform_device_unregister(data->rxdev); -error_dit: - if (data->txdev) - platform_device_unregister(data->txdev); end: if (spdif_np) of_node_put(spdif_np); @@ -109,18 +85,6 @@ end: return ret; } -static int imx_spdif_audio_remove(struct platform_device *pdev) -{ - struct imx_spdif_data *data = platform_get_drvdata(pdev); - - if (data->rxdev) - platform_device_unregister(data->rxdev); - if (data->txdev) - platform_device_unregister(data->txdev); - - return 0; -} - static const struct of_device_id imx_spdif_dt_ids[] = { { .compatible = "fsl,imx-audio-spdif", }, { /* sentinel */ } @@ -134,10 +98,48 @@ static struct platform_driver imx_spdif_driver = { .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, - .remove = imx_spdif_audio_remove, }; -module_platform_driver(imx_spdif_driver); +static struct platform_device *txdev; +static struct platform_device *rxdev; + +static int __init imx_spdif_init(void) +{ + int ret; + + txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(txdev)) + return PTR_ERR(txdev); + + rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(rxdev)) { + ret = PTR_ERR(rxdev); + goto err; + } + + ret = platform_driver_register(&imx_spdif_driver); + if (ret) + goto err1; + + return 0; + +err1: + platform_device_unregister(rxdev); +err: + platform_device_unregister(txdev); + + return ret; +} +module_init(imx_spdif_init); + +static void __exit imx_spdif_exit(void) +{ + platform_driver_unregister(&imx_spdif_driver); + + platform_device_unregister(rxdev); + platform_device_unregister(txdev); +} +module_exit(imx_spdif_exit); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); -- 1.8.4

3 7

[alsa-devel] [RFC PATCH 15/15] ACPI/thinkpad: Fix wrong <acpi/acpi.h> inclusion in Thinkpad ACPI users.
by Lv Zheng 20 Dec '13

20 Dec '13

CONFIG_ACPI dependent code should include <linux/acpi.h> instead of directly including <acpi/acpi.h>. This patch cleans up such wrong inclusions for Thinkpad ACPI users. Cc: Henrique de Moraes Holschuh <ibm-acpi(a)hmh.eng.br> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Cc: ibm-acpi-devel(a)lists.sourceforge.net Cc: platform-driver-x86(a)vger.kernel.org Signed-off-by: Lv Zheng <lv.zheng(a)intel.com> --- drivers/platform/x86/thinkpad_acpi.c | 1 - include/linux/thinkpad_acpi.h | 2 ++ sound/pci/hda/patch_conexant.c | 1 - sound/pci/hda/patch_realtek.c | 1 - 4 files changed, 2 insertions(+), 3 deletions(-) diff --git a/drivers/platform/x86/thinkpad_acpi.c b/drivers/platform/x86/thinkpad_acpi.c index defb6af..c114d7c 100644 --- a/drivers/platform/x86/thinkpad_acpi.c +++ b/drivers/platform/x86/thinkpad_acpi.c @@ -76,7 +76,6 @@ #include <linux/dmi.h> #include <linux/jiffies.h> #include <linux/workqueue.h> -#include <linux/acpi.h> #include <linux/pci_ids.h> #include <linux/thinkpad_acpi.h> #include <sound/core.h> diff --git a/include/linux/thinkpad_acpi.h b/include/linux/thinkpad_acpi.h index 361de59..02928eb 100644 --- a/include/linux/thinkpad_acpi.h +++ b/include/linux/thinkpad_acpi.h @@ -1,6 +1,8 @@ #ifndef __THINKPAD_ACPI_H__ #define __THINKPAD_ACPI_H__ +#include <linux/acpi.h> + /* These two functions return 0 if success, or negative error code (e g -ENODEV if no led present) */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3fbf288..c2da489 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3243,7 +3243,6 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> -#include <acpi/acpi.h> static int (*led_set_func)(int, bool); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34de5dc..17306be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3726,7 +3726,6 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> -#include <acpi/acpi.h> static int (*led_set_func)(int, bool); -- 1.7.10.4

4 4

[alsa-devel] [PATCH] Add support for Focusrite Saffire 6 USB
by Eduard Gilmutdinov 20 Dec '13

20 Dec '13

Signed-off-by: Eduard Gilmutdinov <edgilmutdinov(a)gmail.com> --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 9 +++++---- 2 files changed, 45 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..3454780 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..e9a59e9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -662,8 +662,9 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) /* * Novation Twitch DJ controller + * Focusrite Novation Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ASoC: fsl-sai: Remove fsl_sai_remove()
by Xiubo Li 20 Dec '13

20 Dec '13

There is no need of this function and makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e..1868ec3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -385,19 +385,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) -{ - cpu_dai->playback_dma_data = NULL; - cpu_dai->capture_dma_data = NULL; - - snd_soc_dai_set_drvdata(cpu_dai, NULL); - - return 0; -} - static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, - .remove = fsl_sai_dai_remove, .playback = { .channels_min = 1, .channels_max = 2, -- 1.8.4

1 0

[alsa-devel] [PATCHv3 0/8] ALSA: Add SAI driver and enable SGT15000 codec
by Xiubo Li 20 Dec '13

20 Dec '13

This patch series is mostly Freescale's SAI SoC Digital Audio Interface driver implementation. And the implementation is only compatible with device tree definition. This patch series is based on linux-next and has been tested on Vybrid VF610 Tower board using device tree. Changed in v3: - Add big-endian mode support. - Revise some commit comments, coding styles,etc. - Keep header files ordered by alphabet,etc. - Add MODULE_ALIAS("platform:fsl-sai") for SAI driver. - Rename the files to be in a same pattern with the existed ones. - Remove some not useful functions or code. - Only enable the clock while the device is in active use. - Revise one error message according to Bill. Changed in v2: - Use default settings for the generic dmaengine PCM driver. - Separate receive and transmit setting in most functions, but some couldn't for the HW limitation. - Drop some not reduntant code. - Use devm_snd_soc_register_component() instead of snd_soc_register_component(). - Use devm_snd_soc_register_card() instead of devm_snd_soc_register_card(). - Adjust the code sentences sequence. - Make the namespacing consistent. - Rename CONFIG_SND_SOC_FSL_SGTL5000 to CONFIG_SND_SOC_FSL_SGTL5000_VF610. - Drop some meaningless lines. - Rename the binding document file. Added in v1: - Add SAI SoC Digital Audio Interface driver. - Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610. - Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board. - Add device tree bindings for Freescale SAI. - Revise the bugs about the sgt15000 codec. - Add SGT15000 based audio machine driver. - Enable SGT15000 codec based audio driver node for VF610. - Add device tree bindings for Freescale VF610 sound.

3 15

[alsa-devel] [PATCHv4 0/4] ALSA: Add Freescale SAI driver
by Li.Xiubo＠freescale.com 20 Dec '13

20 Dec '13

This patch series is mostly Freescale's SAI SoC Digital Audio Interface driver implementation. And the implementation is only compatible with device tree definition. This patch series is based on linux-next and has been tested on Vybrid VF610 Tower board using device tree. Changed in v4: - Remove the uneeded function fsl_sai_dai_remove(). - Use snd_soc_dai_init_dma_data(). - Adjust the Konfig file. Changed in v3: - Add big-endian mode support. - Revise some commit comments, coding styles,etc. - Keep header files ordered by alphabet,etc. - Add MODULE_ALIAS("platform:fsl-sai") for SAI driver. - Rename the files to be in a same pattern with the existed ones. - Remove some not useful functions or code. - Only enable the clock while the device is in active use. - Revise one error message according to Bill. Changed in v2: - Use default settings for the generic dmaengine PCM driver. - Separate receive and transmit setting in most functions, but some couldn't for the HW limitation. - Drop some not reduntant code. - Use devm_snd_soc_register_component() instead of snd_soc_register_component(). - Use devm_snd_soc_register_card() instead of devm_snd_soc_register_card(). - Adjust the code sentences sequence. - Make the namespacing consistent. - Rename CONFIG_SND_SOC_FSL_SGTL5000 to CONFIG_SND_SOC_FSL_SGTL5000_VF610. - Drop some meaningless lines. - Rename the binding document file. Added in v1: - Add SAI SoC Digital Audio Interface driver. - Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610. - Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board. - Add device tree bindings for Freescale SAI. - Revise the bugs about the sgt15000 codec. - Add SGT15000 based audio machine driver. - Enable SGT15000 codec based audio driver node for VF610. - Add device tree bindings for Freescale VF610 sound.

1 0

[alsa-devel] [RFC PATCH 1/2] ASoC: wm_adsp: Factor out ADSP2 boot proceedure
by Charles Keepax 19 Dec '13

19 Dec '13

Move the ADSP2 boot proceedure into a work structure in preparation for running it asynchronously with the rest of the audio path bring up. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 186 +++++++++++++++++++++++++------------------ sound/soc/codecs/wm_adsp.h | 5 +- 2 files changed, 112 insertions(+), 79 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fb0c678..4475790 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,107 +1497,136 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -int wm_adsp2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +void wm_adsp2_boot_work(struct work_struct *work) { - struct snd_soc_codec *codec = w->codec; - struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); - struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; - struct wm_coeff_ctl *ctl; - unsigned int val; + struct wm_adsp *dsp = container_of(work, + struct wm_adsp, + boot_work); int ret; + unsigned int val; - dsp->card = codec->card; + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret); + return; + } + val = (val & ARIZONA_SYSCLK_FREQ_MASK) + >> ARIZONA_SYSCLK_FREQ_SHIFT; - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* - * For simplicity set the DSP clock rate to be the - * SYSCLK rate rather than making it configurable. - */ - ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); - if (ret != 0) { - adsp_err(dsp, "Failed to read SYSCLK state: %d\n", - ret); - return ret; - } - val = (val & ARIZONA_SYSCLK_FREQ_MASK) - >> ARIZONA_SYSCLK_FREQ_SHIFT; + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", ret); + return; + } - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + if (dsp->dvfs) { + ret = regmap_read(dsp->regmap, + dsp->base + ADSP2_CLOCKING, &val); if (ret != 0) { - adsp_err(dsp, "Failed to set clock rate: %d\n", - ret); - return ret; + dev_err(dsp->dev, "Failed to read clocking: %d\n", ret); + return; } - if (dsp->dvfs) { - ret = regmap_read(dsp->regmap, - dsp->base + ADSP2_CLOCKING, &val); + if ((val & ADSP2_CLK_SEL_MASK) >= 3) { + ret = regulator_enable(dsp->dvfs); if (ret != 0) { dev_err(dsp->dev, - "Failed to read clocking: %d\n", ret); - return ret; + "Failed to enable supply: %d\n", + ret); + return; } - if ((val & ADSP2_CLK_SEL_MASK) >= 3) { - ret = regulator_enable(dsp->dvfs); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); - return ret; - } - - ret = regulator_set_voltage(dsp->dvfs, - 1800000, - 1800000); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to raise supply: %d\n", - ret); - return ret; - } + ret = regulator_set_voltage(dsp->dvfs, + 1800000, + 1800000); + if (ret != 0) { + dev_err(dsp->dev, + "Failed to raise supply: %d\n", + ret); + return; } } + } - ret = wm_adsp2_ena(dsp); - if (ret != 0) - return ret; + ret = wm_adsp2_ena(dsp); + if (ret != 0) + return; - ret = wm_adsp_load(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_setup_algs(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_setup_algs(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_load_coeff(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load_coeff(dsp); + if (ret != 0) + goto err; - /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp); - if (ret != 0) - goto err; + /* Initialize caches for enabled and unset controls */ + ret = wm_coeff_init_control_caches(dsp); + if (ret != 0) + goto err; - /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp); - if (ret != 0) - goto err; + /* Sync set controls */ + ret = wm_coeff_sync_controls(dsp); + if (ret != 0) + goto err; + + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA, + ADSP2_CORE_ENA); + if (ret != 0) + goto err; + + dsp->running = true; + + return; + +err: + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); +} - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); +int wm_adsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); + struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; + int ret; + + dsp->card = codec->card; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + schedule_work(&dsp->boot_work); + flush_work(&dsp->boot_work); + + if (!dsp->running) + return -EIO; + break; + + case SND_SOC_DAPM_POST_PMU: + if (!dsp->running) + return -EIO; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_START, + ADSP2_START); if (ret != 0) goto err; - - dsp->running = true; break; case SND_SOC_DAPM_PRE_PMD: @@ -1668,6 +1697,7 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); INIT_LIST_HEAD(&adsp->ctl_list); + INIT_WORK(&adsp->boot_work, wm_adsp2_boot_work); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d018dea..20b6b4b 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -59,6 +59,8 @@ struct wm_adsp { struct regulator *dvfs; struct list_head ctl_list; + + struct work_struct boot_work; }; #define WM_ADSP1(wname, num) \ @@ -67,7 +69,8 @@ struct wm_adsp { #define WM_ADSP2(wname, num) \ SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ - wm_adsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) + wm_adsp2_event, SND_SOC_DAPM_PRE_PMU | \ + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; -- 1.7.2.5

2 2

[alsa-devel] [PATCH 0/5] ASoC/MFD: twl6040: Remove codec driver local reg cache
by Peter Ujfalusi 19 Dec '13

19 Dec '13

Hi Mark, It was not straight forward task to remove the local reg cache from the twl6040 codec driver. I also needed to change the MFD part so it will take over the caching of registers via regmap. So far I have the following patches working on top of mainline kernel. Is this close enough of what you had in mind when asked me to look at this? twl4030 codec is going to be a bit more complicated AFAIK but it will come eventually. Regards, Peter --- Peter Ujfalusi (5): ASoC: twl6040: Rename twl6040_is_path_unmuted -> twl6040_can_write_to_chip ASoC: twl6040: Custom caching for sensitive DL1/2 path registers MFD: twl6040: reg_defaults support for regmap ASoC: twl6040: Remove register restore functionality ASoC: twl6040: Remove self managed local reg_cache support drivers/mfd/twl6040.c | 92 +++++++++++++++++- sound/soc/codecs/twl6040.c | 227 +++++++++++++-------------------------------- 2 files changed, 155 insertions(+), 164 deletions(-) -- 1.8.4.2

3 14

[alsa-devel] [PATCH 22/38] sound: ac97: add missing put_device call
by Levente Kurusa 19 Dec '13

19 Dec '13

This is required so that we give up the last reference to the device. Signed-off-by: Levente Kurusa <levex(a)linux.com> --- sound/pci/ac97/ac97_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bf578ba2..55186f8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1964,6 +1964,7 @@ static int snd_ac97_dev_register(struct snd_device *device) snd_ac97_get_short_name(ac97)); if ((err = device_register(&ac97->dev)) < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); + put_device(&ac97->dev); ac97->dev.bus = NULL; return err; } -- 1.8.3.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add warning texts when codec driver Kconfig doesn't match
by Takashi Iwai 19 Dec '13

19 Dec '13

When a Kconfig of a codec driver doesn't match with the controller (CONFIG_SND_HDA_INTEL), it'll result in the non-working automatic probing. Unfortunately kbuild can't give such a restriction, but at least, it's possible to show a warning if such a condition is found. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a patch on top of the previous two patches for making HD-audio codec kconfigs to tristate. sound/pci/hda/Kconfig | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index b0c0481c3339..0e53634dbbd8 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -93,6 +93,9 @@ config SND_HDA_CODEC_REALTEK Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_REALTEK=m + config SND_HDA_CODEC_ANALOG tristate "Build Analog Device HD-audio codec support" select SND_HDA_GENERIC @@ -100,6 +103,9 @@ config SND_HDA_CODEC_ANALOG Say Y or M here to include Analog Device HD-audio codec support in snd-hda-intel driver, such as AD1986A. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_ANALOG=m + config SND_HDA_CODEC_SIGMATEL tristate "Build IDT/Sigmatel HD-audio codec support" select SND_HDA_GENERIC @@ -107,6 +113,9 @@ config SND_HDA_CODEC_SIGMATEL Say Y or M here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_SIGMATEL=m + config SND_HDA_CODEC_VIA tristate "Build VIA HD-audio codec support" select SND_HDA_GENERIC @@ -114,6 +123,9 @@ config SND_HDA_CODEC_VIA Say Y or M here to include VIA HD-audio codec support in snd-hda-intel driver, such as VT1708. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_VIA=m + config SND_HDA_CODEC_HDMI tristate "Build HDMI/DisplayPort HD-audio codec support" help @@ -121,6 +133,9 @@ config SND_HDA_CODEC_HDMI support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_HDMI=m + config SND_HDA_I915 bool default y @@ -133,6 +148,9 @@ config SND_HDA_CODEC_CIRRUS Say Y or M here to include Cirrus Logic codec support in snd-hda-intel driver, such as CS4206. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CIRRUS=m + config SND_HDA_CODEC_CONEXANT tristate "Build Conexant HD-audio codec support" select SND_HDA_GENERIC @@ -140,6 +158,9 @@ config SND_HDA_CODEC_CONEXANT Say Y or M here to include Conexant HD-audio codec support in snd-hda-intel driver, such as CX20549. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CONEXANT=m + config SND_HDA_CODEC_CA0110 tristate "Build Creative CA0110-IBG codec support" select SND_HDA_GENERIC @@ -147,12 +168,18 @@ config SND_HDA_CODEC_CA0110 Say Y or M here to include Creative CA0110-IBG codec support in snd-hda-intel driver, found on some Creative X-Fi cards. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CA0110=m + config SND_HDA_CODEC_CA0132 tristate "Build Creative CA0132 codec support" help Say Y or M here to include Creative CA0132 codec support in snd-hda-intel driver. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CA0132=m + config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 @@ -172,18 +199,27 @@ config SND_HDA_CODEC_CMEDIA Say Y or M here to include C-Media HD-audio codec support in snd-hda-intel driver, such as CMI9880. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CMEDIA=m + config SND_HDA_CODEC_SI3054 tristate "Build Silicon Labs 3054 HD-modem codec support" help Say Y or M here to include Silicon Labs 3054 HD-modem codec (and compatibles) support in snd-hda-intel driver. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_SI3054=m + config SND_HDA_GENERIC tristate "Enable generic HD-audio codec parser" help Say Y or M here to enable the generic HD-audio codec parser in snd-hda-intel driver. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_GENERIC=m + config SND_HDA_POWER_SAVE_DEFAULT int "Default time-out for HD-audio power-save mode" depends on PM -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add a quirk for Plantronics Gamecom 780
by Takashi Iwai 19 Dec '13

19 Dec '13

Plantronics Gamecom 780 headset has a firmware problem, and when the FU 0x09 volume is changed, it results in either too loud or silence except for a very narrow range. This patch provides a workaround, ignoring the node, initialize the volume in a sane value and keep untouched. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65251 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 10 ++++++++++ sound/usb/quirks.c | 14 ++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index cc2dd1f0decb..32af6b741ef5 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -322,6 +322,12 @@ static struct usbmix_name_map hercules_usb51_map[] = { { 0 } /* terminator */ }; +/* Plantronics Gamecom 780 has a broken volume control, better to disable it */ +static struct usbmix_name_map gamecom780_map[] = { + { 9, NULL }, /* FU, speaker out */ + {} +}; + /* * Control map entries */ @@ -358,6 +364,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x046d, 0x09a4), .ignore_ctl_error = 1, }, + { /* Plantronics GameCom 780 */ + .id = USB_ID(0x047f, 0xc010), + .map = gamecom780_map, + }, { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede99dfd..ede753685491 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,6 +660,18 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } +/* quirk for Plantronics GameCom 780 with CM6302 chip */ +static int snd_usb_gamecon780_boot_quirk(struct usb_device *dev) +{ + /* set the initial volume and don't change; other values are either + * too loud or silent due to firmware bug (bko#65251) + */ + u8 buf[2] = { 0x74, 0xdc }; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + UAC_FU_VOLUME << 8, 9 << 8, buf, 2); +} + /* * Novation Twitch DJ controller */ @@ -986,6 +998,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_nativeinstruments_boot_quirk(dev); case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ return snd_usb_fasttrackpro_boot_quirk(dev); + case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */ + return snd_usb_gamecon780_boot_quirk(dev); } return 0; -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add support for Focusrite Saffire 6 USB
by Eduard Gilmutdinov 19 Dec '13

19 Dec '13

This patch add support for playback through external audio card Focusrite Saffire 6 USB. Capture and MIDI is not tested. Signed-off-by: Eduard Gilmutdinov <edgilmutdinov(a)gmail.com> --- diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..158beb4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..3ea9627 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,10 +660,11 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } -/* +/* * Novation Twitch DJ controller + * Focusrite(Novation) Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */

2 1

[alsa-devel] ASoC fixes for v3.13
by Mark Brown 19 Dec '13

19 Dec '13

The following changes since commit 319e2e3f63c348a9b66db4667efa73178e18b17d: Linux 3.13-rc4 (2013-12-15 12:31:33 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc4 for you to fetch changes up to 96b7fe0119b932ad25451d2b6357e727bbe6a309: Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/tegra', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus (2013-12-19 10:25:27 +0000) The diffstat is larger than the description since the pull request is against Linus' tree due the scheduling of your pull request last week compared to mine to you. ---------------------------------------------------------------- ASoC: Fixes for v3.13 The fixes here are all driver specific ones, none of which particularly stand out but all of which are useful to users of those drivers. ---------------------------------------------------------------- Bo Shen (3): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode ASoC: wm8904: fix DSP mode B configuration Charles Keepax (2): ASoC: wm5110: Correct HPOUT3 DAPM route typo ASoC: wm_adsp: Add small delay while polling DSP RAM start Jean-Francois Moine (1): ASoC: kirkwood: Fix the CPU DAI rates Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dma' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/tegra', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver Stephen Warren (2): ASoC: tegra: fix uninitialized variables in set_fmt ASoC: don't leak on error in snd_dmaengine_pcm_register sound/soc/atmel/atmel_ssc_dai.c | 30 ++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8962.c | 13 ++++++++++++ sound/soc/codecs/wm_adsp.c | 10 ++++++--- sound/soc/fsl/imx-wm8962.c | 2 -- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++----------- sound/soc/soc-generic-dmaengine-pcm.c | 38 +++++++++++++++++++++++++---------- sound/soc/soc-pcm.c | 5 +++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 ++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 13 files changed, 105 insertions(+), 45 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm5110: Add FSH for ISRCs
by Charles Keepax 19 Dec '13

19 Dec '13

Currently, the driver only supports configuration of the lower sample rate (FSL) on the ISRCs. With the higher rate being fixed a SYSCLK, this patch adds support for configuring the higher sample rate (FSH). Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 3 +++ 3 files changed, 20 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6bfd803..56d3ff5 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -498,6 +498,22 @@ const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { EXPORT_SYMBOL_GPL(arizona_rate_val); +const struct soc_enum arizona_isrc_fsh[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1, + ARIZONA_ISRC1_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_1, + ARIZONA_ISRC2_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_1, + ARIZONA_ISRC3_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; +EXPORT_SYMBOL_GPL(arizona_isrc_fsh); + const struct soc_enum arizona_isrc_fsl[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2, ARIZONA_ISRC1_FSL_SHIFT, 0xf, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 6641f3d..99a97c0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -186,6 +186,7 @@ extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; +extern const struct soc_enum arizona_isrc_fsh[]; extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 7fb8599..18c7e1f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -280,6 +280,9 @@ SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), +SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), +SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), +SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), -- 1.7.2.5

2 3

[alsa-devel] [PATCH] ASoC: simple-card: Use devm_snd_soc_register_card()
by Xiubo Li 19 Dec '13

19 Dec '13

Makes the code slightly shorter. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7a9b6b4..3d190d0 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -234,14 +234,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.num_links = 1; cinfo->snd_card.dev = &pdev->dev; - return snd_soc_register_card(&cinfo->snd_card); -} - -static int asoc_simple_card_remove(struct platform_device *pdev) -{ - struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; - - return snd_soc_unregister_card(&cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { @@ -257,7 +250,6 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, - .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); -- 1.8.4

3 2

[alsa-devel] [PATCH 1/2] mfd: wm5110: Expose DRE control registers
by Charles Keepax 19 Dec '13

19 Dec '13

Certain use-cases require the DRE to be disabled so expose registers necessary to control the DRE enables. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 2 ++ include/linux/mfd/arizona/registers.h | 4 ++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 338cfbe..abd6713 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -601,6 +601,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x0000043D, 0x0180 }, /* R1085 - DAC Digital Volume 6R */ { 0x0000043E, 0x0080 }, /* R1086 - DAC Volume Limit 6R */ { 0x0000043F, 0x0800 }, /* R1087 - Noise Gate Select 6R */ + { 0x00000440, 0x8FFF }, /* R1088 - DRE Enable */ { 0x00000450, 0x0000 }, /* R1104 - DAC AEC Control 1 */ { 0x00000458, 0x0000 }, /* R1112 - Noise Gate Control */ { 0x00000480, 0x0040 }, /* R1152 - Class W ANC Threshold 1 */ @@ -1631,6 +1632,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_DAC_DIGITAL_VOLUME_6R: case ARIZONA_DAC_VOLUME_LIMIT_6R: case ARIZONA_NOISE_GATE_SELECT_6R: + case ARIZONA_DRE_ENABLE: case ARIZONA_DAC_AEC_CONTROL_1: case ARIZONA_NOISE_GATE_CONTROL: case ARIZONA_PDM_SPK1_CTRL_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 8987814..22916c0 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -3207,6 +3207,10 @@ /* * R1088 (0x440) - DRE Enable */ +#define ARIZONA_DRE3R_ENA 0x0020 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_MASK 0x0020 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_SHIFT 5 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_WIDTH 1 /* DRE3R_ENA */ #define ARIZONA_DRE3L_ENA 0x0010 /* DRE3L_ENA */ #define ARIZONA_DRE3L_ENA_MASK 0x0010 /* DRE3L_ENA */ #define ARIZONA_DRE3L_ENA_SHIFT 4 /* DRE3L_ENA */ -- 1.7.2.5

3 3

[alsa-devel] [PATCH V2 1/2] ASoC: SPEAr: get rid of spear-pcm-audio struct device
by Stephen Warren 18 Dec '13

18 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Modify the SPEAr PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. This is also required as a pre-cursor to removing spear_pcm_request_chan(). Cc: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Rebased on a merge of topic/spear and topic/dma --- sound/soc/spear/spdif_in.c | 9 +++++++-- sound/soc/spear/spdif_out.c | 10 ++++++++-- sound/soc/spear/spear_pcm.c | 18 ++++-------------- sound/soc/spear/spear_pcm.h | 22 ++++++++++++++++++++++ 4 files changed, 41 insertions(+), 18 deletions(-) create mode 100644 sound/soc/spear/spear_pcm.h diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 21a8c954af1c..4627110f3441 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -24,6 +24,7 @@ #include <sound/spear_dma.h> #include <sound/spear_spdif.h> #include "spdif_in_regs.h" +#include "spear_pcm.h" struct spdif_in_params { u32 format; @@ -257,8 +258,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } static struct platform_driver spdif_in_driver = { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index b6ef6f78dc78..731a1e0cfeaa 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -22,6 +22,7 @@ #include <sound/spear_dma.h> #include <sound/spear_spdif.h> #include "spdif_out_regs.h" +#include "spear_pcm.h" struct spdif_out_params { u32 rate; @@ -280,6 +281,7 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; + int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -306,8 +308,12 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } #ifdef CONFIG_PM diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9a02141666ea..f288724961da 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -18,6 +18,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/spear_dma.h> +#include "spear_pcm.h" static const struct snd_pcm_hardware spear_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -47,26 +48,15 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { .prealloc_buffer_size = 16 * 1024, }; -static int spear_soc_platform_probe(struct platform_device *pdev) +int devm_spear_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &spear_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver spear_pcm_driver = { - .driver = { - .name = "spear-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = spear_soc_platform_probe, -}; - -module_platform_driver(spear_pcm_driver); +EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:spear-pcm-audio"); diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h new file mode 100644 index 000000000000..631e2aa1fb33 --- /dev/null +++ b/sound/soc/spear/spear_pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __SPEAR_PCM_H__ +#define __SPEAR_PCM_H__ + +int devm_spear_pcm_platform_register(struct device *dev); + +#endif -- 1.8.1.5

2 2

[alsa-devel] [PATCH V2 1/2] ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device
by Stephen Warren 18 Dec '13

18 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Modify the ep93xx PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. Cc: H Hartley Sweeten <hsweeten(a)visionengravers.com> Cc: Ryan Mallon <rmallon(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Rebased on a merge of topic/ep93xx and topic/dma --- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 8 ++++++++ sound/soc/cirrus/ep93xx-i2s.c | 8 ++++++++ sound/soc/cirrus/ep93xx-pcm.c | 19 +++++-------------- sound/soc/cirrus/ep93xx-pcm.h | 22 ++++++++++++++++++++++ sound/soc/cirrus/simone.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- 7 files changed, 46 insertions(+), 17 deletions(-) create mode 100644 sound/soc/cirrus/ep93xx-pcm.h diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index c43fb214558a..4f900efc437c 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -63,7 +63,7 @@ static struct snd_soc_ops edb93xx_ops = { static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index efa75b5086a4..cc5583da5380 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -24,6 +24,8 @@ #include <linux/platform_data/dma-ep93xx.h> +#include "ep93xx-pcm.h" + /* * Per channel (1-4) registers. */ @@ -394,8 +396,14 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (ret) goto fail; + ret = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (ret) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail: ep93xx_ac97_info = NULL; snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index a57643d6402f..167728a73da2 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -30,6 +30,8 @@ #include <mach/ep93xx-regs.h> #include <linux/platform_data/dma-ep93xx.h> +#include "ep93xx-pcm.h" + #define EP93XX_I2S_TXCLKCFG 0x00 #define EP93XX_I2S_RXCLKCFG 0x04 #define EP93XX_I2S_GLCTRL 0x0C @@ -405,8 +407,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) if (err) goto fail_put_lrclk; + err = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (err) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index fdb8b8feb4ed..198c5405f2ce 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -23,6 +23,8 @@ #include <linux/platform_data/dma-ep93xx.h> +#include "ep93xx-pcm.h" + static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -76,27 +78,16 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .prealloc_buffer_size = 131072, }; -static int ep93xx_soc_platform_probe(struct platform_device *pdev) +int devm_ep93xx_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &ep93xx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver ep93xx_pcm_driver = { - .driver = { - .name = "ep93xx-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = ep93xx_soc_platform_probe, -}; - -module_platform_driver(ep93xx_pcm_driver); +EXPORT_SYMBOL_GPL(devm_ep93xx_pcm_platform_register); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h new file mode 100644 index 000000000000..b7a12a2fae9c --- /dev/null +++ b/sound/soc/cirrus/ep93xx-pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __EP93XX_PCM_H__ +#define __EP93XX_PCM_H__ + +int devm_ep93xx_pcm_platform_register(struct device *dev); + +#endif diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c index 4d094d00c34a..822a19a89e74 100644 --- a/sound/soc/cirrus/simone.c +++ b/sound/soc/cirrus/simone.c @@ -27,7 +27,7 @@ static struct snd_soc_dai_link simone_dai = { .cpu_dai_name = "ep93xx-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-ac97", }; static struct snd_soc_card snd_soc_simone = { diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 69041074f2c1..29238a7476dd 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, -- 1.8.1.5

2 2

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: read chip revision for once
by Shawn Guo 18 Dec '13

18 Dec '13

Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe() reads it out from register, so that we can use it in sgtl5000_enable_regulators() with no need to read register again. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..bd291d2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -115,6 +115,7 @@ struct sgtl5000_priv { struct ldo_regulator *ldo; struct regmap *regmap; struct clk *mclk; + int revision; }; /* @@ -1300,9 +1301,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; int ret; - int rev; int i; int external_vddd = 0; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); @@ -1332,14 +1331,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) * workaround for revision 0x11 and later, * roll back to use internal LDO */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - goto err_regulator_disable; - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - - if (external_vddd && rev >= 0x11) { + if (external_vddd && sgtl5000->revision >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1362,9 +1354,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return 0; -err_regulator_disable: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); err_regulator_free: regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1566,6 +1555,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + sgtl5000->revision = rev; i2c_set_clientdata(client, sgtl5000); -- 1.7.9.5

3 16

[alsa-devel] [PATCH 1/2] ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate
by Takashi Iwai 18 Dec '13

18 Dec '13

So far, CONFIG_SND_HDA_CODEC_* kconfigs have been booleans due to historical reasons. The major reason was that the automatic codec driver probing wouldn't work if user sets a codec driver as a module while the controller driver as a built-in. And, another reason was to avoid exporting symbols of the helper codes when all drivers are built in. But, this sort of "kindness" rather confuses people in the end, especially makes the config refinement via localmodconfig unhappy. Also, a codec module would still work if you re-bind the controller driver via sysfs (although it's no automatic loading), so there might be a slight use case. That said, better to let people fallen into a pitfall than being too smart and restrict something. Let's make things straightforward: now all CONFIG_SND_HDA_CODEC_* become tristate, and all symbols exported unconditionally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 115 ++++++++++------------------------------------ sound/pci/hda/Makefile | 50 ++++++-------------- sound/pci/hda/hda_codec.h | 13 ------ 3 files changed, 37 insertions(+), 141 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 4cdd9ded4563..b0c0481c3339 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -87,126 +87,72 @@ config SND_HDA_PATCH_LOADER This option turns on hwdep and reconfig features automatically. config SND_HDA_CODEC_REALTEK - bool "Build Realtek HD-audio codec support" - default y + tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include Realtek HD-audio codec support in + Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-realtek. - This module is automatically loaded at probing. - config SND_HDA_CODEC_ANALOG - bool "Build Analog Device HD-audio codec support" - default y + tristate "Build Analog Device HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include Analog Device HD-audio codec support in + Say Y or M here to include Analog Device HD-audio codec support in snd-hda-intel driver, such as AD1986A. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-analog. - This module is automatically loaded at probing. - config SND_HDA_CODEC_SIGMATEL - bool "Build IDT/Sigmatel HD-audio codec support" - default y + tristate "Build IDT/Sigmatel HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include IDT (Sigmatel) HD-audio codec support in + Say Y or M here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-idt. - This module is automatically loaded at probing. - config SND_HDA_CODEC_VIA - bool "Build VIA HD-audio codec support" - default y + tristate "Build VIA HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include VIA HD-audio codec support in + Say Y or M here to include VIA HD-audio codec support in snd-hda-intel driver, such as VT1708. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-via. - This module is automatically loaded at probing. - config SND_HDA_CODEC_HDMI - bool "Build HDMI/DisplayPort HD-audio codec support" - default y + tristate "Build HDMI/DisplayPort HD-audio codec support" help - Say Y here to include HDMI and DisplayPort HD-audio codec + Say Y or M here to include HDMI and DisplayPort HD-audio codec support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-hdmi. - This module is automatically loaded at probing. - config SND_HDA_I915 bool default y depends on DRM_I915 config SND_HDA_CODEC_CIRRUS - bool "Build Cirrus Logic codec support" - default y + tristate "Build Cirrus Logic codec support" select SND_HDA_GENERIC help - Say Y here to include Cirrus Logic codec support in + Say Y or M here to include Cirrus Logic codec support in snd-hda-intel driver, such as CS4206. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-cirrus. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CONEXANT - bool "Build Conexant HD-audio codec support" - default y + tristate "Build Conexant HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include Conexant HD-audio codec support in + Say Y or M here to include Conexant HD-audio codec support in snd-hda-intel driver, such as CX20549. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-conexant. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CA0110 - bool "Build Creative CA0110-IBG codec support" - default y + tristate "Build Creative CA0110-IBG codec support" select SND_HDA_GENERIC help - Say Y here to include Creative CA0110-IBG codec support in + Say Y or M here to include Creative CA0110-IBG codec support in snd-hda-intel driver, found on some Creative X-Fi cards. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-ca0110. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CA0132 - bool "Build Creative CA0132 codec support" - default y + tristate "Build Creative CA0132 codec support" help - Say Y here to include Creative CA0132 codec support in + Say Y or M here to include Creative CA0132 codec support in snd-hda-intel driver. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-ca0132. - This module is automatically loaded at probing. - config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 @@ -220,35 +166,22 @@ config SND_HDA_CODEC_CA0132_DSP (ctefx.bin). config SND_HDA_CODEC_CMEDIA - bool "Build C-Media HD-audio codec support" - default y + tristate "Build C-Media HD-audio codec support" select SND_HDA_GENERIC help - Say Y here to include C-Media HD-audio codec support in + Say Y or M here to include C-Media HD-audio codec support in snd-hda-intel driver, such as CMI9880. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-cmedia. - This module is automatically loaded at probing. - config SND_HDA_CODEC_SI3054 - bool "Build Silicon Labs 3054 HD-modem codec support" - default y + tristate "Build Silicon Labs 3054 HD-modem codec support" help - Say Y here to include Silicon Labs 3054 HD-modem codec + Say Y or M here to include Silicon Labs 3054 HD-modem codec (and compatibles) support in snd-hda-intel driver. - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-si3054. - This module is automatically loaded at probing. - config SND_HDA_GENERIC - bool "Enable generic HD-audio codec parser" - default y + tristate "Enable generic HD-audio codec parser" help - Say Y here to enable the generic HD-audio codec parser + Say Y or M here to enable the generic HD-audio codec parser in snd-hda-intel driver. config SND_HDA_POWER_SAVE_DEFAULT diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 5a40c652df41..1fcb118e480a 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -27,43 +27,19 @@ snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o # common driver obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o -# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans) -ifdef CONFIG_SND_HDA_GENERIC -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-generic.o -endif -ifdef CONFIG_SND_HDA_CODEC_REALTEK -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o -endif -ifdef CONFIG_SND_HDA_CODEC_CMEDIA -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o -endif -ifdef CONFIG_SND_HDA_CODEC_ANALOG -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o -endif -ifdef CONFIG_SND_HDA_CODEC_SIGMATEL -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o -endif -ifdef CONFIG_SND_HDA_CODEC_SI3054 -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o -endif -ifdef CONFIG_SND_HDA_CODEC_CIRRUS -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o -endif -ifdef CONFIG_SND_HDA_CODEC_CA0110 -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o -endif -ifdef CONFIG_SND_HDA_CODEC_CA0132 -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o -endif -ifdef CONFIG_SND_HDA_CODEC_CONEXANT -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o -endif -ifdef CONFIG_SND_HDA_CODEC_VIA -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o -endif -ifdef CONFIG_SND_HDA_CODEC_HDMI -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-hdmi.o -endif +# codec drivers +obj-$(CONFIG_SND_HDA_GENERIC) += snd-hda-codec-generic.o +obj-$(CONFIG_SND_HDA_CODEC_REALTEK) += snd-hda-codec-realtek.o +obj-$(CONFIG_SND_HDA_CODEC_CMEDIA) += snd-hda-codec-cmedia.o +obj-$(CONFIG_SND_HDA_CODEC_ANALOG) += snd-hda-codec-analog.o +obj-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += snd-hda-codec-idt.o +obj-$(CONFIG_SND_HDA_CODEC_SI3054) += snd-hda-codec-si3054.o +obj-$(CONFIG_SND_HDA_CODEC_CIRRUS) += snd-hda-codec-cirrus.o +obj-$(CONFIG_SND_HDA_CODEC_CA0110) += snd-hda-codec-ca0110.o +obj-$(CONFIG_SND_HDA_CODEC_CA0132) += snd-hda-codec-ca0132.o +obj-$(CONFIG_SND_HDA_CODEC_CONEXANT) += snd-hda-codec-conexant.o +obj-$(CONFIG_SND_HDA_CODEC_VIA) += snd-hda-codec-via.o +obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 3c484c27048b..2b5d19e48a27 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -681,19 +681,6 @@ snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec, struct snd_dma_buffer *dmab) {} #endif -/* - * Codec modularization - */ - -/* Export symbols only for communication with codec drivers; - * When built in kernel, all HD-audio drivers are supposed to be statically - * linked to the kernel. Thus, the symbols don't have to (or shouldn't) be - * exported unless it's built as a module. - */ -#ifdef MODULE #define EXPORT_SYMBOL_HDA(sym) EXPORT_SYMBOL_GPL(sym) -#else -#define EXPORT_SYMBOL_HDA(sym) -#endif #endif /* __SOUND_HDA_CODEC_H */ -- 1.8.5

1 1

[alsa-devel] [RFC][PATCH 00/17] Enhancement for firewire-lib
by Takashi Sakamoto 18 Dec '13

18 Dec '13

This series of patch extend functionalities of current firewire-lib: - support for both in/out AMDTP stream - support for duplex stream synchronization (blocking mode only) - support for both MIDI/PCM stream - support for both in/out CMP - support for some AV/C general commands Additionally, at the last of this series, some quirks for Fireworks are added. I'm happy to receive your comments, especially for sort processing. Takashi Sakamoto (17): firewire-lib: Rename functions, structure, member for both direction firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse in future firewire-lib: Add support for AMDTP transmit stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: give syt value to handle_out_packet() firewire-lib: Add support for duplex streams with synchronization firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks chipset sound/firewire/amdtp.c | 1014 +++++++++++++++++++++++++++++++++++---------- sound/firewire/amdtp.h | 161 +++++-- sound/firewire/cmp.c | 217 +++++++--- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 149 +++++++ sound/firewire/fcp.h | 16 + sound/firewire/speakers.c | 94 ++--- 8 files changed, 1319 insertions(+), 392 deletions(-) -- 1.8.3.2

3 39

[alsa-devel] [RFC PATCH] ASoC: wm_adsp: Change ADSPs to be mixer widgets rather the PGAs
by Charles Keepax 18 Dec '13

18 Dec '13

PGAs often control the output from a chip and if the DSP is also marked as a PGA it may be powered up after the output has been enabled. This patch changes the ADSP2 cores to be marked as snd_soc_dapm_mixer widgets so they are powered up before any PGAs. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi, Was also considering if it would be worth adding an additional snd_soc_dapm_dsp id? That could sit between mixers and pgas, but I can't really see any obvious issue with treating the DSP as a mixer and it is a much simpler change. Although I am open to writing the other change if it is preferred? Thanks, Charles sound/soc/codecs/wm_adsp.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d018dea..795a3b2 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -66,7 +66,7 @@ struct wm_adsp { wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) #define WM_ADSP2(wname, num) \ - SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ + SND_SOC_DAPM_MIXER_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ wm_adsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; -- 1.7.2.5

2 6

[alsa-devel] [PATCH v3] ASoC: fsl: imx-wm8962: Grant hw_params/free() permission to control FLL
by Nicolin Chen 18 Dec '13

18 Dec '13

From: Nicolin Chen <b42378(a)freescale.com> Previously, we couldn't use hw_params() and hw_free() to open and close FLL becuase there might be a race between two simmultaneous substreams so the FLL configuration would be changed and accordingly mulfunction. Also it wouldn't make sense for bypass path feature of WM8962. So we adopted DAPM way to control it. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level and reconfigure FLL. But after we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() and hw_free() to control FLL will allow us to support flexible use cases, 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k16bit.wav' for example. Thus this patch mainly adds FLL configuration code to hw_params/hw_free() so as to enchance the sound card's capability. Meanwhile in order not to break the bypass path feature, we make both set_bias_level() and hw_xxx() ways coexist. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Added analogue bypass path check in hw_free() to protect bypass route. * Kept sample_rate and sample_format to global static. v2: * Added spinlock potection for boolean fll_locked in PATCH-2 sound/soc/fsl/imx-wm8962.c | 154 +++++++++++++++++++++++++++++++-------------- 1 file changed, 108 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc..d6aa854 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -17,6 +17,7 @@ #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -35,6 +36,8 @@ struct imx_wm8962_data { char platform_name[DAI_NAME_SIZE]; struct clk *codec_clk; unsigned int clk_frequency; + spinlock_t fll_lock; + bool fll_locked; }; struct imx_priv { @@ -51,18 +54,114 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { static int sample_rate = 44100; static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; +static int imx_wm8962_enable_fll(struct snd_soc_dai *codec_dai, u32 rate, + snd_pcm_format_t format) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned long flags; + u32 freq, ret; + + spin_lock_irqsave(&data->fll_lock, flags); + if (data->fll_locked) { + spin_unlock_irqrestore(&data->fll_lock, flags); + return 0; + } + + data->fll_locked = true; + spin_unlock_irqrestore(&data->fll_lock, flags); + + if (format == SNDRV_PCM_FORMAT_S24_LE) + freq = rate * 384; + else + freq = rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, freq); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + freq, SND_SOC_CLOCK_IN); + if (ret) + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + + return ret; +} + +static int imx_wm8962_disable_fll(struct snd_soc_dai *codec_dai) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned long flags; + int ret; + + spin_lock_irqsave(&data->fll_lock, flags); + if (!data->fll_locked) { + spin_unlock_irqrestore(&data->fll_lock, flags); + return 0; + } + + data->fll_locked = false; + spin_unlock_irqrestore(&data->fll_lock, flags); + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret) + dev_err(dev, "failed to stop FLL: %d\n", ret); + + return ret; +} static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + sample_rate = params_rate(params); sample_format = params_format(params); - return 0; + return imx_wm8962_enable_fll(codec_dai, sample_rate, sample_format); +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + u32 mask, bypass = 0; + + mask = WM8962_MIXINL_TO_HPMIXL_MASK | WM8962_MIXINR_TO_HPMIXL_MASK | + WM8962_IN4L_TO_HPMIXL_MASK | WM8962_IN4R_TO_HPMIXL_MASK; + bypass |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_1) & mask; + bypass |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_2) & mask; + bypass |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_1) & mask; + bypass |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_2) & mask; + + /* Don't diable FLL if running multi-substreams or analogue bypass */ + if (codec_dai->active != 1 || bypass) + return 0; + + return imx_wm8962_disable_fll(codec_dai); } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, }; static int imx_wm8962_set_bias_level(struct snd_soc_card *card, @@ -70,60 +169,20 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; if (dapm->dev != codec_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; - else - pll_out = sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + return imx_wm8962_enable_fll(codec_dai, sample_rate, + sample_format); break; case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + return imx_wm8962_disable_fll(codec_dai); break; default: @@ -225,6 +284,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } + data->fll_locked = false; + spin_lock_init(&data->fll_lock); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); -- 1.8.4

2 3

[alsa-devel] [PATCH v2 RESEND 1/2] ASOC: tegra: fix matching of AC97 components
by Lucas Stach 18 Dec '13

18 Dec '13

Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- v2: no changes CC: <alsa-devel(a)alsa-project.org> CC: Mark Brown <broonie(a)linaro.org> CC: Stephen Warren <swarren(a)wwwdotorg.org> --- sound/soc/tegra/tegra_wm9712.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e11963..45b5789 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- 1.8.3.1

3 10

[alsa-devel] [PATCH] ASoC: wm8904: fix DSP mode B configuration
by Bo Shen 18 Dec '13

18 Dec '13

When wm8904 work in DSP mode B, we still need to configure it to work in DSP mode. Or else, it will work in Right Justified mode. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3938fb1..53bbfac 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: - aif1 |= WM8904_AIF_LRCLK_INV; + aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Remove duplicate info message for DSP RAM ready
by Charles Keepax 18 Dec '13

18 Dec '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index de47ad7..456784f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1489,7 +1489,6 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) } adsp_dbg(dsp, "RAM ready after %d polls\n", count); - adsp_info(dsp, "RAM ready after %d polls\n", count); return 0; } -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: mmp-pcm: config pcm slave via generic dmaengine
by Qiao Zhou 18 Dec '13

18 Dec '13

use snd_dmaengine_pcm_prepare_slave_config to set slave config, and remove the max_burst_size = 4 hard code. select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 18 +++--------------- 2 files changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4db74a0..6473052 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,7 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM help Say Y if you want to add support for codecs attached to diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 7929e19..682ee52 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -67,27 +67,15 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config slave_config; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!dma_params) - return 0; - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + ret = + snd_dmaengine_pcm_prepare_slave_config(substream, params, + &slave_config); if (ret) return ret; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->addr; - slave_config.dst_maxburst = 4; - } else { - slave_config.src_addr = dma_params->addr; - slave_config.src_maxburst = 4; - } - ret = dmaengine_slave_config(chan, &slave_config); if (ret) return ret; -- 1.7.0.4

4 9

[alsa-devel] [RFC PATCH] ALSA: hda - Explicitly keep codec powered up in hdmi_present_sense
by David Henningsson 18 Dec '13

18 Dec '13

This should help us avoid the following mutex deadlock: [] mutex_lock+0x2a/0x50 [] hdmi_present_sense+0x53/0x3a0 [snd_hda_codec_hdmi] [] generic_hdmi_resume+0x5a/0x70 [snd_hda_codec_hdmi] [] hda_call_codec_resume+0xec/0x1d0 [snd_hda_codec] [] snd_hda_power_save+0x1e4/0x280 [snd_hda_codec] [] codec_exec_verb+0x5f/0x290 [snd_hda_codec] [] snd_hda_codec_read+0x5b/0x90 [snd_hda_codec] [] snd_hdmi_get_eld_size+0x1e/0x20 [snd_hda_codec_hdmi] [] snd_hdmi_get_eld+0x2c/0xd0 [snd_hda_codec_hdmi] [] hdmi_present_sense+0x9a/0x3a0 [snd_hda_codec_hdmi] [] hdmi_repoll_eld+0x34/0x50 [snd_hda_codec_hdmi] Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) So; I came across the above stack trace when testing the latest daily HDA driver. It only happened once, and jack polling was enabled while testing, if that matters. As you can see, the hdmi_present_sense is called recursively, and my patch doesn't really fix that; it should only make the consequences less severe (because now the recursion would be in the very beginning, i e, at snd_hda_power_up). I first thought just unlocking the mutex before calling pin_get_eld could do, but not sure if we really want to do that. Nevertheless, I haven't done much of the power management work anyway, so I'd like you to double check it and potentially come up with better ideas if you have any. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f5060fc..977db17 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1496,11 +1496,14 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * specification worked this way. Hence, we just ignore the data in * the unsolicited response to avoid custom WARs. */ - int present = snd_hda_pin_sense(codec, pin_nid); + int present; bool update_eld = false; bool eld_changed = false; bool ret; + snd_hda_power_up(codec); + present = snd_hda_pin_sense(codec, pin_nid); + mutex_lock(&per_pin->lock); pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (pin_eld->monitor_present) @@ -1573,6 +1576,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) jack->block_report = !ret; mutex_unlock(&per_pin->lock); + snd_hda_power_down(codec); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [V2 PATCH] ALSA: hda - Add Dell headset detection quirk for one more laptop model
by Hui Wang 18 Dec '13

18 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280640, no external microphone can be detected when plugging a 3-ring headset. Using ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix this problem. The codec (Vendor ID: 0x10ec0255) on the machine belongs to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3578f11..5b96698 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4309,6 +4309,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.8.1.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Add Dell headset detection quirk for one more laptop model
by Hui Wang 18 Dec '13

18 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280640, no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. The codec (Vendor ID: 0x10ec0255) on the machine belongs to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This patch was generated against the master branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3578f11..4d7c602 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4309,6 +4309,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.8.1.2

2 5

[alsa-devel] USB sound auto-suspend not working
by Sarah Sharp 18 Dec '13

18 Dec '13

Hi Oliver and Takashi, I've noticed that in the last couple kernel releases or so, I can't get USB webcams to suspend. It turns out that the USB sound interface is keeping the device active, even when the device is not playing sound. This goes back as far as 3.10, but I haven't tried older kernels. This is testing on Ubuntu 13.10. Commit 88a8516a2128a6d078a106ead48092240e8a138f "ALSA: usbaudio: implement USB autosuspend" went into kernel 2.6.39. The commit message says the device is prevented from suspending if the pcm or midi channel files are open. I plugged in a USB speaker, and ran lsof to see which files were open (output is attached). AFAICT, only the USB sound device's control files are open, and I don't have any midi files. Any ideas as to why USB sound auto-suspend isn't working? Sarah Sharp

4 4

Re: [alsa-devel] [PATCH 2/2] ASoC: sgtl5000: clean up sgtl5000_enable_regulators()
by Li.Xiubo＠freescale.com 18 Dec '13

18 Dec '13

> On Fri, Dec 13, 2013 at 02:43:03PM +0800, Shawn Guo wrote: > > Function sgtl5000_enable_regulators() is somehow odd in handling the > > optional external VDDD supply. The driver can only enable this supply > > on SGTL5000 chip before revision 0x11, and of course when this > > external VDDD is present. It currently does something like below. > > > > 1. Check if regulator_bulk_get() on VDDA, VDDIO and VDDD will fail. If > > it fails, VDDD must be absent and it falls on internal LDO by calling > > sgtl5000_replace_vddd_with_ldo(). Otherwise, VDDD is used. And in > > either case, regulator_bulk_enable() will be called to enable > > 3 supplies. > > > > 2. In case that SGTL5000 revision is later than 0x11, even if external > > VDDD is present, it has to roll back the 'enable' and 'get' calls > > with regulator_bulk_disable() and regulator_bulk_free(), and starts > > over again by calling sgtl5000_replace_vddd_with_ldo() and > > regulator_bulk_enable(). > > > > Such back and forth calls sequence is complicated and unnecessary. > > Also, since commit 4ddfebd (regulator: core: Provide a dummy regulator > > with full constraints), regulator_bulk_get() will always succeeds > > because of the dummy regulator. Thus the VDDD detection is broken. > > > > The patch changes the flow to something like the following, which > > should be more reasonable and clear, and also fix the VDDD detection > breakage. > > > > 1. Check if we're running a chip before revision 0x11, on which an > > external VDDD can possibly be an option. > > > > 2. If it is an early revision, call regulator_get_optional() to detect > > whether an external VDDD supply is available. > > > > 3. If external VDDD is present, call sgtl5000_replace_vddd_with_ldo() to > > update sgtl5000->supplies info. > > > > 4. Drop regulator_bulk_get() call in sgtl5000_replace_vddd_with_ldo(), > > and call it in sgtl5000_enable_regulators() no matter it's an > > external VDDD or internal LDO. > > > > 5. Call regulator_bulk_enable() to enable these 3 regulators. > > > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > > Apart from that separate issue we're discussing, do you have any comment on > this patch itself? Hi Shawn, Sorry for late. It's looks good to me. -- Xiubo

1 0

[alsa-devel] Add support for Focusrite Saffire 6 USB
by Eduard Gilmutdinov 17 Dec '13

17 Dec '13

Hello! I have external audio card Focusrite Saffire 6 USB. This device works with the Linux kernel if I add the following corrections: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..158beb4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..3ea9627 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,10 +660,11 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } -/* +/* * Novation Twitch DJ controller + * Focusrite(Novation) Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ Help me to add this corrections in the Linux kernel mainline. Thank you!

2 1

[alsa-devel] [PATCH][RESEND] ALSA: Add SNDRV_PCM_STATE_PAUSED case in wait_for_avail function
by JongHo Kim 17 Dec '13

17 Dec '13

When the process is sleeping at the SNDRV_PCM_STATE_PAUSED state from the wait_for_avail function, the sleep process will be woken by timeout(10 seconds). Even if the sleep process wake up by timeout, by this patch, the process will continue with sleep and wait for the other state. Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e03b46..a210467 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: Fix the CPU DAI rates
by Jean-Francois Moine 17 Dec '13

17 Dec '13

This patch fixes the rates declared in the CPU DAI parameters: - SNDRV_PCM_RATE_KNOT and the discrete rates SNDRV_PCM_RATE_xxx should not be used with SNDRV_PCM_RATE_CONTINUOUS, - SNDRV_PCM_RATE_CONTINUOUS asks for rate_min and rate_max, - the device may do streaming down to 5512Hz. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++----------- 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0b18f65..3920a5e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -473,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, @@ -494,17 +494,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: wm5110: Correct HPOUT3 DAPM route typo
by Charles Keepax 17 Dec '13

17 Dec '13

Reported-by: Kyung-Kwee Ryu <kyung-kwee.ryu(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9eb8517..e052ea5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1340,7 +1340,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AEC Loopback", "HPOUT3L", "OUT3L" }, { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, - { "HPOUT3R", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/4] ASoC: arizona: Use async writes
by Mark Brown 17 Dec '13

17 Dec '13

From: Mark Brown <broonie(a)linaro.org> Where possible write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/arizona.c | 163 +++++++++++++++++++++++++-------------------- 1 file changed, 89 insertions(+), 74 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eb9f5d4d8928..6bfd8031c0c9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -93,7 +93,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (!priv->spk_ena && manual_ena) { - snd_soc_write(codec, 0x4f5, 0x25a); + regmap_write_async(arizona->regmap, 0x4f5, 0x25a); priv->spk_ena_pending = true; } break; @@ -105,12 +105,13 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, return -EBUSY; } - snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, - 1 << w->shift, 1 << w->shift); + regmap_update_bits_async(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 1 << w->shift); if (priv->spk_ena_pending) { msleep(75); - snd_soc_write(codec, 0x4f5, 0xda); + regmap_write_async(arizona->regmap, 0x4f5, 0xda); priv->spk_ena_pending = false; priv->spk_ena++; } @@ -119,16 +120,19 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, if (manual_ena) { priv->spk_ena--; if (!priv->spk_ena) - snd_soc_write(codec, 0x4f5, 0x25a); + regmap_write_async(arizona->regmap, + 0x4f5, 0x25a); } - snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, - 1 << w->shift, 0); + regmap_update_bits_async(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 0); break; case SND_SOC_DAPM_POST_PMD: if (manual_ena) { if (!priv->spk_ena) - snd_soc_write(codec, 0x4f5, 0x0da); + regmap_write_async(arizona->regmap, + 0x4f5, 0x0da); } break; } @@ -687,6 +691,7 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, int event) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct arizona *arizona = priv->arizona; unsigned int mask = 1 << w->shift; unsigned int val; @@ -709,7 +714,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, if (priv->arizona->hpdet_magic) val = 0; - snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val); + regmap_update_bits_async(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1, + mask, val); return arizona_out_ev(w, kcontrol, event); } @@ -864,6 +870,8 @@ EXPORT_SYMBOL_GPL(arizona_set_sysclk); static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; int lrclk, bclk, mode, base; base = dai->driver->base; @@ -920,17 +928,19 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, - bclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, - ARIZONA_AIF1TX_LRCLK_INV | - ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, - ARIZONA_AIF1RX_LRCLK_INV | - ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, - ARIZONA_AIF1_FMT_MASK, mode); + regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | + ARIZONA_AIF1_BCLK_MSTR, + bclk); + regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); return 0; } @@ -1182,18 +1192,22 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, if (ret != 0) return ret; - snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, - ARIZONA_AIF1TX_BCPF_MASK, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, - ARIZONA_AIF1RX_BCPF_MASK, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, - ARIZONA_AIF1TX_WL_MASK | - ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, - ARIZONA_AIF1RX_WL_MASK | - ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); return 0; } @@ -1446,31 +1460,31 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, struct arizona_fll_cfg *cfg, int source, bool sync) { - regmap_update_bits(arizona->regmap, base + 3, - ARIZONA_FLL1_THETA_MASK, cfg->theta); - regmap_update_bits(arizona->regmap, base + 4, - ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); - regmap_update_bits(arizona->regmap, base + 5, - ARIZONA_FLL1_FRATIO_MASK, - cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); - regmap_update_bits(arizona->regmap, base + 6, - ARIZONA_FLL1_CLK_REF_DIV_MASK | - ARIZONA_FLL1_CLK_REF_SRC_MASK, - cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | - source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits_async(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits_async(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); if (sync) - regmap_update_bits(arizona->regmap, base + 0x7, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); else - regmap_update_bits(arizona->regmap, base + 0x9, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); - regmap_update_bits(arizona->regmap, base + 2, - ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, - ARIZONA_FLL1_CTRL_UPD | cfg->n); + regmap_update_bits_async(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); } static bool arizona_is_enabled_fll(struct arizona_fll *fll) @@ -1503,9 +1517,9 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits_async(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); @@ -1515,15 +1529,15 @@ static void arizona_enable_fll(struct arizona_fll *fll, use_sync = true; } } else if (fll->sync_src >= 0) { - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits_async(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, sync, fll->sync_src, false); - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); return; @@ -1534,11 +1548,12 @@ static void arizona_enable_fll(struct arizona_fll *fll, * sync source. */ if (use_sync && fll->sync_freq > 100000) - regmap_update_bits(arizona->regmap, fll->base + 0x17, - ARIZONA_FLL1_SYNC_BW, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); else - regmap_update_bits(arizona->regmap, fll->base + 0x17, - ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + regmap_update_bits_async(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, + ARIZONA_FLL1_SYNC_BW); if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); @@ -1546,14 +1561,14 @@ static void arizona_enable_fll(struct arizona_fll *fll, /* Clear any pending completions */ try_wait_for_completion(&fll->ok); - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, 0); - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, - ARIZONA_FLL1_SYNC_ENA); + regmap_update_bits_async(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); ret = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); @@ -1566,8 +1581,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, -- 1.8.5.1

2 5

[alsa-devel] [PATCH] ASoC: dmaengine: Add config parameter NULL check.
by Xiubo Li 17 Dec '13

17 Dec '13

Because the "ASoC: dmaengine-pcm: Provide default config" has provided us one defualt config of DMA. When using this, the config parameter of devm_snd_dmaengine_pcm_register() will be NULL, so here we need to have a check before using it. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 7483922..2a6c569 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -299,7 +299,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, !dev->of_node) return 0; - if (config->dma_dev) { + if (config && config->dma_dev) { /* * If this warning is seen, it probably means that your Linux * device structure does not match your HW device structure. @@ -317,7 +317,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = "rx-tx"; else name = dmaengine_pcm_dma_channel_names[i]; - if (config->chan_names[i]) + if (config && config->chan_names[i]) name = config->chan_names[i]; chan = dma_request_slave_channel_reason(dev, name); if (IS_ERR(chan)) { -- 1.8.4

3 2

[alsa-devel] [PATCH V2] ASOC:DAPM: update DPCM runtime when mixer/mux changes
by Nenghua Cao 17 Dec '13

17 Dec '13

From: Nenghua Cao <nhcao(a)marvell.com> V2 -> V1: Removes the flag and update DPCM always when mixer/mux changes, according to Takashi's and Mark's suggestion. >> Usually backward compatibility is concerned when something new breaks >> the existing ones. In your case, always updating DPCM would work, >> too, even without an extra flag; it's just suboptimal. >Right, and if we have separate versions then at some point we'll end up >having to define DPCM versions of everything which is going to get >tedious and error prone. V1 -> V0: refined patch, and add flag into soc_mixer_control and soc_enum instead of the alsa core, according to Takashi's suggestion. > You don't need to extend the ALSA core API for such a purpose. It's > ASoC-specific, so rather extend struct soc_mixer_control instead, if > such a new flag is really needed. Nenghua Cao (1): ASOC:DAPM: update DPCM runtime when mixer/mux changes sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 1 files changed, 24 insertions(+), 4 deletions(-)

3 6

[alsa-devel] [PATCH 1/2] ASoC: ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap()
by Lars-Peter Clausen 16 Dec '13

16 Dec '13

From: Fengguang Wu <fengguang.wu(a)intel.com> devm_request_and_ioremap() has been deprecated in favour of devm_ioremap_resource(). Fixes the following coccinelle warning: sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195 Generated by: coccinelle/api/devm_ioremap_resource.cocci Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/adi/axi-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 0822c77..7f91a86 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -192,9 +192,9 @@ static int axi_i2s_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, res); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base, &axi_i2s_regmap_config); -- 1.8.0

2 3

[alsa-devel] Volume issues: Logitech S-150 USB Digital Speaker / "usb_audio: Warning! Unlikely big volume range"
by Linus Lüssing 16 Dec '13

16 Dec '13

Hi! I'm currently trying to use a Logitech S-150 USB Digital Speaker. Unfortunately, the volume range is way off, i.e. only 1% and 2% settings in the alsamixer are actually usable for general playback. dmesg: ----- [ 9850.768093] usb 1-1: new full-speed USB device number 2 using uhci_hcd [ 9850.948103] usb 1-1: New USB device found, idVendor=1130, idProduct=1620 [ 9850.948123] usb 1-1: New USB device strings: Mfr=0, Product=2, SerialNumber=0 [ 9850.948138] usb 1-1: Product: USB AUDIO [ 9852.353857] hidraw: raw HID events driver (C) Jiri Kosina [ 9852.397038] usb_audio: Warning! Unlikely big volume range (=17152), cval->res is probably wrong. [ 9852.397054] usb_audio: [2] FU [PCM Playback Volume] ch = 2, val = 0/17152/1<6>[ 9852.399123] usbcore: registered new interface driver snd-usb-audio [ 9852.412386] usbcore: registered new interface driver usbhid [ 9852.412400] usbhid: USB HID core driver [ 9852.562358] input: USB AUDIO as /devices/pci0000:00/0000:00:1d.0/usb1/1-1/1-1:1.2/input/input12 [ 9852.563157] hid-generic 0003:1130:1620.0001: input,hidraw0: USB HID v1.10 Device [USB AUDIO ] on usb-0000:00:1d.0-1/input2 ----- lsusb -d 1130:1620 (for lsusb -vvv, see attachment) ----- Bus 001 Device 002: ID 1130:1620 Tenx Technology, Inc. ----- The detected vendor does not seem to match, so all in all looks like a kernel bug? If you need any additional information or someone testing a kernel patch, just let me know. Cheers, Linus PS: My current workaround is to use more fine grained settings via amixer.

2 1

[alsa-devel] [PATCH -next] ASoC: tegra20-ac97: add missing clk_disable_unprepare() on error path
by Wei Yongjun 16 Dec '13

16 Dec '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add the missing clk_disable_unprepare() before return from tegra20_ac97_platform_probe() in the error handling case. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/tegra/tegra20_ac97.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index d8b98d7..cf5e1cf 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -393,7 +393,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); if (ret) { dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, @@ -401,7 +401,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -417,6 +417,8 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_clk_disable_unprepare: + clk_disable_unprepare(ac97->clk_ac97); err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put:

3 2

[alsa-devel] [PATCH -next] ASoC: tegra20-ac97: fix error return code in tegra20_ac97_platform_probe()
by Wei Yongjun 16 Dec '13

16 Dec '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return -ENODEV instead of 0 when no GPIO supplied in tegra20_ac97_platform_probe(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/tegra/tegra20_ac97.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index d8b98d7..3ea9738 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -358,6 +358,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } } else { dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); + ret = -ENODEV; goto err_clk_put; } @@ -365,6 +366,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) "nvidia,codec-sync-gpio", 0); if (!gpio_is_valid(ac97->sync_gpio)) { dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); + ret = -ENODEV; goto err_clk_put; }

2 1

[alsa-devel] [PATCH 0/3]: tinycompress: header and rate fixes
by Vinod Koul 16 Dec '13

16 Dec '13

This patch series update the header file to verion 0.1.2. This brings the 64bit ABI compatiblity issue fix. Users have complianed about use of SNDRV_RATE_xxx for passing the sample rates as it doesnt scale to what decoder user. So time to junk this and use raw rates as is. Vinod Koul (3): tinycompress: update the header to kernel API v0.1.2 tinycompress: cplay: use the numeric sample rate tinycompress: remove usage of SNDRV_RATE_xxx cplay.c | 2 +- include/sound/compress_offload.h | 23 ++++++++++++++--------- include/sound/compress_params.h | 9 +++++++++ include/tinycompress/tinycompress.h | 20 -------------------- utils.c | 34 ---------------------------------- 5 files changed, 24 insertions(+), 64 deletions(-) -- ~Vinod

1 3

[alsa-devel] [PATCH] ALSA: compress: change the way sample rates are sent to kernel
by Vinod Koul 16 Dec '13

16 Dec '13

The usage of SNDRV_RATES is not effective as we can have rates like 12000 or some other ones used by decoders. This change the usage of this to use the raw Hz values to be sent to kernel Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/compress_params.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index 602dc6c..1114e38 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -324,7 +324,7 @@ union snd_codec_options { /** struct snd_codec_desc - description of codec capabilities * @max_ch: Maximum number of audio channels - * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this + * @sample_rates: Sampling rates in Hz, use values like 48000 for this * @bit_rate: Indexed array containing supported bit rates * @num_bitrates: Number of valid values in bit_rate array * @rate_control: value is specified by SND_RATECONTROLMODE defines. -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ALSA: Add SNDRV_PCM_STATE_PAUSED case in wait_for_avail function
by JongHo Kim 16 Dec '13

16 Dec '13

When the process is sleeping at the SNDRV_PCM_STATE_PAUSED state from the wait_for_avail function, the sleep process will be woken by timeout(10 seconds). Even if the sleep process wake up by timeout, by this patch, the process will continue with sleep and wait for the other state. Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e03b46..a210467 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", -- 1.7.9.5

2 1

[alsa-devel] While writing to speaker sometimes segmentation fault comes.
by ranjith vs 16 Dec '13

16 Dec '13

Hello, I am streaming audio from vlc,in the following formats sampling frequency 44100 stereo 16 bit.On the receiver side receiveing pcm frames and writing to speaker using alsa interface. But sometimes i'm getting segmentation fault in the snd_pcm_writei function. And one more thing is that eventhough I try to set the buffer size to 4096. It is setting it's own value.But i'm writing the buffer of length which I received from pcm_avail_upadte function. Thanks & Regards, Ranjith

2 3

Re: [alsa-devel] [PATCH]ASOC:DAPM: update DPCM runtime when mixer/mux changes
by anish singh 16 Dec '13

16 Dec '13

<This patch is dependent on the $subject so i have used $subject instead of different one> I can see only one usage of: snd_soc_dapm_mixer_update_power and zero user of: snd_soc_dapm_mux_update_power() And after this $subject patch I think both of these functions becomes redundant. It will help in reducing the code size if we can get rid of this dead code.We can also remove code from tlv320aic3x.c and just let it use the core implementation. grep -rniI "snd_soc_dapm_mixer_update_power" sound/ sound/soc/soc-dapm.c:2149: int snd_soc_dapm_mixer_update_power sound/soc/codecs/tlv320aic3x.c:174: snd_soc_dapm_mixer_update_power grep -rniI "snd_soc_dapm_mux_update_power" sound/ no users. I would send a patch if experts agree.

2 1

[alsa-devel] [PATCH] ALSA: compress: Fix 64bit ABI incompatibility
by Takashi Iwai 16 Dec '13

16 Dec '13

snd_pcm_uframes_t is defined as unsigned long so it would take different sizes depending on 32 or 64bit architectures. As we don't want this ABI incompatibility, and there is no real 64bit user yet, let's make it the fixed size with __u32. Also bump the protocol version number to 0.1.2. Acked-by: Vinod Koul <vinod.koul(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/compress_offload.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index d630163b9a2e..5759810e1c1b 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -67,8 +67,8 @@ struct snd_compr_params { struct snd_compr_tstamp { __u32 byte_offset; __u32 copied_total; - snd_pcm_uframes_t pcm_frames; - snd_pcm_uframes_t pcm_io_frames; + __u32 pcm_frames; + __u32 pcm_io_frames; __u32 sampling_rate; }; -- 1.8.5

2 2

[alsa-devel] [BUG] Race in spdif initialisation / ASoC oopses on module removal
by Russell King - ARM Linux 16 Dec '13

16 Dec '13

So, I have the following as modules, and Ubuntu being Ubuntu, it loads them in an apparantly random order on each boot. The order in which it loaded it this time was (most recent first): snd_soc_fsl_spdif imx_sdma imx_pcm_dma snd_soc_imx_spdif snd_soc_spdif_tx which produced these messages: imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-sdma 20ec000.sdma: initialized imx-sdma 20ec000.sdma: loaded firmware 1.1 Unloading and reloading snd_soc_imx_spdif then caused this: fsl-spdif-dai 2004000.spdif: Missing dma channel for stream: 0 fsl-spdif-dai 2004000.spdif: ASoC: pcm constructor failed: -22 imx-spdif sound-spdif.17: ASoC: can't create pcm S/PDIF PCM Playback :-22 imx-spdif sound-spdif.17: ASoC: failed to instantiate card -22 imx-spdif sound-spdif.17: snd_soc_register_card failed: -22 imx-spdif: probe of sound-spdif.17 failed with error -22 fsl-spdif-dai 2004000.spdif: Missing dma channel for stream: 0 fsl-spdif-dai 2004000.spdif: ASoC: pcm constructor failed: -22 imx-spdif sound-spdif.17: ASoC: can't create pcm S/PDIF PCM Playback :-22 imx-spdif sound-spdif.17: ASoC: failed to instantiate card -22 imx-spdif sound-spdif.17: snd_soc_register_card failed: -22 and unloading all of them (except for imx_sdma and snd_soc_spdif_tx) and reloading them in this order (most recent first): snd_soc_imx_spdif snd_soc_fsl_spdif imx_pcm_dma then produced: imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok Clearly something isn't right here. Unloading snd_soc_imx_spdif to then try and re-provoke the problem then caused this: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90() sysfs: can not remove 'dapm_widget', no directory Modules linked in: snd_soc_imx_spdif(-) snd_soc_fsl_spdif imx_pcm_dma fuse bnep rfcomm bluetooth imx_sdma imx_thermal imx2_wdt snd_soc_spdif_tx hid_cypress [last unloaded: imx_pcm_dma] CPU: 0 PID: 1301 Comm: rmmod Not tainted 3.13.0-rc4+ #387 Backtrace: [<c0012640>] (dump_backtrace) from [<c00127dc>] (show_stack+0x18/0x1c) r6:00000393 r5:c016bfa4 r4:00000000 r3:00000000 [<c00127c4>] (show_stack) from [<c0683a18>] (dump_stack+0x70/0x90) [<c06839a8>] (dump_stack) from [<c00244e0>] (warn_slowpath_common+0x74/0x94) r4:dab89d30 r3:00000000 [<c002446c>] (warn_slowpath_common) from [<c00245a4>] (warn_slowpath_fmt+0x38/0x40) r8:c689ce00 r7:00000000 r6:00000000 r5:c077c04c r4:dba2a318 [<c0024570>] (warn_slowpath_fmt) from [<c016bfa4>] (sysfs_hash_and_remove+0x84/0x90) r3:c077c04c r2:c06b3c38 [<c016bf20>] (sysfs_hash_and_remove) from [<c016a00c>] (sysfs_remove_file_ns+0x18/0x1c) r7:da916900 r6:00000000 r5:db2c900c r4:dba2a318 [<c0169ff4>] (sysfs_remove_file_ns) from [<c03687d4>] (device_remove_file+0x20/0x24) [<c03687b4>] (device_remove_file) from [<c04f532c>] (snd_soc_dapm_free+0x1c/0x22c) [<c04f5310>] (snd_soc_dapm_free) from [<c04f0c98>] (soc_remove_codec+0x34/0x98) r10:dabae59c r9:00000000 r8:c689ce00 r7:da916900 r6:00000000 r5:db2c900c r4:dba2a200 r3:00000000 [<c04f0c64>] (soc_remove_codec) from [<c04f3188>] (soc_remove_dai_links.clone.28+0x3c0/0x3ec) r4:00000000 r3:00000000 [<c04f2dc8>] (soc_remove_dai_links.clone.28) from [<c04f3248>] (snd_soc_unregister_card+0x94/0xc8) r10:d8435000 r9:db924204 r8:d8435000 r7:d079d480 r6:00000001 r5:00000608 r4:dabae4b8 [<c04f31b4>] (snd_soc_unregister_card) from [<c04ffa70>] (devm_card_release+0x14/0x18) r6:00000003 r5:db924010 r4:dab89e60 r3:c04ffa5c [<c04ffa5c>] (devm_card_release) from [<c036ec74>] (release_nodes+0x190/0x1f8) [<c036eae4>] (release_nodes) from [<c036ed94>] (devres_release_all+0x38/0x54) r10:00000000 r9:dab88000 r8:c000eb44 r7:bef87600 r6:db924044 r5:bf131014 r4:db924010 [<c036ed5c>] (devres_release_all) from [<c036b7a8>] (__device_release_driver+0x80/0xd4) r4:db924010 r3:bf12f000 [<c036b728>] (__device_release_driver) from [<c036bf28>] (driver_detach+0xbc/0xc0) r5:bf131014 r4:db924010 [<c036be6c>] (driver_detach) from [<c036b490>] (bus_remove_driver+0x54/0x98) r6:dab89f3c r5:bf131058 r4:bf131014 r3:db0c8000 [<c036b43c>] (bus_remove_driver) from [<c036c588>] (driver_unregister+0x30/0x4c) r4:bf131014 r3:c68a7a80 [<c036c558>] (driver_unregister) from [<c036d21c>] (platform_driver_unregister+0x14/0x18) r4:00000000 r3:bf12f2d0 [<c036d208>] (platform_driver_unregister) from [<bf12f2e4>] (imx_spdif_driver_exit+0x14/0xd30 [snd_soc_imx_spdif]) [<bf12f2d0>] (imx_spdif_driver_exit [snd_soc_imx_spdif]) from [<c008d764>] (SyS_delete_module+0x140/0x190) [<c008d624>] (SyS_delete_module) from [<c000e980>] (ret_fast_syscall+0x0/0x48) r7:00000081 r6:000120a8 r5:bef87600 r4:00000880 ---[ end trace 09423e64ab60df46 ]--- Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = dab9c000 [00000008] *pgd=2062f831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] SMP ARM Modules linked in: snd_soc_imx_spdif(-) snd_soc_fsl_spdif imx_pcm_dma fuse bnep rfcomm bluetooth imx_sdma imx_thermal imx2_wdt snd_soc_spdif_tx hid_cypress [last unloaded: imx_pcm_dma] CPU: 0 PID: 1301 Comm: rmmod Tainted: G W 3.13.0-rc4+ #387 task: db0c8000 ti: dab88000 task.ti: dab88000 PC is at soc_remove_codec+0x78/0x98 LR is at trace_hardirqs_on+0x14/0x18 pc : [<c04f0cdc>] lr : [<c0066514>] psr: 600f0013 sp : dab89dc8 ip : 00000020 fp : dab89ddc r10: dabae59c r9 : 00000000 r8 : c689ce00 r7 : da916900 r6 : 00000000 r5 : db2c900c r4 : dba2a200 r3 : 00000000 r2 : 00100100 r1 : dabac810 r0 : dabae5f8 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 2ab9c059 DAC: 00000015 Process rmmod (pid: 1301, stack limit = 0xdab88240) Stack: (0xdab89dc8 to 0xdab8a000) 9dc0: 00000000 00000000 dab89e24 dab89de0 c04f3188 c04f0c70 9de0: d84355bc dabae5a0 00200200 00100100 00200200 00100100 d079d480 dabae4b8 9e00: 00000608 00000001 d079d480 d8435000 db924204 d8435000 dab89e44 dab89e28 9e20: c04f3248 c04f2dd4 c04ffa5c dab89e60 db924010 00000003 dab89e54 dab89e48 9e40: c04ffa70 c04f31c0 dab89e94 dab89e58 c036ec74 c04ffa68 dab89e94 a00f0013 9e60: dabae400 d079d480 00000002 db924010 bf131014 db924044 bef87600 c000eb44 9e80: dab88000 00000000 dab89eac dab89e98 c036ed94 c036eaf0 bf12f000 db924010 9ea0: dab89ec4 dab89eb0 c036b7a8 c036ed68 db924010 bf131014 dab89ee4 dab89ec8 9ec0: c036bf28 c036b734 db0c8000 bf131014 bf131058 dab89f3c dab89efc dab89ee8 9ee0: c036b490 c036be78 c68a7a80 bf131014 dab89f14 dab89f00 c036c588 c036b448 9f00: bf12f2d0 00000000 dab89f24 dab89f18 c036d21c c036c564 dab89f34 dab89f28 9f20: bf12f2e4 c036d214 dab89fa4 dab89f38 c008d764 bf12f2dc 00000000 5f646e73 9f40: 5f636f73 5f786d69 69647073 bef80066 dab89f74 dab89f60 c005f0d4 c0069020 9f60: 0077a008 00000000 00000880 bef87600 000120a8 00000081 bf131058 00000880 9f80: dab89f84 00000000 00000880 bef87600 000120a8 00000081 00000000 dab89fa8 9fa0: c000e980 c008d630 00000880 bef87600 bef87600 00000880 00009778 bef875f4 9fc0: 00000880 bef87600 000120a8 00000081 00000001 000120bc 00000001 00000000 9fe0: b6f03840 bef875fc 00008f75 b6f0384c 800f0010 bef87600 00000000 00000000 Backtrace: [<c04f0c64>] (soc_remove_codec) from [<c04f3188>] (soc_remove_dai_links.clone.28+0x3c0/0x3ec) r4:00000000 r3:00000000 [<c04f2dc8>] (soc_remove_dai_links.clone.28) from [<c04f3248>] (snd_soc_unregister_card+0x94/0xc8) r10:d8435000 r9:db924204 r8:d8435000 r7:d079d480 r6:00000001 r5:00000608 r4:dabae4b8 [<c04f31b4>] (snd_soc_unregister_card) from [<c04ffa70>] (devm_card_release+0x14/0x18) r6:00000003 r5:db924010 r4:dab89e60 r3:c04ffa5c [<c04ffa5c>] (devm_card_release) from [<c036ec74>] (release_nodes+0x190/0x1f8) [<c036eae4>] (release_nodes) from [<c036ed94>] (devres_release_all+0x38/0x54) r10:00000000 r9:dab88000 r8:c000eb44 r7:bef87600 r6:db924044 r5:bf131014 r4:db924010 [<c036ed5c>] (devres_release_all) from [<c036b7a8>] (__device_release_driver+0x80/0xd4) r4:db924010 r3:bf12f000 [<c036b728>] (__device_release_driver) from [<c036bf28>] (driver_detach+0xbc/0xc0) r5:bf131014 r4:db924010 [<c036be6c>] (driver_detach) from [<c036b490>] (bus_remove_driver+0x54/0x98) r6:dab89f3c r5:bf131058 r4:bf131014 r3:db0c8000 [<c036b43c>] (bus_remove_driver) from [<c036c588>] (driver_unregister+0x30/0x4c) r4:bf131014 r3:c68a7a80 [<c036c558>] (driver_unregister) from [<c036d21c>] (platform_driver_unregister+0x14/0x18) r4:00000000 r3:bf12f2d0 [<c036d208>] (platform_driver_unregister) from [<bf12f2e4>] (imx_spdif_driver_exit+0x14/0xd30 [snd_soc_imx_spdif]) [<bf12f2d0>] (imx_spdif_driver_exit [snd_soc_imx_spdif]) from [<c008d764>] (SyS_delete_module+0x140/0x190) [<c008d624>] (SyS_delete_module) from [<c000e980>] (ret_fast_syscall+0x0/0x48) r7:00000081 r6:000120a8 r5:bef87600 r4:00000880 Code: e594100c e5842068 e584306c e5913080 (e5930008) ---[ end trace 09423e64ab60df47 ]---

2 1

[alsa-devel] Where to find recent fix for HD-Audio runtime PM?
by Lin, Mengdong 16 Dec '13

16 Dec '13

Hi Takashi, If convenient, would you please point out your previous patches to make HD-Audio runtime PM stable? I guess 'ALSA: hda - Refactor pm notification in hda_codec.c' is only one of them. We observed audio failure (dead communication) after S3 if runtime PM is enabled on Haswell with Kernel v3.9. We hope to back port your previous fix and further check this issue. Many thanks! Mengdong

2 2

[alsa-devel] [PATCH v2] ASoC: SGTL5000: Fix kernel failed while trying to get optional VDDD supply.
by Xiubo Li 16 Dec '13

16 Dec '13

The SGTL5000 requires 2 external power supplies: VDDA and VDDIO. An optional third external power supply VDDD may be provided externally to achieve lower power.If an external supply is not used for VDDD, the SGTL5000 driver will register it's own regulator device, and then provides the VDDD supply consumer, and now there will be two regulator devices exist, local regulator for VDDD and platform regulator for VDDIO, VDDA. ******************* * *---------|3.3V VDDIO * * * SGTL5000 codec *---x VDDD * * * *---------|3.3V VDDA ******************* If an external supply is not used for VDDD, in the DT or architecture-specific file, only "VDDA-supply" and "VDDIO-supply" properties will be presented. This caused the following kernel failed while trying to get the external VDDD supply before trying to register it's own regulator device. sgtl5000 0-000a: Failed to get supply 'VDDD': -19 Here use regulator_get_optional() trying to look at the fact that whether the external VDDD supply is used or not. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..90d6d1e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1298,6 +1298,21 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) return 0; } +static int sgtl5000_external_vddd_used(struct snd_soc_codec *codec) +{ + struct regulator *consumer; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + consumer = regulator_get_optional(codec->dev, + sgtl5000->supplies[VDDD].supply); + if (IS_ERR(consumer)) + return 0; + + regulator_put(consumer); + + return 1; +} + static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { int reg; @@ -1310,11 +1325,15 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + if (sgtl5000_external_vddd_used(codec)) { + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - if (!ret) + if (ret) + return ret; + external_vddd = 1; - else { + } else { ret = sgtl5000_replace_vddd_with_ldo(codec); if (ret) return ret; -- 1.8.4

4 4

[alsa-devel] [PATCH] ALSA: hda - limit mic boost on Asus UX31[A, E]
by Oleksij Rempel 15 Dec '13

15 Dec '13

This both devices need limit for internal dmic. Signed-off-by: Oleksij Rempel <linux(a)rempel-privat.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4ad9d1..b617aa4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3798,6 +3798,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -4075,6 +4076,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269VB_FIXUP_DMIC, + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4189,8 +4196,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), - SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 1.8.3.2

2 7

[alsa-devel] [RFC][PATCH 00/13] bebob: a new driver for BridgeCo BeBoB based device
by Takashi Sakamoto 15 Dec '13

15 Dec '13

In previous series of patch, I showed enhancement of current firewire-lib. So I should show the way to use new functions in each driver side. This driver is under development but functionalities for streaming seems to be fixed. I'm happy to receive your comments. This is a new driver for BridgeCo BeBoB based device. FFADO (successor of FreeBoB project) currently give driver and mixer control in user space. My aim of this series of patch is just to give a support for kernel streaming. In theory, this driver just support kernel streaming and related functionality. But for my debugging, this module still includes some functionality which should be in user space. Additionally, there is a heavily customized version of BeBoB. For some reasons, this driver give a few functionality to control such device. Takashi Sakamoto (13): bebob: Add skelton for BridgeCo BeBoB based devices bebob: Add command and stream management bebob: Add proc interface for debug bebob: Add functionality to change sampling rate and control interface bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface to lock/unlock streaming bebob: Prepare for device specific operations bebob: add support for M-Audio Firewire series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for Terratec PHASE series bebob: Add support for Yamaha GO series bebob: Add support for Presonus FIREBOX/FIREPOD include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 29 ++ sound/firewire/Makefile | 1 + sound/firewire/bebob/Makefile | 6 + sound/firewire/bebob/bebob.c | 450 ++++++++++++++++ sound/firewire/bebob/bebob.h | 255 +++++++++ sound/firewire/bebob/bebob_command.c | 423 +++++++++++++++ sound/firewire/bebob/bebob_control.c | 331 ++++++++++++ sound/firewire/bebob/bebob_focusrite.c | 390 ++++++++++++++ sound/firewire/bebob/bebob_hwdep.c | 207 ++++++++ sound/firewire/bebob/bebob_maudio.c | 910 +++++++++++++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 182 +++++++ sound/firewire/bebob/bebob_pcm.c | 420 +++++++++++++++ sound/firewire/bebob/bebob_presonus.c | 75 +++ sound/firewire/bebob/bebob_proc.c | 164 ++++++ sound/firewire/bebob/bebob_stream.c | 724 ++++++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 143 ++++++ sound/firewire/bebob/bebob_yamaha.c | 176 +++++++ 19 files changed, 4889 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_control.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_presonus.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c -- 1.8.3.2

2 20

[alsa-devel] [PATCH] ASOC:DAPM: extend dapm kcontrol to support runtime route update
by Nenghua Cao 15 Dec '13

15 Dec '13

From: nhcao <nhcao(a)marvell.com> DPCM can dynamically alter the FE to BE PCM links at runtime based on mixer setting updates. DAPM has provided common get/put function for mixer/mux. But these function doesn't call soc_dpcm_runtime_update() func to update pcm links. This patch defines a DPCM DPCM kcontrol. For this control, the common get/put function will update FE to BE links dynamically. This patch has no impact to the current code. Change-Id: I45e291f467e4fa17e21aa8923c2b712d52067aca Signed-off-by: nhcao <nhcao(a)marvell.com> --- include/sound/control.h | 4 ++++ include/sound/soc-dapm.h | 32 ++++++++++++++++++++++++++++++++ sound/core/control.c | 1 + sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 4 files changed, 61 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 5358892..3788284 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -44,6 +44,8 @@ struct snd_kcontrol_new { unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ + /* this kcontrol impact fe<->be relationship at runtime */ + unsigned char dpcm_checked; snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; snd_kcontrol_put_t *put; @@ -63,6 +65,8 @@ struct snd_kcontrol { struct list_head list; /* list of controls */ struct snd_ctl_elem_id id; unsigned int count; /* count of same elements */ + /* this kcontrol impact fe<->be relationship at runtime */ + unsigned char dpcm_checked; snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; snd_kcontrol_put_t *put; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 56ebdfc..8962884 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -336,6 +336,38 @@ struct device; .put = snd_soc_dapm_put_pin_switch, \ .private_value = (unsigned long)xname } +/* dapm & dpcm kcontrol types */ +#define SOC_DAPM_DPCM_SINGLE(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DAPM_DPCM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DAPM_DPCM_ENUM(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = snd_soc_dapm_put_enum_double, \ + .private_value = (unsigned long)&xenum } +#define SOC_DAPM_DPCM_ENUM_VIRT(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_virt, \ + .put = snd_soc_dapm_put_enum_virt, \ + .private_value = (unsigned long)&xenum } +#define SOC_DAPM_DPCM_VALUE_ENUM(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_value_enum_double, \ + .put = snd_soc_dapm_put_value_enum_double, \ + .private_value = (unsigned long)&xenum } + /* dapm stream operations */ #define SND_SOC_DAPM_STREAM_NOP 0x0 #define SND_SOC_DAPM_STREAM_START 0x1 diff --git a/sound/core/control.c b/sound/core/control.c index d8aa206..34df2c4 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -254,6 +254,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); + kctl.dpcm_checked = ncontrol->dpcm_checked; kctl.info = ncontrol->info; kctl.get = ncontrol->get; kctl.put = ncontrol->put; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcade13..b568183 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2868,6 +2868,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val; int connect, change; struct snd_soc_dapm_update update; + int ret = 0; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, @@ -2901,12 +2902,16 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, card->update = &update; } - soc_dapm_mixer_update_power(card, kcontrol, connect); + ret = soc_dapm_mixer_update_power(card, kcontrol, connect); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2955,6 +2960,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2978,12 +2984,16 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -3019,6 +3029,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; + int ret = 0; if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; @@ -3028,9 +3039,13 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, value = ucontrol->value.enumerated.item[0]; change = dapm_kcontrol_set_value(kcontrol, value); if (change) - soc_dapm_mux_update_power(card, kcontrol, value, e); + ret = soc_dapm_mux_update_power(card, kcontrol, value, e); mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -3097,6 +3112,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -3120,12 +3136,16 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); -- 1.7.0.4

4 4

[alsa-devel] Add support for Focusrite Saffire 6 USB
by Эдуард Гильмутдинов 14 Dec '13

14 Dec '13

Hello! I have external audio card Focusrite Saffire 6 USB. This device works with the Linux kernel if I add the following corrections: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..158beb4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..3ea9627 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,10 +660,11 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } -/* +/* * Novation Twitch DJ controller + * Focusrite(Novation) Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ Help me to add this corrections in the Linux kernel mainline. Thank you!

1 0

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa: Disable timer-scheduling for PCMs with the BATCH flag
by Raymond Yau 14 Dec '13

14 Dec '13

2013/12/6 Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> > On Sat, 2013-11-30 at 18:07 +0100, Lars-Peter Clausen wrote: > > PCM Devices which have the BATCH flag set update the PCM pointer only > with > > period size granularity. Using timer based scheduling does not have any > > advantage in this mode. For one devices which have that flag set usually > update > > the position pointer in software after getting the period interrupt. So > > disabling the period interrupt is not possible for this kind of devices. > > Furthermore writing to or reading from the buffer slice for the current > period > > is not possible since the position inside the buffer is not known. On > the other > > hand the tsched algorithm seems to get easily confused for this kind of > > hardware, which results in garbled audio output. This typically means > that timer > > based scheduling needs to be manually disabled on systems with such > devices. > > Auto disabling tsched in this case allows these systems to run with the > default > > configuration. > > > > Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > --- > > src/modules/alsa/alsa-util.c | 4 ++++ > > 1 file changed, 4 insertions(+) > > > > diff --git a/src/modules/alsa/alsa-util.c b/src/modules/alsa/alsa-util.c > > index 75f5858..4b24e47 100644 > > --- a/src/modules/alsa/alsa-util.c > > +++ b/src/modules/alsa/alsa-util.c > > @@ -245,6 +245,10 @@ int pa_alsa_set_hw_params( > > if (!pa_alsa_pcm_is_hw(pcm_handle)) > > _use_tsched = false; > > > > + /* The PCM pointer is only updated with period granularity */ > > + if (snd_pcm_hw_params_is_batch(hwparams)) > > + _use_tsched = false; > > + > > #if (SND_LIB_VERSION >= ((1<<16)|(0<<8)|24)) /* API additions in 1.0.24 > */ > > if (_use_tsched) { > > > > Thanks! I applied the patch. > > -- > Are you patch sufficient ? In alsa-lib , snd_pcm_sw_params_set_avail_min always return zero int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val) { assert(pcm && params); /* Fix avail_min if it's below period size. The period_size * defines the minimal wake-up timing accuracy, so it doesn't * make sense to set below that. */ if (val < pcm->period_size) val = pcm->period_size; params->avail_min = val; return 0; } http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… /* We need at last one frame in the used part of the buffer */ - avail_min = (snd_pcm_uframes_t) u->hwbuf_unused_frames + 1; + avail_min = (snd_pcm_uframes_t) u->hwbuf_unused / u->frame_size + 1; do pulseaudio need to keep at least one periods instead of more than 1 frames for those driver which update hwptr in periods ?

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Correct S24_3LE support
by Peter Ujfalusi 13 Dec '13

13 Dec '13

Correct the hw_params callback to configure the codec correctly in case of S24_3LE format. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb4b386..31b6ae2334bc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: -- 1.8.5.1

4 11

[alsa-devel] ASoC updates for v3.13
by Mark Brown 13 Dec '13

13 Dec '13

The following changes since commit 374b105797c3d4f29c685f3be535c35f5689b30e: Linux 3.13-rc3 (2013-12-06 09:34:04 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc3 for you to fetch changes up to e20ab019e28dcf09c2727aa69e2a073ed66718b3: Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/tegra' and 'asoc/fix/wm8962' into asoc-linus (2013-12-12 00:30:04 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few driver and error handling fixes plus a fix to ensure that we mute streams when we should. The Atmel trigger addition is a fix to ensure that we do the correct sequence of interactions with the hardware. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dma' into asoc-linus Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/tegra' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver Stephen Warren (2): ASoC: tegra: fix uninitialized variables in set_fmt ASoC: don't leak on error in snd_dmaengine_pcm_register sound/soc/atmel/atmel_ssc_dai.c | 30 ++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/wm8962.c | 13 ++++++++++++ sound/soc/fsl/imx-wm8962.c | 2 -- sound/soc/soc-generic-dmaengine-pcm.c | 38 +++++++++++++++++++++++++---------- sound/soc/soc-pcm.c | 5 +++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 ++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 9 files changed, 84 insertions(+), 28 deletions(-)

2 2

Re: [alsa-devel] [PATCH] ASOC:DAPM: extend dapm kcontrol to support runtime route update
by Mark Brown 13 Dec '13

13 Dec '13

On Thu, Dec 12, 2013 at 02:56:08AM -0800, Nenghua Cao wrote: Don't top post. Please also fix your mailer to word wrap within paragaphs, it makes your mails much more legible. > For common mix/mux which aren't related with DPCM, it doesn't need > to do DPCM update. And it still can use the old controls. But for > the mix/mux which impacts fe<->be link, they need to do this works. > For them, they call use the new set of controls. I think it can > make our code has better backward-compatibility. How does this improve backwards compatibility?

3 6

[alsa-devel] [PATCH V2] ASOC:DAPM: update DPCM runtime when mixer/mux changes
by Nenghua Cao 13 Dec '13

13 Dec '13

From: Nenghua Cao <nhcao(a)marvell.com> V2 -> V1: Removes the flag and update DPCM always when mixer/mux changes, according to Takashi's and Mark's suggestion. >> Usually backward compatibility is concerned when something new breaks >> the existing ones. In your case, always updating DPCM would work, >> too, even without an extra flag; it's just suboptimal. >Right, and if we have separate versions then at some point we'll end up >having to define DPCM versions of everything which is going to get >tedious and error prone. V1 -> V0: refined patch, and add flag into soc_mixer_control and soc_enum instead of the alsa core, according to Takashi's suggestion. > You don't need to extend the ALSA core API for such a purpose. It's > ASoC-specific, so rather extend struct soc_mixer_control instead, if > such a new flag is really needed. Nenghua Cao (1): ASOC:DAPM: update DPCM runtime when mixer/mux changes sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 1 files changed, 24 insertions(+), 4 deletions(-)

1 1

[alsa-devel] [PATCH] ASoC: omap-pcm: Lower the dma coherent mask to 32bits
by Peter Ujfalusi 13 Dec '13

13 Dec '13

The underlying API dma_coerce_mask_and_coherent() checks the requested bits mask against the size of dma_addr_t which is 32bits on OMAP. Because of the previously used 64bits mask audio was not probing after commit c9bd5e6 (and 4dcfa6007). 32bits for the DMA_BIT_MASK looks to be the correct one to use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, This is the fix for OMAP audio for 3.13-rc2+. The dma coherent changes went in between -rc1 and -rc2 which broke things because omap-pcm was using 64bits for dma coherent mask. Regards, Peter sound/soc/omap/omap-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b8fa986..01d59d0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -202,7 +202,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; -- 1.8.4.5

3 17

[alsa-devel] [PATCH V1] ASOC:DAPM: extend dapm kcontrol to support runtime route update
by Nenghua Cao 13 Dec '13

13 Dec '13

From: Nenghua Cao <nhcao(a)marvell.com> V1 -> V0: refined patch, and add flag into soc_mixer_control and soc_enum instead of the alsa core, according to Takashi's suggestion. >You don't need to extend the ALSA core API for such a purpose. It's >ASoC-specific, so rather extend struct soc_mixer_control instead, if >such a new flag is really needed. nhcao (1): ASOC:DAPM: extend dapm kcontrol to support runtime route update include/sound/soc-dapm.h | 14 ++++++++++++++ include/sound/soc.h | 19 +++++++++++++++++++ sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 3 files changed, 57 insertions(+), 4 deletions(-)

2 2

[alsa-devel] [PATCH] alsa-lib: minor malloc changes
by Olivier Langlois 13 Dec '13

13 Dec '13

replace malloc + memset with calloc since calloc may skip the memset if returned memory comes directly from sbrk() use malloc instead of malloc if first thing done with allocated memory is to flip all bits to 1. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/alisp/alisp.c | 3 +-- src/pcm/pcm_ladspa.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/src/alisp/alisp.c b/src/alisp/alisp.c index 1796c33..3c61bb1 100644 --- a/src/alisp/alisp.c +++ b/src/alisp/alisp.c @@ -3202,12 +3202,11 @@ int alsa_lisp(struct alisp_cfg *cfg, struct alisp_instance **_instance) struct alisp_object *p, *p1; int i, j, retval = 0; - instance = (struct alisp_instance *)malloc(sizeof(struct alisp_instance)); + instance = (struct alisp_instance *)calloc(1, sizeof(struct alisp_instance)); if (instance == NULL) { nomem(); return -ENOMEM; } - memset(instance, 0, sizeof(struct alisp_instance)); instance->verbose = cfg->verbose && cfg->vout; instance->warning = cfg->warning && cfg->wout; instance->debug = cfg->debug && cfg->dout; diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 0a9c52e..9ce5242 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -1326,7 +1326,7 @@ static int snd_pcm_ladspa_parse_bindings(snd_pcm_ladspa_plugin_t *lplug, count = (unsigned int)(channel + 1); } if (count > 0) { - array = (unsigned int *)calloc(count, sizeof(unsigned int)); + array = (unsigned int *)malloc(count * sizeof(unsigned int)); if (! array) return -ENOMEM; memset(array, 0xff, count * sizeof(unsigned int)); -- 1.8.5.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Dell headset detection quirk for three laptop models
by Hui Wang 13 Dec '13

13 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280610, 0x10280629 or 0x1028063e, no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. The codecs on these machines belong to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c5f3e1b..3578f11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4300,11 +4300,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: tegra: update module reset list for Tegra124
by Stephen Warren 12 Dec '13

12 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra124 adds a number of extra modules into the configlink bus, which must be taken out of reset before the bus is used. Update the AHUB driver to know about these extra modules (the AHUB HW module hosts the configlink bus). Based-on-work-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Based-on-work-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This patch depends on "ASoC: tegra: use reset framework" to compile, which is ack'd and slated to go through a (large) topic branch in the Tegra tree. So, we can either: a) Merge that Tegra topic branch into the ASoC tree, then apply this. Note that I haven't created the topic branch yet, since I'm still waiting for DMA dependencies to be applied. b) Apply this change to the Tegra tree too. This change isn't directly related to the changes in the Tegra tree; it just makes use of the new reset controller feature that's introduced there. .../bindings/sound/nvidia,tegra30-ahub.txt | 9 +++++++++ sound/soc/tegra/tegra30_ahub.c | 20 ++++++++++++++++---- 2 files changed, 25 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt index 32de7ec789aa..946e2ac46091 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -30,6 +30,15 @@ Required properties: Tegra114 and later additionally require: - amx - adx + Tegra124 and later additionally require: + - amx1 + - adx1 + - afc0 + - afc1 + - afc2 + - afc3 + - afc4 + - afc5 - ranges : The bus address mapping for the configlink register bus. Can be empty since the mapping is 1:1. - dmas : Must contain an entry for each entry in clock-names. diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 342cd4fff0a5..d6f4c9940e0c 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -335,9 +335,13 @@ EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); #define MOD_LIST_MASK_TEGRA30 BIT(0) #define MOD_LIST_MASK_TEGRA114 BIT(1) +#define MOD_LIST_MASK_TEGRA124 BIT(2) #define MOD_LIST_MASK_TEGRA30_OR_LATER \ - (MOD_LIST_MASK_TEGRA30 | MOD_LIST_MASK_TEGRA114) + (MOD_LIST_MASK_TEGRA30 | MOD_LIST_MASK_TEGRA114 | \ + MOD_LIST_MASK_TEGRA124) +#define MOD_LIST_MASK_TEGRA114_OR_LATER \ + (MOD_LIST_MASK_TEGRA114 | MOD_LIST_MASK_TEGRA124) static const struct { const char *rst_name; @@ -352,8 +356,16 @@ static const struct { { "dam1", MOD_LIST_MASK_TEGRA30_OR_LATER }, { "dam2", MOD_LIST_MASK_TEGRA30_OR_LATER }, { "spdif", MOD_LIST_MASK_TEGRA30_OR_LATER }, - { "amx", MOD_LIST_MASK_TEGRA114 }, - { "adx", MOD_LIST_MASK_TEGRA114 }, + { "amx", MOD_LIST_MASK_TEGRA114_OR_LATER }, + { "adx", MOD_LIST_MASK_TEGRA114_OR_LATER }, + { "amx1", MOD_LIST_MASK_TEGRA124 }, + { "adx1", MOD_LIST_MASK_TEGRA124 }, + { "afc0", MOD_LIST_MASK_TEGRA124 }, + { "afc1", MOD_LIST_MASK_TEGRA124 }, + { "afc2", MOD_LIST_MASK_TEGRA124 }, + { "afc3", MOD_LIST_MASK_TEGRA124 }, + { "afc4", MOD_LIST_MASK_TEGRA124 }, + { "afc5", MOD_LIST_MASK_TEGRA124 }, }; #define LAST_REG(name) \ @@ -492,7 +504,7 @@ static struct tegra30_ahub_soc_data soc_data_tegra114 = { }; static struct tegra30_ahub_soc_data soc_data_tegra124 = { - .mod_list_mask = MOD_LIST_MASK_TEGRA114, + .mod_list_mask = MOD_LIST_MASK_TEGRA124, .set_audio_cif = tegra124_ahub_set_cif, }; -- 1.8.1.5

2 2

[alsa-devel] snd-usb driver
by Lucas Takejame 12 Dec '13

12 Dec '13

Hello guys, I'm new here. The company that i work for is developing a LV2 host pedal board (http://portalmod.com/en/index.html) and my job now is to find if I can improve in anyway our usb driver (which is the kernel's snd-usb driver). I know the code is already optimized but I was hoping that, since we only use one specific sound card and we won't need a generic driver, I could change something in the code to optimize it latency wise. I have only a brief knowledge on the USB protocol and even less on writing drivers, so I'm here first to ask if it's possible to improve latency by changing the driver's code and, if possible, if you guys could give me some directions on how to do it. Thanks in advance, Lucas

4 4

[alsa-devel] how to create alsa nodes?
by loody 12 Dec '13

12 Dec '13

Dear all: In destop x86 system, the udev will automatically create the nodes in snd directory, such as pcmC0D0c, pcmC0D1c, etc. I try to let the usb sound device work on mips platform, but I have no idea what is the proper way to create these nodes. Should I 1. create them one by one by hand from copying the attributes in my PC (if I have 2 or more usb sound devices, I have to first plug them in pc and monitor the nodes) 2. smarter way to do so? -- Regards, miloody

7 19

[alsa-devel] ALSA: SNDRV_PCM_STATE_PAUSED
by JongHo Kim 12 Dec '13

12 Dec '13

Hi I develop ALSA SOC Part and use the tinyalsa library on application space. We recently used pause ioctl command(SNDRV_PCM_IOCTL_PAUSE) because the music application was needed to pause and resume feature. So the music application was achieved to want pause and resume feature using the SNDRV_PCM_IOCTL_PAUSE. but I found some problem in alsa kernel framework The music application process on pause state wake up in 10 seconds because the wait time is set by 10 seconds in wait_for_avail function. I realized that if the 'runtime->no_period_wakeup' parameter is true, the wait_time will be set by the MAX_SCHEDULE_TIMEOUT. I had searched other kernel source and library source. I added the SNDRV_PCM_INFO_NO_PERIOD_WAKEUP flag at my snd_pcm_hardware structure and added the SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP flag at music application(use tinyalsa library) Result was good! The music application perfectly works the pause and resume feature. but I have some worries if DMA or I2S was stop on the running, I can't more see the 'write error (DMA or IRQ trouble?)' message and the music application process may be sleep eternally. I cautiously recommend next code with a solution to the above worries. ------------------------------------------------------------------------------------- diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c old mode 100644 new mode 100755 index 41b3dfe..3284940 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1936,6 +1936,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", ------------------------------------------------------------------------------------- I tested code. code works good! but I am concerned about the side effects please review my idea.

2 1

[alsa-devel] [PATCH] ALSA: hda - Add enable_msi=0 workaround for four HP machines
by David Henningsson 12 Dec '13

12 Dec '13

While enabling these machines, we found we would sometimes lose an interrupt if we change hardware volume during playback, and that disabling msi fixed this issue. (Losing the interrupt caused underruns and crackling audio, as the one second timeout is usually bigger than the period size.) The machines were all machines from HP, running AMD Hudson controller, and Realtek ALC282 codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1260225 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index af86c71..440c355 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3446,6 +3446,10 @@ static void check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] = { + SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: drop sgtl5000_set_bias_level() call from sgtl5000_remove()
by Shawn Guo 12 Dec '13

12 Dec '13

As shown below, there will be a kernel Oops with imx-sgtl5000 module in the sequence of install -> remove -> install again. It happens because during the removal of module imx-sgtl5000, the following calls will set sgtl5000 regmap into cache only mode. sgtl5000_remove(codec) sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF) regcache_cache_only(sgtl5000->regmap, true) This will cause the follow calls fail with -EBUSY error code in the next imx-sgtl5000 module installation. sgtl5000_probe(codec) sgtl5000_enable_regulators(codec) regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg) _regmap_read(map, reg, val) if (map->cache_only) return -EBUSY; Looking at function sgtl5000_set_bias_level(), we can find it essentially does the following 3 things in case of SND_SOC_BIAS_OFF. 1. Set regmap into cache only mode with regcache_cache_only() call 2. Disable supplies by calling regulator_bulk_disable() 3. Set codec->dapm.bias_level as SND_SOC_BIAS_OFF For 1, it's clearly incorrect and that's why we see the failure here. For 2, it's unnecessary since sgtl5000_remove() already calls regulator_bulk_disable() to disable supplies. For 3, it does not make any sense either, since the bias level will be initialized to SND_SOC_BIAS_STANDBY in the next probe anyway. Let's drop the sgtl5000_set_bias_level() call from sgtl5000_remove() to fix the kernel Oops below. $ lsmod Module Size Used by snd_soc_imx_sgtl5000 3514 0 snd_soc_sgtl5000 13677 2 snd_soc_imx_audmux 5324 1 snd_soc_imx_sgtl5000 snd_soc_fsl_ssi 8091 2 imx_pcm_dma 1132 1 snd_soc_fsl_ssi $ rmmod snd_soc_imx_sgtl5000 $ lsmod Module Size Used by snd_soc_sgtl5000 13677 0 snd_soc_imx_audmux 5324 0 snd_soc_fsl_ssi 8091 0 imx_pcm_dma 1132 1 snd_soc_fsl_ssi $ insmod snd-soc-imx-sgtl5000.ko 0-000a supply VDDD not found, using dummy regulator Unable to handle kernel NULL pointer dereference at virtual address 000000d0 pgd = beec8000 [000000d0] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: snd_soc_imx_sgtl5000(+) snd_soc_sgtl5000 snd_soc_imx_audmux snd_soc_fsl_ssi imx_pcm_dma [last unloaded: snd_soc_imx_sgtl5000] CPU: 0 PID: 1814 Comm: insmod Not tainted 3.13.0-rc1+ #1571 task: bed8da00 ti: bdc94000 task.ti: bdc94000 PC is at __lock_acquire+0x6c/0x1cd8 LR is at lock_acquire+0x68/0x7c pc : [<8005e8d0>] lr : [<80060a44>] psr: 20000093 sp : bdc95a78 ip : 80d86bfc fp : bdc95af4 r10: 00000000 r9 : 00000000 r8 : 00000000 r7 : 000000d0 r6 : 8085e560 r5 : 00000080 r4 : 00000002 r3 : 00000000 r2 : 00000000 r1 : 00000000 r0 : 000000d0 Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 4eec804a DAC: 00000015 Process insmod (pid: 1814, stack limit = 0xbdc94240) Stack: (0xbdc95a78 to 0xbdc96000) 5a60: 000003e8 00000003 5a80: 8086d388 bed8dde8 00000000 00000008 bdc95b1c bdc95aa0 bdc94000 bed8da00 5aa0: bdc95b14 80d86bfc 6c13633a 00000002 00000000 000000a1 6c6740a1 00004d82 5ac0: bdc94000 00000003 bed8da00 00000000 bdc94000 60000013 00000000 bf6a8e60 5ae0: bed8da00 bdc94000 bdc95b2c bdc95af8 80060a44 8005e870 00000002 00000000 5b00: 00000000 80252b58 00000000 bf3e841c 00000098 80252b58 80d86bfc 00000000 5b20: bdc95b84 bdc95b30 8060603c 800609e8 00000002 00000000 80252b58 bdc95b48 5b40: 8006149c 800612c8 bdc95bac bdc95b58 806062c4 80061494 00000002 bf6a8e60 5b60: bf3e8410 fffffff0 bf3e8410 bf6a8e60 8086ad70 00000001 bdc95bac bdc95b88 5b80: 80252b58 80605fec bed8da00 bf11b000 bf3e8410 fffffff0 bf3e8410 bf3e841c 5ba0: bdc95bc4 bdc95bb0 802d72d4 80252b1c bed8da00 bf3e8680 bdc95bdc bdc95bc8 5bc0: 7f00e8a8 802d7278 bed8da00 bed92000 bdc95c1c bdc95be0 7f00f240 7f00e88c 5be0: 00000000 00000000 00000000 00000000 00000000 bed92000 bec52064 7f010718 5c00: 8088cc54 8088cd20 8088cd18 bed92118 bdc95c4c bdc95c20 804979f0 7f00eec8 5c20: 8064fb28 bdc95c30 bec52064 00000000 bed9340c beec6b00 8088ccc8 00000000 5c40: bdc95cbc bdc95c50 804985d4 80497880 bec52074 80296618 bddabf40 00000003 5c60: 00000005 816105c4 bec521a4 bec521b4 bec521cc bec52084 bec521ac bec521dc 5c80: 8088cd28 00000000 00000054 00000000 bdc95cbc bf9a4010 00000000 bec52010 5ca0: 81605c30 8160e880 80dca5e8 bec52064 bdc95cec bdc95cc0 7f01a278 80497b84 5cc0: 00000000 00000003 bf9a4010 7f01aa00 bf9a4044 00000000 7f01aa00 00000000 5ce0: bdc95d04 bdc95cf0 80327758 7f01a03c 80ddc868 bf9a4010 bdc95d2c bdc95d08 5d00: 80325bcc 80327744 00000000 bf9a4010 7f01aa00 bf9a4044 00000000 7f01c000 5d20: bdc95d4c bdc95d30 80325d7c 80325ac0 bf9917dc 7f01aa00 80325ce0 00000000 5d40: bdc95d74 bdc95d50 80324320 80325cec bf8048a8 bf9917d0 bf3e8ed8 7f01aa00 5d60: bef38680 80875a40 bdc95d84 bdc95d78 803256c4 803242d0 bdc95dac bdc95d88 5d80: 803252c4 803256b0 7f01a974 bdc95d98 7f01aa00 00000001 80000000 7f01aa50 5da0: bdc95dc4 bdc95db0 80326434 803251f0 bdc94000 00000001 bdc95dd4 bdc95dc8 5dc0: 803270ec 803263c0 bdc95de4 bdc95dd8 7f01c018 803270a8 bdc95e7c bdc95de8 5de0: 800088b8 7f01c00c 8006149c 800612c8 7f01c000 bdc95e00 800a7854 80061494 5e00: 815b85b0 815b85a0 00000000 00000000 00000000 00005c0b bdc95e44 00000003 5e20: bddab140 00000001 00000000 7f01aa44 80dca5e8 bddabd80 bdc95e64 bdc95e48 5e40: 800ccc4c 800d6584 bdc95f48 00000001 00000001 bdc95f48 00000001 00000001 5e60: 7f01aa50 7f01aa44 80dca5e8 bddabd80 bdc95f44 bdc95e80 80087788 80008884 5e80: 7f01aa50 00007fff 80085684 000002d2 ffffffff 00000000 7f01aa50 00000000 5ea0: c1934720 bdc95eb0 7f01ab80 00000000 7f01aa8c bdc94000 7f01ab70 bdc95f20 5ec0: 6e72656b 00006c65 00000000 00000000 00000000 00000000 00000000 00000000 5ee0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 5f00: 00000000 00000000 00000000 00000000 00000000 807d1008 20000013 000021a2 5f20: 76f19000 76ef6d50 00000080 8000e9c4 bdc94000 00000000 bdc95fa4 bdc95f48 5f40: 80088004 80086064 c1933000 000021a2 c19343b0 c19342d4 c1934e28 00000b80 5f60: 00000cd0 00000000 00000000 00000000 00000016 00000017 0000000e 0000000b 5f80: 00000009 00000000 00000000 00000000 785d2d98 785d2088 00000000 bdc95fa8 5fa0: 8000e800 80087f2c 00000000 785d2d98 76f19000 000021a2 76ef6d50 00000002 5fc0: 00000000 785d2d98 785d2088 00000080 00000000 76ef6d50 000021a2 00000000 5fe0: 00000000 7eb3cc0c 76ef0b07 76e72d94 80000010 76f19000 ffffffff fffbdff7 Backtrace: [<8005e864>] (__lock_acquire+0x0/0x1cd8) from [<80060a44>] (lock_acquire+0x68/0x7c) [<800609dc>] (lock_acquire+0x0/0x7c) from [<8060603c>] (mutex_lock_nested+0x5c/0x3b8) r7:00000000 r6:80d86bfc r5:80252b58 r4:00000098 [<80605fe0>] (mutex_lock_nested+0x0/0x3b8) from [<80252b58>] (debugfs_remove_recursive+0x48/0x154) [<80252b10>] (debugfs_remove_recursive+0x0/0x154) from [<802d72d4>] (regulator_unregister+0x68/0xec) r8:bf3e841c r7:bf3e8410 r6:fffffff0 r5:bf3e8410 r4:bf11b000 r3:bed8da00 [<802d726c>] (regulator_unregister+0x0/0xec) from [<7f00e8a8>] (ldo_regulator_remove+0x28/0x40 [snd_soc_sgtl5000]) r4:bf3e8680 r3:bed8da00 [<7f00e880>] (ldo_regulator_remove+0x0/0x40 [snd_soc_sgtl5000]) from [<7f00f240>] (sgtl5000_probe+0x384/0x4ac [snd_soc_sgtl5000]) r4:bed92000 r3:bed8da00 [<7f00eebc>] (sgtl5000_probe+0x0/0x4ac [snd_soc_sgtl5000]) from [<804979f0>] (soc_probe_codec+0x17c/0x304) [<80497874>] (soc_probe_codec+0x0/0x304) from [<804985d4>] (snd_soc_register_card+0xa5c/0x14f8) [<80497b78>] (snd_soc_register_card+0x0/0x14f8) from [<7f01a278>] (imx_sgtl5000_probe+0x248/0x2ec [snd_soc_imx_sgtl5000]) [<7f01a030>] (imx_sgtl5000_probe+0x0/0x2ec [snd_soc_imx_sgtl5000]) from [<80327758>] (platform_drv_probe+0x20/0x50) [<80327738>] (platform_drv_probe+0x0/0x50) from [<80325bcc>] (driver_probe_device+0x118/0x22c) r5:bf9a4010 r4:80ddc868 [<80325ab4>] (driver_probe_device+0x0/0x22c) from [<80325d7c>] (__driver_attach+0x9c/0xa0) r8:7f01c000 r7:00000000 r6:bf9a4044 r5:7f01aa00 r4:bf9a4010 r3:00000000 [<80325ce0>] (__driver_attach+0x0/0xa0) from [<80324320>] (bus_for_each_dev+0x5c/0x90) r6:00000000 r5:80325ce0 r4:7f01aa00 r3:bf9917dc [<803242c4>] (bus_for_each_dev+0x0/0x90) from [<803256c4>] (driver_attach+0x20/0x28) r6:80875a40 r5:bef38680 r4:7f01aa00 [<803256a4>] (driver_attach+0x0/0x28) from [<803252c4>] (bus_add_driver+0xe0/0x1d8) [<803251e4>] (bus_add_driver+0x0/0x1d8) from [<80326434>] (driver_register+0x80/0xfc) r7:7f01aa50 r6:80000000 r5:00000001 r4:7f01aa00 [<803263b4>] (driver_register+0x0/0xfc) from [<803270ec>] (__platform_driver_register+0x50/0x64) r5:00000001 r4:bdc94000 [<8032709c>] (__platform_driver_register+0x0/0x64) from [<7f01c018>] (imx_sgtl5000_driver_init+0x18/0x24 [snd_soc_imx_sgtl5000]) [<7f01c000>] (imx_sgtl5000_driver_init+0x0/0x24 [snd_soc_imx_sgtl5000]) from [<800088b8>] (do_one_initcall+0x40/0x170) [<80008878>] (do_one_initcall+0x0/0x170) from [<80087788>] (load_module+0x1730/0x1ec8) [<80086058>] (load_module+0x0/0x1ec8) from [<80088004>] (SyS_init_module+0xe4/0xe8) [<80087f20>] (SyS_init_module+0x0/0xe8) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r6:785d2088 r5:785d2d98 r4:00000000 Code: e59fcdd4 e59c3000 e3530000 0a00002e (e5973000) ---[ end trace 30f8d1c16c188524 ]--- Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..9aa63ca 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1455,8 +1455,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) { struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), -- 1.7.9.5

2 2

[alsa-devel] [PATCH 00/31] ARM: tegra: use common reset and DMA bindings
by Stephen Warren 12 Dec '13

12 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This series implements a common reset framework driver for Tegra, and updates all relevant Tegra drivers to use it. It also removes the custom DMA bindings and replaced them with the standard DMA DT bindings. Historically, the Tegra clock driver has exported a custom API for module reset. This series removes that API, and transitions DT and drivers to the new reset framework. The custom API used a "struct clk" to identify which module to reset, and consequently some DT bindings and drivers required clocks to be provided where they really needed just a reset identifier instead. Due to this known deficiency, I have always considered most Tegra bindings to be unstable. This series removes this excuse for instability, although I still consider some Tegra bindings unstable due to the need to convert to the common DMA bindings. Historically, Tegra DMA channels have been represented in DT using a custom nvidia,dma-request-selector property. Now that standard DMA DT bindings exist, convert all Tegra bindings, DTs, and drivers to use the standard instead. This series makes a DT-ABI-incompatible change to: - Require reset specifiers in DT where relevant. - Require standard DMA specifiers. - Remove clock specifiers from DT where they were only needed for reset. - Remove legacy DMA specifier properties. I anticipate merging this whole series into the Tegra and arm-soc trees as its own branch, due to internal dependencies. This branch will be stable and can then be merged into any other subsystem trees should any conflicts arise. This series depends on Peter's Tegra clock driver rework, available at git://nv-tegra.nvidia.com/user/pdeschrijver/linux tegra-clk-tegra124-0 (or whatever version of that gets included in 3.14) Cc: ac100(a)lists.launchpad.net Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: alsa-devel(a)alsa-project.org Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Dan Williams <dan.j.williams(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: devel(a)driverdev.osuosl.org Cc: devicetree(a)vger.kernel.org Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Dmitry Torokhov <dtor(a)mail.ru> Cc: dri-devel(a)lists.freedesktop.org Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Julian Andres Klode <jak(a)jak-linux.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-pci(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-tegra(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: Marc Dietrich <marvin24(a)gmx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Mike Turquette <mturquette(a)linaro.org> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: pdeschrijver(a)nvidia.com Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Terje Bergström <tbergstrom(a)nvidia.com> Cc: treding(a)nvidia.com Cc: Wolfram Sang <wsa(a)the-dreams.de> Stephen Warren (31): ARM: tegra: add missing clock documentation to DT bindings ARM: tegra: document reset properties in DT bindings ARM: tegra: document use of standard DMA DT bindings ARM: tegra: update DT files to add reset properties ARM: tegra: update DT files to add DMA properties ARM: tegra: select the reset framework clk: tegra: implement a reset driver pci: tegra: use reset framework drm/tegra: use reset framework ARM: tegra: pass reset to tegra_powergate_sequence_power_up() dma: add channel request API that supports deferred probe dma: tegra: use reset framework dma: tegra: register as an OF DMA controller ASoC: dmaengine: support deferred probe for DMA channels ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config ASoC: tegra: use reset framework ASoC: tegra: call pm_runtime APIs around register accesses ASoC: tegra: allocate AHUB FIFO during probe() not startup() ASoC: tegra: convert to standard DMA DT bindings i2c: tegra: use reset framework staging: nvec: use reset framework spi: tegra: use reset framework spi: tegra: convert to standard DMA DT bindings serial: tegra: use reset framework serial: tegra: convert to standard DMA DT bindings Input: tegra-kbc - use reset framework USB: EHCI: tegra: use reset framework ARM: tegra: remove legacy clock entries from DT ARM: tegra: remove legacy DMA entries from DT clk: tegra: remove legacy reset APIs clk: tegra: remove bogus PCIE_XCLK .../bindings/arm/tegra/nvidia,tegra20-pmc.txt | 1 + .../bindings/clock/nvidia,tegra114-car.txt | 4 + .../bindings/clock/nvidia,tegra124-car.txt | 4 + .../bindings/clock/nvidia,tegra20-car.txt | 4 + .../bindings/clock/nvidia,tegra30-car.txt | 4 + .../devicetree/bindings/dma/tegra20-apbdma.txt | 9 ++ .../bindings/gpu/nvidia,tegra20-host1x.txt | 124 +++++++++++++++ .../devicetree/bindings/i2c/nvidia,tegra20-i2c.txt | 27 +++- .../bindings/input/nvidia,tegra20-kbc.txt | 9 ++ .../bindings/mmc/nvidia,tegra20-sdhci.txt | 9 ++ .../devicetree/bindings/nvec/nvidia,nvec.txt | 12 ++ .../bindings/pci/nvidia,tegra20-pcie.txt | 28 ++-- .../devicetree/bindings/pwm/nvidia,tegra20-pwm.txt | 9 ++ .../devicetree/bindings/rtc/nvidia,tegra20-rtc.txt | 3 + .../bindings/serial/nvidia,tegra20-hsuart.txt | 19 ++- .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-wm8753.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-wm9712.txt | 7 +- .../bindings/sound/nvidia,tegra20-ac97.txt | 20 ++- .../bindings/sound/nvidia,tegra20-i2s.txt | 19 ++- .../bindings/sound/nvidia,tegra30-ahub.txt | 54 +++++-- .../bindings/sound/nvidia,tegra30-i2s.txt | 11 +- .../bindings/spi/nvidia,tegra114-spi.txt | 24 ++- .../bindings/spi/nvidia,tegra20-sflash.txt | 20 ++- .../bindings/spi/nvidia,tegra20-slink.txt | 20 ++- .../bindings/timer/nvidia,tegra20-timer.txt | 3 + .../bindings/timer/nvidia,tegra30-timer.txt | 3 + .../bindings/usb/nvidia,tegra20-ehci.txt | 7 +- arch/arm/boot/dts/tegra114.dtsi | 142 ++++++++++++++--- arch/arm/boot/dts/tegra20-paz00.dts | 2 + arch/arm/boot/dts/tegra20.dtsi | 132 ++++++++++++++-- arch/arm/boot/dts/tegra30.dtsi | 171 +++++++++++++++++---- arch/arm/mach-tegra/Kconfig | 2 + arch/arm/mach-tegra/powergate.c | 8 +- drivers/clk/tegra/clk-periph-gate.c | 22 --- drivers/clk/tegra/clk-periph.c | 40 ----- drivers/clk/tegra/clk-tegra114.c | 3 +- drivers/clk/tegra/clk-tegra124.c | 2 +- drivers/clk/tegra/clk-tegra20.c | 9 +- drivers/clk/tegra/clk-tegra30.c | 10 +- drivers/clk/tegra/clk.c | 55 ++++++- drivers/clk/tegra/clk.h | 3 +- drivers/dma/acpi-dma.c | 12 +- drivers/dma/dmaengine.c | 44 +++++- drivers/dma/of-dma.c | 12 +- drivers/dma/tegra20-apb-dma.c | 49 +++++- drivers/gpu/drm/tegra/Kconfig | 1 + drivers/gpu/drm/tegra/dc.c | 9 +- drivers/gpu/drm/tegra/drm.h | 3 + drivers/gpu/drm/tegra/gr3d.c | 22 ++- drivers/gpu/drm/tegra/hdmi.c | 14 +- drivers/i2c/busses/i2c-tegra.c | 13 +- drivers/input/keyboard/tegra-kbc.c | 13 +- drivers/pci/host/pci-tegra.c | 52 +++++-- drivers/spi/Kconfig | 3 + drivers/spi/spi-tegra114.c | 66 ++++---- drivers/spi/spi-tegra20-sflash.c | 18 ++- drivers/spi/spi-tegra20-slink.c | 66 ++++---- drivers/staging/nvec/nvec.c | 11 +- drivers/staging/nvec/nvec.h | 5 +- drivers/tty/serial/serial-tegra.c | 86 +++++------ drivers/usb/host/ehci-tegra.c | 14 +- include/dt-bindings/clock/tegra20-car.h | 2 +- include/dt-bindings/clock/tegra30-car.h | 2 +- include/linux/clk/tegra.h | 7 - include/linux/dmaengine.h | 7 + include/linux/of_dma.h | 9 +- include/linux/tegra-powergate.h | 4 +- include/sound/dmaengine_pcm.h | 6 + sound/soc/soc-generic-dmaengine-pcm.c | 82 +++++++--- sound/soc/tegra/Kconfig | 2 + sound/soc/tegra/tegra20_ac97.c | 11 -- sound/soc/tegra/tegra20_i2s.c | 20 +-- sound/soc/tegra/tegra30_ahub.c | 125 +++++++++------ sound/soc/tegra/tegra30_ahub.h | 11 +- sound/soc/tegra/tegra30_i2s.c | 97 ++++++------ sound/soc/tegra/tegra30_i2s.h | 3 + sound/soc/tegra/tegra_pcm.c | 17 +- sound/soc/tegra/tegra_pcm.h | 5 + 81 files changed, 1448 insertions(+), 558 deletions(-) -- 1.8.1.5

7 38

[alsa-devel] [PATCH] ASoC: dmaengine: fix deferred probe detection
by Stephen Warren 11 Dec '13

11 Dec '13

Check the return value of dma_request_slave_channel_reason() to see if deferred probe happens, not the variable the return value will be assigned to later. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Fixes: 5eda87b890f8 ("ASoC: dmaengine: support deferred probe for DMA channels") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 585eaa69e8c3..7483922f6ee3 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -321,7 +321,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = config->chan_names[i]; chan = dma_request_slave_channel_reason(dev, name); if (IS_ERR(chan)) { - if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER) + if (PTR_ERR(chan) == -EPROBE_DEFER) return -EPROBE_DEFER; pcm->chan[i] = NULL; } else { -- 1.8.1.5

2 1

[alsa-devel] [PATCH 1/2 v2] crec: Initial version of a compressed capture utility
by Charles Keepax 11 Dec '13

11 Dec '13

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Change-Id: I79b0f22598b3ea3b8e99865804799dd013e9d02b Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Use memset rather than manually clear snd_codec fields - Change fopen to "wb" rather than "w+b" I haven't changed the control switch letters, but if there are strong feelings on which letter to use I am happy to update this. Thanks, Charles Android.mk | 10 ++ crec.c | 417 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 440 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..2976df3 --- /dev/null +++ b/crec.c @@ -0,0 +1,417 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + FILE *file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = fopen(name, "wb"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error writing output file header: %d\n", + ferror(file)); + goto file_exit; + } + + memset(&codec, 0, sizeof(codec)); + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = compress_get_alsa_rate(rate); + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = fwrite(buffer, 1, read, file); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %d\n", + ferror(file)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = fseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", stderror(errno)); + goto buf_exit; + } + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error updating output file header: %d\n", + ferror(file)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + fclose(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + fclose(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

3 7

[alsa-devel] [PATCH V3] ASoC: dmaengine: support deferred probe for DMA channels
by Stephen Warren 11 Dec '13

11 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Enhance dmaengine_pcm_request_chan_of() to support deferred probe for DMA channels, by using the new dma_request_slave_channel_or_err() API. This prevents snd_dmaengine_pcm_register() from succeeding without acquiring DMA channels due to the relevant DMA controller not yet being registered. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Mark, this is the patch which requires the dependency from the DMA tree, i.e. what I mentioned in the pull request that I sent just a little while ago. You'd previously ack'd it so I could take it through the Tegra tree, but it's probably better if instead you put it into ASoC tree first, then I pull your (topic/dma?) branch into the Tegra tree. (I've validated that your current topic/dma, plus a merge of the DMA tree's topic/defer_probe, plus this patch, plus a merge of next-20131210 to pick up the rest of ASoC etc., runs without conflicts and builds at least for Tegra without issues). v3: * Fix missing braces around else block. v2: * Call dmaengine_pcm_release_chan() if dmaengine_pcm_request_chan_of() fails, to clean up any partial allocations. --- sound/soc/soc-generic-dmaengine-pcm.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5b70c556fba3..585eaa69e8c3 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -287,16 +287,17 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; -static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, +static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev, const struct snd_dmaengine_pcm_config *config) { unsigned int i; const char *name; + struct dma_chan *chan; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) - return; + return 0; if (config->dma_dev) { /* @@ -318,13 +319,22 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = dmaengine_pcm_dma_channel_names[i]; if (config->chan_names[i]) name = config->chan_names[i]; - pcm->chan[i] = dma_request_slave_channel(dev, name); + chan = dma_request_slave_channel_reason(dev, name); + if (IS_ERR(chan)) { + if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER) + return -EPROBE_DEFER; + pcm->chan[i] = NULL; + } else { + pcm->chan[i] = chan; + } if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) break; } if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) pcm->chan[1] = pcm->chan[0]; + + return 0; } static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) @@ -360,7 +370,9 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev, config); + ret = dmaengine_pcm_request_chan_of(pcm, dev, config); + if (ret) + goto err_free_dma; if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) ret = snd_soc_add_platform(dev, &pcm->platform, -- 1.8.1.5

2 1

[alsa-devel] [GIT PULL] ASoC: DMA deferred probe dependencies
by Stephen Warren 11 Dec '13

11 Dec '13

Mark, The following branch/patch is a dependency for my "ASoC: dmaengine: support deferred probe for DMA channels". If you could pull the branch below into your topic/dma, then merge my ASoC patch, that would be great. I would then like to merge your topic/dma into the Tegra tree as a baseline for the Tegra conversion to the standard DMA DT bindings. Vinod has confirmed this his topic/defer_probe branch is stable, and won't be rebased: https://lkml.org/lkml/2013/12/10/463 Re: [PATCH V4] dma: add channel request API that supports deferred probe Thanks. ---------------------------------------------------------------- The following changes since commit 6ce4eac1f600b34f2f7f58f9cd8f0503d79e42ae: Linux 3.13-rc1 are available in the git repository at: git://git.infradead.org/users/vkoul/slave-dma.git topic/defer_probe for you to fetch changes up to 0ad7c00057dc1640647c1dc81ccbd009de17a767: dma: add channel request API that supports deferred probe ---------------------------------------------------------------- Stephen Warren (1): dma: add channel request API that supports deferred probe drivers/dma/dmaengine.c | 35 +++++++++++++++++++++++++++++++---- drivers/dma/of-dma.c | 15 +++++++++------ include/linux/dmaengine.h | 8 ++++++++ 3 files changed, 48 insertions(+), 10 deletions(-)

2 4

[alsa-devel] [PATCH] ASoC: fsi: remove original filter from fsi_dma_probe()
by Kuninori Morimoto 11 Dec '13

11 Dec '13

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Remove original filter from fsi_dma_probe(), and use SH-DMA suitable filter. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b33ca7c..6101055 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -232,9 +232,9 @@ struct fsi_stream { * these are for DMAEngine */ struct dma_chan *chan; - struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int dma_id; int loop_cnt; int additional_pos; }; @@ -1410,15 +1410,6 @@ static void fsi_dma_do_work(struct work_struct *work) } } -static bool fsi_dma_filter(struct dma_chan *chan, void *param) -{ - struct sh_dmae_slave *slave = param; - - chan->private = slave; - - return true; -} - static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { schedule_work(&io->work); @@ -1446,15 +1437,34 @@ static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; + int is_play = fsi_stream_is_play(fsi, io); dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); + io->chan = dma_request_slave_channel_compat(mask, + shdma_chan_filter, (void *)io->dma_id, + dev, is_play ? "tx" : "rx"); + if (io->chan) { + struct dma_slave_config cfg; + int ret; + + cfg.slave_id = io->dma_id; + cfg.dst_addr = 0; /* use default addr */ + cfg.src_addr = 0; /* use default addr */ + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + ret = dmaengine_slave_config(io->chan, &cfg); + if (ret < 0) { + dma_release_channel(io->chan); + io->chan = NULL; + } + } + if (!io->chan) { /* switch to PIO handler */ - if (fsi_stream_is_play(fsi, io)) + if (is_play) fsi->playback.handler = &fsi_pio_push_handler; else fsi->capture.handler = &fsi_pio_pop_handler; @@ -1960,7 +1970,7 @@ static void fsi_handler_init(struct fsi_priv *fsi, fsi->capture.priv = fsi; if (info->tx_id) { - fsi->playback.slave.shdma_slave.slave_id = info->tx_id; + fsi->playback.dma_id = info->tx_id; fsi->playback.handler = &fsi_dma_push_handler; } } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/8][RFC] a driver for Fireworks based devices
by Takashi Sakamoto 11 Dec '13

11 Dec '13

This series of patch is update of my previous one: [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062614.html This series of patch is based on this series: [RFC][PATCH 00/17] Enhancement for firewire-lib http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069163.h… I hope to get comments, especially for hwdep interface for Fireworks command/response which was discussed below: include/uapi/firewire.h for other firewire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2013-October/067743.ht… Takashi Sakamoto (8): fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 19 + sound/firewire/Kconfig | 15 + sound/firewire/Makefile | 1 + sound/firewire/fireworks/Makefile | 5 + sound/firewire/fireworks/fireworks.c | 344 +++++++++++++++++ sound/firewire/fireworks/fireworks.h | 239 ++++++++++++ sound/firewire/fireworks/fireworks_command.c | 401 ++++++++++++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 310 +++++++++++++++ sound/firewire/fireworks/fireworks_midi.c | 176 +++++++++ sound/firewire/fireworks/fireworks_pcm.c | 459 +++++++++++++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 216 +++++++++++ sound/firewire/fireworks/fireworks_stream.c | 364 ++++++++++++++++++ sound/firewire/fireworks/fireworks_transaction.c | 340 +++++++++++++++++ 14 files changed, 2891 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

1 8

[alsa-devel] [PATCH] ALSA: hda - Add static DAC/pin mapping for AD1986A codec
by Takashi Iwai 11 Dec '13

11 Dec '13

AD1986A codec is a pretty old codec and has really many hidden restrictions. One of such is that each DAC is dedicated to certain pin although there are possible connections. Currently, the generic parser tries to assign individual DACs as much as possible, and this lead to two bad situations: connections where the sound actually doesn't work, and connections conflicting other channels. We may fix this by trying to find the best connections more harder, but as of now, it's easier to give some hints for paired DAC/pin connections and honor them if available, since such a hint is needed only for specific codecs (right now only AD1986A, and there will be unlikely any others in future). Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 23 ++++++++++++++++++++++- sound/pci/hda/hda_generic.h | 3 +++ sound/pci/hda/patch_analog.c | 10 ++++++++++ 3 files changed, 35 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 249cd4615908..c7f6d1cab606 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -474,6 +474,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx) memset(path, 0, sizeof(*path)); } +/* return a DAC if paired to the given pin by codec driver */ +static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + const hda_nid_t *list = spec->preferred_dacs; + + if (!list) + return 0; + for (; *list; list += 2) + if (*list == pin) + return list[1]; + return 0; +} + /* look for an empty DAC slot */ static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin, bool is_digital) @@ -1192,7 +1206,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, continue; } - dacs[i] = look_for_dac(codec, pin, false); + dacs[i] = get_preferred_dac(codec, pin); + if (dacs[i]) { + if (is_dac_already_used(codec, dacs[i])) + badness += bad->shared_primary; + } + + if (!dacs[i]) + dacs[i] = look_for_dac(codec, pin, false); if (!dacs[i] && !i) { /* try to steal the DAC of surrounds for the front */ for (j = 1; j < num_outs; j++) { diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 7e45cb44d151..0929a06df812 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -249,6 +249,9 @@ struct hda_gen_spec { const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; + /* preferred pin/DAC pairs; an array of paired NIDs */ + const hda_nid_t *preferred_dacs; + /* loopback mixing mode */ bool aamix_mode; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cac015be3325..699262a3e07a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -340,6 +340,14 @@ static int patch_ad1986a(struct hda_codec *codec) { int err; struct ad198x_spec *spec; + static hda_nid_t preferred_pairs[] = { + 0x1a, 0x03, + 0x1b, 0x03, + 0x1c, 0x04, + 0x1d, 0x05, + 0x1e, 0x03, + 0 + }; err = alloc_ad_spec(codec); if (err < 0) @@ -360,6 +368,8 @@ static int patch_ad1986a(struct hda_codec *codec) * So, let's disable the shared stream. */ spec->gen.multiout.no_share_stream = 1; + /* give fixed DAC/pin pairs */ + spec->gen.preferred_dacs = preferred_pairs; /* AD1986A can't manage the dynamic pin on/off smoothly */ spec->gen.auto_mute_via_amp = 1; -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - One more Dell headset detection quirk
by Hui Wang 11 Dec '13

11 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280624, no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. BugLink: https://bugs.launchpad.net/bugs/1259790 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This patch was generated against the master branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 15300d3..c5f3e1b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5096,6 +5096,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - hdmi: Fix IEC958 ctl indexes for some simple HDMI devices
by Anssi Hannula 11 Dec '13

11 Dec '13

In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF outputs will have their IEC958 controls created starting from index 16 and the HDMI controls will be created starting from index 0. However, HDMI simple_playback_build_controls() as used by old VIA and NVIDIA codecs incorrectly requests the IEC958 controls to be created with an S/PDIF type instead of HDMI. In case the card has other codecs that have HDMI outputs, the controls will be created with wrong index=16, causing them to e.g. be unreachable by the ALSA "hdmi" alias. Fix that by making simple_playback_build_controls() request controls with HDMI indexes. Not many cards have an affected configuration, but e.g. ASUS M3N78-VM contains an integrated NVIDIA HDA "card" with: - a VIA codec that has, among others, an S/PDIF pin incorrectly labelled as an HDMI pin, and - an NVIDIA MCP7x HDMI codec. Reported-by: MysterX on #openelec Tested-by: MysterX on #openelec Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> # 3.8+ --- sound/pci/hda/patch_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 556d47231f11..f5060fc7c303 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2326,8 +2326,9 @@ static int simple_playback_build_controls(struct hda_codec *codec) int err; per_cvt = get_cvt(spec, 0); - err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid, - per_cvt->cvt_nid); + err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid, + per_cvt->cvt_nid, + HDA_PCM_TYPE_HDMI); if (err < 0) return err; return simple_hdmi_build_jack(codec, 0); -- 1.8.1.5

2 1

[alsa-devel] fading in/out PCM audio
by Mat Gros 10 Dec '13

10 Dec '13

Hi, I'm using ALSA on Pandaboard. Is it possible to activate fading in/out ramps via the ALSA API? Greetings, Mat

2 1

[alsa-devel] [PATCH] ALSA: at73c213: clk_round_rate() can return a zero upon error
by Paul Walmsley 10 Dec '13

10 Dec '13

Treat both negative and zero return values from clk_round_rate() as errors. This is needed since subsequent patches will convert clk_round_rate()'s return value to be an unsigned type, rather than a signed type, since some clock sources can generate rates higher than (2^31)-1 Hz. Eventually, when calling clk_round_rate(), only a return value of zero will be considered a error. All other values will be considered valid rates. The comparison against values less than 0 is kept to preserve the correct behavior in the meantime. Signed-off-by: Paul Walmsley <pwalmsley(a)nvidia.com> Cc: Hans-Christian Egtvedt <egtvedt(a)samfundet.no> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- Applies on v3.13-rc3. See also: http://marc.info/?l=linux-arm-kernel&m=138542591313620&w=2 sound/spi/at73c213.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 8e3d9a6c7a3b..25c38afaee49 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -174,7 +174,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) dac_rate_new = 8 * (ssc_rate / ssc_div); status = clk_round_rate(chip->board->dac_clk, dac_rate_new); - if (status < 0) + if (status <= 0) return status; /* Ignore difference smaller than 256 Hz. */

3 4

[alsa-devel] [PATCH] ALSA: hda - Mute all aamix inputs as default
by Takashi Iwai 10 Dec '13

10 Dec '13

Not all channels have been initialized, so far, especially when aamix NID itself doesn't have amps but its leaves have. This patch fixes these holes. Otherwise you might get unexpected loopback inputs, e.g. from surround channels. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c4671d00babd..249cd4615908 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4297,6 +4297,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return AC_PWRST_D3; } +/* mute all aamix inputs initially; parse up to the first leaves */ +static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) +{ + int i, nums; + const hda_nid_t *conn; + bool has_amp; + + nums = snd_hda_get_conn_list(codec, mix, &conn); + has_amp = nid_has_mute(codec, mix, HDA_INPUT); + for (i = 0; i < nums; i++) { + if (has_amp) + snd_hda_codec_amp_stereo(codec, mix, + HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); + else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) + snd_hda_codec_amp_stereo(codec, conn[i], + HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); + } +} /* * Parse the given BIOS configuration and set up the hda_gen_spec @@ -4435,6 +4455,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* mute all aamix input initially */ + if (spec->mixer_nid) + mute_all_mixer_nid(codec, spec->mixer_nid); + dig_only: parse_digital(codec); -- 1.8.5

1 0

[alsa-devel] [PATCH V2 REPOST] ASoC: dmaengine: support deferred probe for DMA channels
by Stephen Warren 10 Dec '13

10 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Enhance dmaengine_pcm_request_chan_of() to support deferred probe for DMA channels, by using the new dma_request_slave_channel_or_err() API. This prevents snd_dmaengine_pcm_register() from succeeding without acquiring DMA channels due to the relevant DMA controller not yet being registered. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Mark, this is the patch which requires the dependency from the DMA tree, i.e. what I mentioned in the pull request that I sent just a little while ago. You'd previously ack'd it so I could take it through the Tegra tree, but it's probably better if instead you put it into ASoC tree first, then I pull your (topic/dma?) branch into the Tegra tree. (I've validated that your current topic/dma, plus a merge of the DMA tree's topic/defer_probe, plus this patch, plus a merge of next-20131210 to pick up the rest of ASoC etc., runs without conflicts and builds at least for Tegra without issues). v2: * Call dmaengine_pcm_release_chan() if dmaengine_pcm_request_chan_of() fails, to clean up any partial allocations. sound/soc/soc-generic-dmaengine-pcm.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5b70c556fba3..053c0fa9740d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -287,16 +287,17 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; -static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, +static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev, const struct snd_dmaengine_pcm_config *config) { unsigned int i; const char *name; + struct dma_chan *chan; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) - return; + return 0; if (config->dma_dev) { /* @@ -318,13 +319,21 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = dmaengine_pcm_dma_channel_names[i]; if (config->chan_names[i]) name = config->chan_names[i]; - pcm->chan[i] = dma_request_slave_channel(dev, name); + chan = dma_request_slave_channel_reason(dev, name); + if (IS_ERR(chan)) { + if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER) + return -EPROBE_DEFER; + pcm->chan[i] = NULL; + } else + pcm->chan[i] = chan; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) break; } if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) pcm->chan[1] = pcm->chan[0]; + + return 0; } static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) @@ -360,7 +369,9 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev, config); + ret = dmaengine_pcm_request_chan_of(pcm, dev, config); + if (ret) + goto err_free_dma; if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) ret = snd_soc_add_platform(dev, &pcm->platform, -- 1.8.1.5

2 2

[alsa-devel] [PATCH 0/5] ASoC: remove dmaengine compat_request_channel
by Stephen Warren 10 Dec '13

10 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This series removes almost all users of the ASoC dmaengine compat_request_channel() function, and its implementation. The ux500 driver still uses compat_request_channel. I believe that Lars-Peter is working on removing its use from that driver. That patch or series would need to be inserted between 4/5 and 5/5 in this series. This series is based on ASoC's topic/dma branch. I have only compile-tested these patches. Stephen Warren (5): ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device ASoC: ep93xx: remove custom DMA alloc compat function ASoC: SPEAr: get rid of spear-pcm-audio struct device ASoC: SPEAr: remove custom DMA alloc compat function ASoC: dmaengine: remove compat_request_channel include/sound/dmaengine_pcm.h | 3 --- include/sound/spear_dma.h | 1 - sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 21 ++++++++++++++++-- sound/soc/cirrus/ep93xx-i2s.c | 23 +++++++++++++++++-- sound/soc/cirrus/ep93xx-pcm.c | 39 +++++++------------------------- sound/soc/cirrus/ep93xx-pcm.h | 24 ++++++++++++++++++++ sound/soc/cirrus/simone.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 3 --- sound/soc/spear/spdif_in.c | 25 +++++++++++++++++---- sound/soc/spear/spdif_out.c | 26 ++++++++++++++++++---- sound/soc/spear/spear_pcm.c | 42 ++++++++++------------------------- sound/soc/spear/spear_pcm.h | 26 ++++++++++++++++++++++ 14 files changed, 156 insertions(+), 83 deletions(-) create mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/spear/spear_pcm.h -- 1.8.1.5

5 24

[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 10 Dec '13

10 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- This will need merging with the fix patch I sent earlier, I'l do that assuming this is OK. sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 7 +++++-- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 42 insertions(+), 45 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23a9893..4a88e36c82ec 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_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_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_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_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b494d4..ee23194f7ab8 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315a3eaf..fb09a1c5f75b 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..eab0050d4579 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256fc4b2c..6a5e4bf6ac96 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.5.1

2 10

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 10 Dec '13

10 Dec '13

> Codec: ATI R6xx HDMI >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x1002aa01 >> Subsystem Id: 0x00aa0100 >> Revision Id: 0x100300 >> >> seem that your codec may get Audio-Video Association Port ID and Sink >> Information for your DENON-AVAMP >> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Revision ID 3 and later (0x100300 as reported by procfs codec#X) have support for providing more information than the previous revisions (but only if supported by the display driver). manufacture_id 0x0 product_id 0x0 port_id 0x0 you have to find out those values return by hda_codec_read ATI_VERB_GET_SINK_INFO_DATA as your codec are revision 3 and your graphic driver got the product_id, manufacture_id, port_id Manufacturer: DON Model 26 Serial Number 16843009 Made week 0 of 2011 Monitor name: DENON Audio data block Linear PCM, max channels 7 Supported sample rates (kHz): 192 176.4 96 88.2 48 44.1 32 Supported sample sizes (bits): 24 20 16 DTS, max channels 5 Supported sample rates (kHz): 96 88.2 48 44.1 Maximum bit rate: 1016 kHz AC-3, max channels 5 Supported sample rates (kHz): 48 44.1 32 Maximum bit rate: 1016 kHz DTS-HD, max channels 7 Supported sample rates (kHz): 192 176.4 96 88.2 48 44.1 Dolby Digital+, max channels 7 Supported sample rates (kHz): 48 44.1 MAT (MLP), max channels 7 Supported sample rates (kHz): 192 176.4 96 88.2 48 44.1 Speaker allocation data block Do those SAD in EDID in xorg.log match with the specification of your Denon HDMI receiver ? speakers [0x5f] FL/FR LFE FC RL/RR RC RLC/RRC sad_count 6 sad0_coding_type [0x1] LPCM sad0_channels 8 sad0_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 sad0_bits [0xe0000] 16 20 24 sad1_coding_type [0x2] AC-3 sad1_channels 6 sad1_rates [0xe0] 32000 44100 48000 sad1_max_bitrate 640000 sad2_coding_type [0x7] DTS sad2_channels 6 sad2_rates [0x6c0] 44100 48000 88200 96000 sad2_max_bitrate 1536000 sad3_coding_type [0xa] E-AC-3/DD+ (Dolby Digital Plus) sad3_channels 8 sad3_rates [0xc0] 44100 48000 sad4_coding_type [0xb] DTS-HD sad4_channels 8 sad4_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 sad5_coding_type [0xc] MLP (Dolby TrueHD) sad5_channels 8 sad5_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 seem emulated ELD 's max bitrate are different from those in SAD of EDID Do AC3 or DTS passthrough hdmi work (using alsa and pulseaudio) or not ? Do your Denon HDMI receiver support 7.1 ? Profiles: output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 400) output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3300) output:hdmi-stereo-extra1: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 200) output:hdmi-surround-extra1: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3100) output:hdmi-stereo-extra2: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 200) output:hdmi-surround-extra2: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3100) output:hdmi-stereo-extra3: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 200) output:hdmi-surround-extra3: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 3100) seem no 7.1 in pulseaudio ? Do pulseaudio select the first HDMI as active port the default sink ? I am not sure. > seem no output of pactl stat http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/modules/alsa/mi… you may comment the last three hdmi devices until monitor_presence bug is fixed

2 1

[alsa-devel] [PATCH] sound: memalloc.h - fix wrong truncation of dma_addr_t
by Stefano Panella 10 Dec '13

10 Dec '13

When running a 32bit kernel the hda_intel driver is still reporting a 64bit dma_mask if the HW supports it. >From sound/pci/hda/hda_intel.c: /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); else { pci_set_dma_mask(pci, DMA_BIT_MASK(32)); pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)); } which means when there is a call to dma_alloc_coherent from snd_malloc_dev_pages a machine address bigger than 32bit can be returned. This can be true in particular if running the 32bit kernel as a pv dom0 under the Xen Hypervisor or PAE on bare metal. The problem is that when calling setup_bdle to program the BLE the dma_addr_t returned from the dma_alloc_coherent is wrongly truncated from snd_sgbuf_get_addr if running a 32bit kernel: static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, size_t offset) { struct snd_sg_buf *sgbuf = dmab->private_data; dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr; addr &= PAGE_MASK; return addr + offset % PAGE_SIZE; } where PAGE_MASK in a 32bit kernel is zeroing the upper 32bit af addr. Without this patch the HW will fetch the 32bit truncated address, which is not the one obtained from dma_alloc_coherent and will result to a non working audio but can corrupt host memory at a random location. The current patch apply to v3.13-rc3-74-g6c843f5 Signed-off-by: Stefano Panella <stefano.panella(a)citrix.com> Reviewed-by: Frediano Ziglio <frediano.ziglio(a)citrix.com> --- include/sound/memalloc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index af99839..5f73785 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -108,7 +108,7 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, { struct snd_sg_buf *sgbuf = dmab->private_data; dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr; - addr &= PAGE_MASK; + addr &= ~((dma_addr_t)PAGE_SIZE - 1); return addr + offset % PAGE_SIZE; } -- 1.7.9.5

2 1

[alsa-devel] [PATCHv8][ 1/5] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 10 Dec '13

10 Dec '13

Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v7->v8: - fixed the vendor prefixes in the dts bindings. - updated the code and the documentation accordingly. - removed the need of an audio-codec property, since it's available in the ssi bindings. - the dai codec is now retrived trough the dt in order to support machines that have its codec on another i2c bus number. - driver compatible is now similar to the ones of the other drivers. ChangeLog v6->v7: - The cleanups went into another patch. - The support for fsl_ssi doesn't depend on dt anymore. - platform_name = "imx-ssi.0" is still set when in non-dt mode. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 108 +++++++++++++++++--- 3 files changed, 121 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..0d7985c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,21 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,asoc-tlv320" +- eukrea,model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- fsl,mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 5983740..eb093d5 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -15,8 +15,11 @@ * */ +#include <linux/errno.h> #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,6 +29,7 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 @@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) { + /* fsl_ssi lacks the set_fmt ops. */ + if (ret && ret != -ENOTSUPP) { dev_err(cpu_dai->dev, "Failed to set the cpu dai format.\n"); return ret; @@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, "Failed to set the codec sysclk.\n"); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); - if (ret) { + /* fsl_ssi lacks the set_sysclk ops */ + if (ret && ret != -EINVAL) { dev_err(cpu_dai->dev, "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; @@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", - .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, }; static struct snd_soc_card eukrea_tlv320 = { - .name = "cpuimx-audio", .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, @@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; - if (machine_is_eukrea_cpuimx27()) { + eukrea_tlv320.dev = &pdev->dev; + if (np) { + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); + if (ret) { + dev_err(&pdev->dev, + "eukrea,model node missing or invalid.\n"); + goto err; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + if (!ssi_np) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + if (codec_np) + eukrea_tlv320_dai.codec_of_node = codec_np; + else + dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-int-port node missing or invalid.\n"); + return ret; + } + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-ext-port node missing or invalid.\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_of_node = ssi_np; + eukrea_tlv320_dai.platform_of_node = ssi_np; + } else { + eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0"; + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a"; + eukrea_tlv320.name = "cpuimx-audio"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,asoc-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing or invalid audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } - eukrea_tlv320.dev = &pdev->dev; ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,asoc-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove, -- 1.7.9.5

3 6

[alsa-devel] [PATCH v2 0/2] Enhance imx-wm8962 machine driver
by Nicolin Chen 10 Dec '13

10 Dec '13

Although there's no direct relationship between any of these two patches, I recommend to apply them in order so as to circumvent merging conflicts. Changlog: v2: * Nothing changed for PATCH-1. * Added spinlock potection for boolean fll_locked in PATCH-2 * Dropped PATCH-3, will try to revise it next time. Nicolin Chen (2): ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver ASoC: fsl: imx-wm8962: Grant hw_params/free() permission to control FLL sound/soc/fsl/imx-wm8962.c | 152 +++++++++++++++++++++++++++++---------------- 1 file changed, 99 insertions(+), 53 deletions(-) -- 1.8.4

3 10

[alsa-devel] caiaq driver: audio 4 dj and jackd
by Gabriel M. Beddingfield 10 Dec '13

10 Dec '13

Is it possible to make all 4 ins/outs in the Audio 4 DJ available through the first substream device? A user was having trouble using the NI Audio 4 DJ device with jackd. When jack connected, this 4-in/4-out device would appear as 2-in/2-out. This was the result of the fact that the channels are split up into stereo substreams. I.e. the first two channels are found in hw:x,0,0 and the last two channels are in hw:x,0,1. The work-around was to use the `multi' plug-in to make these 2 PCM's appear as a single PCM. There's several situations where this is a Good Thing, because it makes it easy for the user to treat the streams independently. BUT... in the pro-audio case this is not a good thing at all. Ideally, substream 0 would support 1, 2, or 4 channels (for both playback and capture). When substream 0 is using more than 2 channels, then substream 1 would return EBUSY when in use. So... would it be possible to do this in the caiaq driver? -gabriel

3 9

[alsa-devel] [PATCH] ASoC: rsnd: gen: fixup Gen2 channel size
by Kuninori Morimoto 10 Dec '13

10 Dec '13

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Gen2 has 0 - 9, total 10 channels, not 9 channels. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/gen.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index bf066f7..d0ab203 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -217,11 +217,11 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, /* single address mapping */ #define RSND_GEN2_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9) + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10) /* multi address mapping */ #define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9) + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10) static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - A Dell headset detection quirk
by Hui Wang 10 Dec '13

10 Dec '13

On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623), no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435 CC: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c5ea483..7122f2c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5034,6 +5034,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), -- 1.8.1.2

2 2

[alsa-devel] [PATCH] ALSA: atmel_abdac: clk_round_rate() can return a zero upon error
by Paul Walmsley 10 Dec '13

10 Dec '13

Treat both negative and zero return values from clk_round_rate() as errors. This is needed since subsequent patches will convert clk_round_rate()'s return value to be an unsigned type, rather than a signed type, since some clock sources can generate rates higher than (2^31)-1 Hz. Eventually, when calling clk_round_rate(), only a return value of zero will be considered a error; all other values will be considered valid rates. The comparison against values less than 0 is kept to preserve the correct behavior in the meantime. Signed-off-by: Paul Walmsley <pwalmsley(a)nvidia.com> Cc: Hans-Christian Egtvedt <egtvedt(a)samfundet.no> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- Applies on v3.13-rc3. See also: http://marc.info/?l=linux-arm-kernel&m=138542591313620&w=2 sound/atmel/abdac.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 721d8fd..3519518 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -354,7 +354,7 @@ static int set_sample_rates(struct atmel_abdac *dac) /* we start at 192 kHz and work our way down to 5112 Hz */ while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); - if (new_rate < 0) + if (new_rate <= 0) break; /* make sure we are below the ABDAC clock */ if (index < MAX_NUM_RATES &&

2 1

Re: [alsa-devel] [PATCH V2] ASoC: add RT286 CODEC driver
by Bard Liao 10 Dec '13

10 Dec '13

> -----Original Message----- > From: Leon Romanovsky [mailto:leon@leon.nu] > Sent: Wednesday, December 04, 2013 10:37 PM > To: Bard Liao > Cc: Mark Brown; Liam Girdwood; Oder Chiou; alsa-devel(a)alsa-project.org; > Flove > Subject: Re: [alsa-devel] [PATCH V2] ASoC: add RT286 CODEC driver > > On Wed, Dec 4, 2013 at 3:41 PM, <bardliao(a)realtek.com> wrote: > > From: Bard Liao <bardliao(a)realtek.com> > > > > This patch adds the ALC286 codec driver. > > > > Signed-off-by: Bard Liao <bardliao(a)realtek.com> > > > > Please add change log while you are sending verX patch. Please see the change log as below. * Add platform data to determine if combo jack and/or IRQ is using or not. * Implement rt286_hw_read/write in regmap. * Configure codec in runtime instead of in probe. * Other minor changes (coding style, remove version control, remove empty functions, share code ...) Should I send another patch with these change log? > > > > + > > +static unsigned int rt286_reg_cache[TOTAL_NODE_ID + 1] = { > You don't need TOTAL_NODE_ID + 1 in array declaration. Without "TOTAL_NODE_ID + 1", I should always set [NODE_ID_MIXER_IN2] = 0x0000, right? Ie. Assign the latest array item a value. > > > + [NODE_ID_DAC_OUT1] = 0x7f7f, > > + [NODE_ID_DAC_OUT2] = 0x7f7f, > > + [NODE_ID_SPDIF] = 0x0000, > > + [NODE_ID_ADC_IN1] = 0x4343, > > + [NODE_ID_ADC_IN2] = 0x4343, > > + [NODE_ID_MIC1] = 0x0000, > > + [NODE_ID_MIXER_IN] = 0x000b, > > + [NODE_ID_MIXER_OUT1] = 0x0002, > > + [NODE_ID_MIXER_OUT2] = 0x0000, > > + [NODE_ID_SPK_OUT] = 0x0000, > > + [NODE_ID_HP_OUT] = 0x0000, > > + [NODE_ID_MIXER_IN1] = 0x0000, > > + [NODE_ID_MIXER_IN2] = 0x0000, > > +}; Thanks.

2 1

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 10 Dec '13

10 Dec '13

2013/12/9 John Ettedgui <john.ettedgui(a)gmail.com> > > I am using the gitcompile script to configure, compile... > There is no configure by default, I was able to autogen it, but then I had > issues while building so I reverted back to gitcompile > [ 4.841881] ALSA hda_intel.c:3170 0000:01:00.1: Handle VGA-switcheroo audio client [ 4.841885] ALSA hda_intel.c:3364 0000:01:00.1: Using LPIB position fix [ 4.841886] ALSA hda_intel.c:3510 0000:01:00.1: Force to non-snoop mode [ 4.841920] snd_hda_intel 0000:01:00.1: irq 47 for MSI/MSI-X [ 4.841940] ALSA hda_intel.c:3635 0000:01:00.1: chipset global capabilities = 0x0 seem the max number of playback streams is hardcoded in the driver [ 4.842211] [drm] radeon kernel modesetting enabled. [ 4.844508] ALSA hda_intel.c:1244 0000:01:00.1: codec_mask = 0x1 [ 4.844630] ALSA hda_intel.c:1716 0000:01:00.1: codec #0 probed OK [ 4.844632] ALSA hda_intel.c:1837 0000:01:00.1: Enable sync_write for stable communication [ 4.851623] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=0 [ 4.851654] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 4.851685] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=7 Presence_Detect=0 ELD_Valid=0 [ 4.851717] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=9 Presence_Detect=0 ELD_Valid=0 [ 4.851751] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=11 Presence_Detect=0 ELD_Valid=0 [ 4.851787] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=13 Presence_Detect=0 ELD_Valid=0 [ 4.851839] input: HDA ATI HDMI HDMI/DP,pcm=11 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input12 [ 4.851898] input: HDA ATI HDMI HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input11 [ 4.851948] input: HDA ATI HDMI HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input10 [ 4.852014] input: HDA ATI HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input9 [ 4.852075] input: HDA ATI HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input8 [ 4.852129] input: HDA ATI HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input7 Does this mean monitor is plugged and graphic driver is requesting monitor to provide EDID ? 5.340797] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=0 [ 5.340812] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 5.340825] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340838] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340844] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=5 Presence_Detect=1 ELD_Valid=0 [ 5.340857] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=5 Presence_Detect=1 ELD_Valid=1 [ 5.340870] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340882] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340885] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=7 Presence_Detect=1 ELD_Valid=0 [ 5.340898] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=7 Presence_Detect=1 ELD_Valid=1 [ 5.340911] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340923] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340926] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=9 Presence_Detect=1 ELD_Valid=0 [ 5.340939] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=9 Presence_Detect=1 ELD_Valid=1 [ 5.340952] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340965] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340967] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=11 Presence_Detect=1 ELD_Valid=0 [ 5.340980] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=11 Presence_Detect=1 ELD_Valid=1 [ 5.340993] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.341005] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.341008] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=13 Presence_Detect=1 ELD_Valid=0 [ 5.341021] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=13 Presence_Detect=1 ELD_Valid=1 [ 5.341034] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.341046] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.504525] [drm] Radeon Display Connectors [ 5.504527] [drm] Connector 0: [ 5.504528] [drm] DP-1 [ 5.504529] [drm] HPD4 [ 5.504530] [drm] DDC: 0x6530 0x6530 0x6534 0x6534 0x6538 0x6538 0x653c 0x653c [ 5.504531] [drm] Encoders: [ 5.504532] [drm] DFP1: INTERNAL_UNIPHY2 [ 5.504533] [drm] Connector 1: [ 5.504534] [drm] DP-2 [ 5.504535] [drm] HPD5 [ 5.504536] [drm] DDC: 0x6540 0x6540 0x6544 0x6544 0x6548 0x6548 0x654c 0x654c [ 5.504537] [drm] Encoders: [ 5.504538] [drm] DFP2: INTERNAL_UNIPHY2 [ 5.504538] [drm] Connector 2: [ 5.504539] [drm] HDMI-A-1 [ 5.504540] [drm] HPD1 [ 5.504541] [drm] DDC: 0x6550 0x6550 0x6554 0x6554 0x6558 0x6558 0x655c 0x655c [ 5.504542] [drm] Encoders: [ 5.504543] [drm] DFP3: INTERNAL_UNIPHY1 [ 5.504544] [drm] Connector 3: [ 5.504545] [drm] DVI-I-1 [ 5.504545] [drm] HPD6 [ 5.504547] [drm] DDC: 0x6580 0x6580 0x6584 0x6584 0x6588 0x6588 0x658c 0x658c [ 5.504547] [drm] Encoders: [ 5.504548] [drm] DFP4: INTERNAL_UNIPHY [ 5.504549] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 5.640237] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 5.640254] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640267] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640270] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=5 Presence_Detect=1 ELD_Valid=1 [ 5.640282] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640295] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640297] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=7 Presence_Detect=1 ELD_Valid=1 [ 5.640310] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640322] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640324] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=9 Presence_Detect=1 ELD_Valid=1 [ 5.640336] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640349] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640351] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=11 Presence_Detect=1 ELD_Valid=1 [ 5.640363] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640376] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640378] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=13 Presence_Detect=1 ELD_Valid=1 [ 5.640390] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640403] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 7.906] (II) RADEON(0): EDID for output DisplayPort-0 [ 7.920] (II) RADEON(0): EDID for output DisplayPort-1 [ 7.983] (II) RADEON(0): EDID for output HDMI-0 [ 7.983] (II) RADEON(0): Manufacturer: DON Model: 26 Serial#: 16843009 [ 7.983] (II) RADEON(0): Year: 2011 Week: 0 [ 7.983] (II) RADEON(0): EDID Version: 1.3 [ 7.983] (II) RADEON(0): Monitor name: DENON-AVAMP [ 7.983] (II) RADEON(0): Number of EDID sections to follow: 1 [ 7.983] (II) RADEON(0): EDID (in hex): [ 7.983] (II) RADEON(0): 00ffffffffffff0011ee260001010101 [ 7.983] (II) RADEON(0): 00150103800000780adaffa3584aa229 [ 7.983] (II) RADEON(0): 17494b00000001010101010101010101 [ 7.983] (II) RADEON(0): 010101010101023a801871382d40582c [ 7.983] (II) RADEON(0): 4500ba882100001e011d8018711c1620 [ 7.983] (II) RADEON(0): 582c2500ba882100009e000000fc0044 [ 7.983] (II) RADEON(0): 454e4f4e2d4156414d500a20000000fd [ 7.983] (II) RADEON(0): 00173d0f440f000a202020202020010d [ 7.983] (II) RADEON(0): 020335714d9005200403020706010e23 [ 7.983] (II) RADEON(0): 0f24320f7f073d1ec01507505f7e0157 [ 7.983] (II) RADEON(0): 0600677e00835f0000e305030167030c [ 7.983] (II) RADEON(0): 001500b82d011d007251d01e206e2855 [ 7.983] (II) RADEON(0): 00ba882100001e8c0ad08a20e02d1010 [ 7.983] (II) RADEON(0): 3e9600ba88210000188c0ad08a20e02d [ 7.983] (II) RADEON(0): 10103e96000b88210000188c0aa01451 [ 7.983] (II) RADEON(0): f01600267c4300ba88210000980000e7 [ 8.003] (II) RADEON(0): EDID for output DVI-0 [ 8.003] (II) RADEON(0): Output DisplayPort-0 disconnected [ 8.003] (II) RADEON(0): Output DisplayPort-1 disconnected [ 8.003] (II) RADEON(0): Output HDMI-0 connected [ 8.003] (II) RADEON(0): Output DVI-0 disconnected Does this mean the graphic driver received EDID from DENON-AVAMP and pass ELD to audio driver ? [ 8.228364] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=1 [ 8.228380] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 8.228393] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 8.228406] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 8.377652] type=1006 audit(1386558737.773:2): pid=515 uid=0 old auid=4294967295 new auid=1000 old ses=4294967295 new ses=1 res=1 [ 8.520795] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.521113] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.521434] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.521727] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.526625] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 8.526640] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 8.526652] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 8.261] (II) config/udev: Adding drm device (/dev/dri/card0) [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=9 (/dev/input/event8) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=10 (/dev/input/event7) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=11 (/dev/input/event6) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=3 (/dev/input/event11) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=7 (/dev/input/event10) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.263] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=8 (/dev/input/event9) [ 8.263] (II) No input driver specified, ignoring this device. [ 8.263] (II) This device may have been added with another device file. [ 8.263] (II) config/udev: Adding input device Logitech USB Receiver (/dev/input/event0)

5 6

[alsa-devel] [PATCH V2 1/3] ASoC: restructure dmaengine_pcm_request_chan_of()
by Stephen Warren 09 Dec '13

09 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Restructure the internals of dmaengine_pcm_request_chan_of() as a loop over all channels to be allocated. This makes it easier to add logic that applies to all allocated channels, without having to duplicate that logic in each of the half-duplex/full-duplex paths. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2 (new as a separate patch): * Re-ordered patches, so that patch 3/3 in this series could be available in the ASoC tree without having to wait for the patches to the DMA tree the implement deferred probe during channel allocation. This whole series is a dependency for some patches that will go through the Tegra tree. If it could be applied to a topic branch in the ASoC tree that would be great. the overall dependency structure will be: a) This series. Can be applied on top of e.g. v3.13-rc1. b) Patches to the DMA tree, which implement deferred probe. Can be applied on top of e.g. v3.13-rc1. c) ASoC modifications to support deferred probe during DMA ch allocation This will depend on both (a) and (b). d) Tegra modifications that depend on (c). --- sound/soc/soc-generic-dmaengine-pcm.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96ce1f4..88b011b80058 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -288,21 +288,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev) { unsigned int i; + const char *name; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) return; - if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); - pcm->chan[1] = pcm->chan[0]; - } else { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = dma_request_slave_channel(dev, - dmaengine_pcm_dma_channel_names[i]); - } + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; + i++) { + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + name = "rx-tx"; + else + name = dmaengine_pcm_dma_channel_names[i]; + pcm->chan[i] = dma_request_slave_channel(dev, name); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; } + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + pcm->chan[1] = pcm->chan[0]; } /** -- 1.8.1.5

3 7

Re: [alsa-devel] [PATCH v4] ASoC: tlv320aic3x: no mono controls 3007 model
by Mark Brown 09 Dec '13

09 Dec '13

On Thu, Dec 05, 2013 at 09:54:02AM +0100, Jan Weitzel wrote: > if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite > registers for class-d amplifier. > classd amplifier controls are only used for AIC3X_MODEL_3007. Applied, thanks.

1 0

[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 09 Dec '13

09 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 38 insertions(+), 44 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23..4a88e36 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_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_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_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_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4..67d9fa9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,7 +963,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.4.3

2 15

[alsa-devel] [PATCH v2 1/4] dt: Add bindings documentation for the ADI AXI-I2S controller
by Lars-Peter Clausen 09 Dec '13

09 Dec '13

This patch adds the devicetree documentation for the ADI AXI-SPDIF audio controller. The controller has: * One set of memory mapped register * Two clocks, one for the memory mapped register interface, one used as the audio reference clock * One DMA interface each for the transmit and receive data Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org --- Changes since v1: * Use <&a ...>, <&b ...> instead of <&a ..., &b ...> in the examples --- .../devicetree/bindings/sound/adi,axi-i2s.txt | 31 ++++++++++++++++++++++ 1 file changed, 31 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,axi-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 0000000..5875ca4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,31 @@ +ADI AXI-I2S controller + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + i2s: i2s@0x77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>, <&ps7_dma 1>; + dma-names = "tx", "rx"; + }; -- 1.8.0

2 4

[alsa-devel] [PATCH] ASoC: tegra: fix uninitialized variables in set_fmt
by Stephen Warren 09 Dec '13

09 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case, "val" is never assigned to, but left uninitialized. The other case does initialized it. Fix this by initializing val at the start of the function, and only ever ORing into it. Update the handling of "mask" so it works the same way for consistency. Update tegra20_spdif.c to use the same code-style for consistency, even though it doesn't happen to suffer from the same problem at present. Cc: <stable(a)vger.kernel.org> Fixes: 0f163546a772 ("ASoC: tegra: use regmap more directly") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 +++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 364bf6a907e1..8c819f811470 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 08bc6931c7c7..8c7c1028e579 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 231a785b3921..02247fee1cf7 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; -- 1.8.1.5

3 2

[alsa-devel] [PATCH 0/3] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 09 Dec '13

09 Dec '13

Hi, This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). * BYTE: Same as BURST but with a byte level granularity. I've only included the last one for completeness and I'm not sure if we really need it. Even if the hardware is able to report the amount of transferred data with a byte level granularity the lower bits will not carry too much meaning. When using burst transfers the lower bits will remain the same for longer periods and then rapidly increment during the burst transfer. The upper bits though will increment with the actual frequency with which the data is consumed. And that's what applications are typically interested in. - Lars Lars-Peter Clausen (3): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check DMA residue granularity drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 17 ++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 37 +++++++++++++++++++++++++++++++++-- 3 files changed, 53 insertions(+), 2 deletions(-) -- 1.8.0

2 4

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Anssi Hannula 09 Dec '13

09 Dec '13

07.12.2013 09:18, Raymond Yau kirjoitti: > I just changed my RadeonHD 4670 for a 7870, and now when I type > "aplay -l" I get many HDMI codecs instead of just one like for the > previous > > > Do your card specification support Discrete Digital Multi-Point Audio > (DDMA) /?/ > > DDMA enables Radeon™HD 7700-7900 Series GPUs with the ability to output > multiple and independent audio streams simultaneously through digital > interfaces that support audio, such as DisplayPort and HDMI®. Each audio > stream can be multi-channel (up to 8 channels) AFAIK yes, the multi-hdmi-device AMD cards are "DDMA". [...] > monitor_present 1 > eld_valid 1 > monitor_name > connection_type HDMI > eld_version [0x2] CEA-861D or below > edid_version [0x0] no CEA EDID Timing Extension block present > manufacture_id 0x0 > product_id 0x0 > port_id 0x0 > support_hdcp 0 > support_ai 0 > audio_sync_delay 0 > speakers [0x5f] FL/FR LFE FC RL/RR RC RLC/RRC > sad_count 6 > sad0_coding_type [0x1] LPCM > sad0_channels 8 > sad0_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 > sad0_bits [0xe0000] 16 20 24 > sad1_coding_type [0x2] AC-3 > sad1_channels 6 > sad1_rates [0xe0] 32000 44100 48000 > sad1_max_bitrate 640000 > sad2_coding_type [0x7] DTS > sad2_channels 6 > sad2_rates [0x6c0] 44100 48000 88200 96000 > sad2_max_bitrate 1536000 > sad3_coding_type [0xa] E-AC-3/DD+ (Dolby Digital Plus) > sad3_channels 8 > sad3_rates [0xc0] 44100 48000 > sad4_coding_type [0xb] DTS-HD > sad4_channels 8 > sad4_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 > sad5_coding_type [0xc] MLP (Dolby TrueHD) > sad5_channels 8 > sad5_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > > e.g. HDMI status message in function hdmi_present_sense in patch_hdmi.c > > need to find out why eld->monitor_present was set for all six complexes > > the value return by function atihdmi_pin_get_eld since it change > eld->eld_valid to false when it is negative When CONFIG_SND_DEBUG_VERBOSE is enabled, hdmi_present_sense() prints the actual hw Presence_Detect/ELD_Valid before eld->eld_valid is overridden to 0 (and hdmi_intrinsic_event() prints the hot plug event values). Alex, do you have information regarding why "Presence detect" (aka monitor_present above) seems to be set for all HDMI pins in the response of Pin Sense 0xF09 verb (or if it is even expected), and/or if we should use alternate means of getting jack status? CONFIG_SND_DEBUG_VERBOSE log might also be interesting, I wonder if eld_valid is also 1 for every pin or not (before ALSA forces 0). Also, I wonder if all monitor_present fields are 1 with fglrx driver as well, or when all displays are unplugged... [...] -- Anssi Hannula

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable stereo mix as default for AD and VIA codecs
by Takashi Iwai 09 Dec '13

09 Dec '13

AD and VIA codecs had stereo mixer input enabled as default before moving to the generic parser, and people think the lack of such a regression. In this patch, the stereo mixer input is added back to the input selection if no auto-mic is available, and if it's not disabled explicitly via hint. This should satisfy most of demands, i.e. stereo mix on desktop machines like what it worked before, and it still keeps the new auto-mic feature on laptops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 19 ++++++++++++++++++- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_via.c | 1 + 3 files changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 058318de1af4..44be167d1cad 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3033,6 +3033,8 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin, spec->imux_pins[imux->num_items] = pin; snd_hda_add_imux_item(imux, label, cfg_idx, NULL); imux_added = true; + if (spec->dyn_adc_switch) + spec->dyn_adc_idx[imux_idx] = c; } } @@ -3130,7 +3132,9 @@ static int create_input_ctls(struct hda_codec *codec) } } - if (mixer && spec->add_stereo_mix_input) { + /* add stereo mix when explicitly enabled via hint */ + if (mixer && spec->add_stereo_mix_input && + snd_hda_get_bool_hint(codec, "add_stereo_mix_input") > 0) { err = parse_capture_source(codec, mixer, CFG_IDX_MIX, num_adcs, "Stereo Mix", 0); if (err < 0) @@ -4403,6 +4407,19 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* add stereo mix if available and not enabled yet */ + if (!spec->auto_mic && spec->mixer_nid && + spec->add_stereo_mix_input && + spec->input_mux.num_items > 1 && + snd_hda_get_bool_hint(codec, "add_stereo_mix_input") < 0) { + err = parse_capture_source(codec, spec->mixer_nid, + CFG_IDX_MIX, spec->num_all_adcs, + "Stereo Mix", 0); + if (err < 0) + return err; + } + + err = create_capture_mixers(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 34d86ec5d3dd..05280033c302 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -194,6 +194,7 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) codec->no_sticky_stream = 1; spec->gen.indep_hp = 1; + spec->gen.add_stereo_mix_input = 1; err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); if (err < 0) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0bc20ef5687a..f84195f3ea31 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -138,6 +138,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; + spec->gen.add_stereo_mix_input = 1; return spec; } -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Split verb definitions into sound/hda_verbs.h
by Takashi Iwai 09 Dec '13

09 Dec '13

Since there are more HD-audio compatible codecs, move the definitions of HD-audio verbs into common header location, include/sound, so that it can be included cleanly from other drivers than HD-audio driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_verbs.h | 554 ++++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 547 +-------------------------------------------- 2 files changed, 555 insertions(+), 546 deletions(-) create mode 100644 include/sound/hda_verbs.h diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h new file mode 100644 index 000000000000..d0509db6d0ec --- /dev/null +++ b/include/sound/hda_verbs.h @@ -0,0 +1,554 @@ +/* + * HD-audio codec verbs + */ + +#ifndef __SOUND_HDA_VERBS_H +#define __SOUND_HDA_VERBS_H + +/* + * nodes + */ +#define AC_NODE_ROOT 0x00 + +/* + * function group types + */ +enum { + AC_GRP_AUDIO_FUNCTION = 0x01, + AC_GRP_MODEM_FUNCTION = 0x02, +}; + +/* + * widget types + */ +enum { + AC_WID_AUD_OUT, /* Audio Out */ + AC_WID_AUD_IN, /* Audio In */ + AC_WID_AUD_MIX, /* Audio Mixer */ + AC_WID_AUD_SEL, /* Audio Selector */ + AC_WID_PIN, /* Pin Complex */ + AC_WID_POWER, /* Power */ + AC_WID_VOL_KNB, /* Volume Knob */ + AC_WID_BEEP, /* Beep Generator */ + AC_WID_VENDOR = 0x0f /* Vendor specific */ +}; + +/* + * GET verbs + */ +#define AC_VERB_GET_STREAM_FORMAT 0x0a00 +#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00 +#define AC_VERB_GET_PROC_COEF 0x0c00 +#define AC_VERB_GET_COEF_INDEX 0x0d00 +#define AC_VERB_PARAMETERS 0x0f00 +#define AC_VERB_GET_CONNECT_SEL 0x0f01 +#define AC_VERB_GET_CONNECT_LIST 0x0f02 +#define AC_VERB_GET_PROC_STATE 0x0f03 +#define AC_VERB_GET_SDI_SELECT 0x0f04 +#define AC_VERB_GET_POWER_STATE 0x0f05 +#define AC_VERB_GET_CONV 0x0f06 +#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07 +#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08 +#define AC_VERB_GET_PIN_SENSE 0x0f09 +#define AC_VERB_GET_BEEP_CONTROL 0x0f0a +#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f +/* f10-f1a: GPIO */ +#define AC_VERB_GET_GPIO_DATA 0x0f15 +#define AC_VERB_GET_GPIO_MASK 0x0f16 +#define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 +#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 +#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a +#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c +/* f20: AFG/MFG */ +#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 +#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d +#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e +#define AC_VERB_GET_HDMI_ELDD 0x0f2f +#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30 +#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31 +#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 +#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 +#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 + +/* + * SET verbs + */ +#define AC_VERB_SET_STREAM_FORMAT 0x200 +#define AC_VERB_SET_AMP_GAIN_MUTE 0x300 +#define AC_VERB_SET_PROC_COEF 0x400 +#define AC_VERB_SET_COEF_INDEX 0x500 +#define AC_VERB_SET_CONNECT_SEL 0x701 +#define AC_VERB_SET_PROC_STATE 0x703 +#define AC_VERB_SET_SDI_SELECT 0x704 +#define AC_VERB_SET_POWER_STATE 0x705 +#define AC_VERB_SET_CHANNEL_STREAMID 0x706 +#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707 +#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708 +#define AC_VERB_SET_PIN_SENSE 0x709 +#define AC_VERB_SET_BEEP_CONTROL 0x70a +#define AC_VERB_SET_EAPD_BTLENABLE 0x70c +#define AC_VERB_SET_DIGI_CONVERT_1 0x70d +#define AC_VERB_SET_DIGI_CONVERT_2 0x70e +#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f +#define AC_VERB_SET_GPIO_DATA 0x715 +#define AC_VERB_SET_GPIO_MASK 0x716 +#define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 +#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f +#define AC_VERB_SET_EAPD 0x788 +#define AC_VERB_SET_CODEC_RESET 0x7ff +#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d +#define AC_VERB_SET_HDMI_DIP_INDEX 0x730 +#define AC_VERB_SET_HDMI_DIP_DATA 0x731 +#define AC_VERB_SET_HDMI_DIP_XMIT 0x732 +#define AC_VERB_SET_HDMI_CP_CTRL 0x733 +#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 + +/* + * Parameter IDs + */ +#define AC_PAR_VENDOR_ID 0x00 +#define AC_PAR_SUBSYSTEM_ID 0x01 +#define AC_PAR_REV_ID 0x02 +#define AC_PAR_NODE_COUNT 0x04 +#define AC_PAR_FUNCTION_TYPE 0x05 +#define AC_PAR_AUDIO_FG_CAP 0x08 +#define AC_PAR_AUDIO_WIDGET_CAP 0x09 +#define AC_PAR_PCM 0x0a +#define AC_PAR_STREAM 0x0b +#define AC_PAR_PIN_CAP 0x0c +#define AC_PAR_AMP_IN_CAP 0x0d +#define AC_PAR_CONNLIST_LEN 0x0e +#define AC_PAR_POWER_STATE 0x0f +#define AC_PAR_PROC_CAP 0x10 +#define AC_PAR_GPIO_CAP 0x11 +#define AC_PAR_AMP_OUT_CAP 0x12 +#define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 +#define AC_PAR_HDMI_LPCM_CAP 0x20 + +/* + * AC_VERB_PARAMETERS results (32bit) + */ + +/* Function Group Type */ +#define AC_FGT_TYPE (0xff<<0) +#define AC_FGT_TYPE_SHIFT 0 +#define AC_FGT_UNSOL_CAP (1<<8) + +/* Audio Function Group Capabilities */ +#define AC_AFG_OUT_DELAY (0xf<<0) +#define AC_AFG_IN_DELAY (0xf<<8) +#define AC_AFG_BEEP_GEN (1<<16) + +/* Audio Widget Capabilities */ +#define AC_WCAP_STEREO (1<<0) /* stereo I/O */ +#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */ +#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */ +#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */ +#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */ +#define AC_WCAP_STRIPE (1<<5) /* stripe */ +#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */ +#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */ +#define AC_WCAP_CONN_LIST (1<<8) /* connection list */ +#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */ +#define AC_WCAP_POWER (1<<10) /* power control */ +#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */ +#define AC_WCAP_CP_CAPS (1<<12) /* content protection */ +#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */ +#define AC_WCAP_DELAY (0xf<<16) +#define AC_WCAP_DELAY_SHIFT 16 +#define AC_WCAP_TYPE (0xf<<20) +#define AC_WCAP_TYPE_SHIFT 20 + +/* supported PCM rates and bits */ +#define AC_SUPPCM_RATES (0xfff << 0) +#define AC_SUPPCM_BITS_8 (1<<16) +#define AC_SUPPCM_BITS_16 (1<<17) +#define AC_SUPPCM_BITS_20 (1<<18) +#define AC_SUPPCM_BITS_24 (1<<19) +#define AC_SUPPCM_BITS_32 (1<<20) + +/* supported PCM stream format */ +#define AC_SUPFMT_PCM (1<<0) +#define AC_SUPFMT_FLOAT32 (1<<1) +#define AC_SUPFMT_AC3 (1<<2) + +/* GP I/O count */ +#define AC_GPIO_IO_COUNT (0xff<<0) +#define AC_GPIO_O_COUNT (0xff<<8) +#define AC_GPIO_O_COUNT_SHIFT 8 +#define AC_GPIO_I_COUNT (0xff<<16) +#define AC_GPIO_I_COUNT_SHIFT 16 +#define AC_GPIO_UNSOLICITED (1<<30) +#define AC_GPIO_WAKE (1<<31) + +/* Converter stream, channel */ +#define AC_CONV_CHANNEL (0xf<<0) +#define AC_CONV_STREAM (0xf<<4) +#define AC_CONV_STREAM_SHIFT 4 + +/* Input converter SDI select */ +#define AC_SDI_SELECT (0xf<<0) + +/* stream format id */ +#define AC_FMT_CHAN_SHIFT 0 +#define AC_FMT_CHAN_MASK (0x0f << 0) +#define AC_FMT_BITS_SHIFT 4 +#define AC_FMT_BITS_MASK (7 << 4) +#define AC_FMT_BITS_8 (0 << 4) +#define AC_FMT_BITS_16 (1 << 4) +#define AC_FMT_BITS_20 (2 << 4) +#define AC_FMT_BITS_24 (3 << 4) +#define AC_FMT_BITS_32 (4 << 4) +#define AC_FMT_DIV_SHIFT 8 +#define AC_FMT_DIV_MASK (7 << 8) +#define AC_FMT_MULT_SHIFT 11 +#define AC_FMT_MULT_MASK (7 << 11) +#define AC_FMT_BASE_SHIFT 14 +#define AC_FMT_BASE_48K (0 << 14) +#define AC_FMT_BASE_44K (1 << 14) +#define AC_FMT_TYPE_SHIFT 15 +#define AC_FMT_TYPE_PCM (0 << 15) +#define AC_FMT_TYPE_NON_PCM (1 << 15) + +/* Unsolicited response control */ +#define AC_UNSOL_TAG (0x3f<<0) +#define AC_UNSOL_ENABLED (1<<7) +#define AC_USRSP_EN AC_UNSOL_ENABLED + +/* Unsolicited responses */ +#define AC_UNSOL_RES_TAG (0x3f<<26) +#define AC_UNSOL_RES_TAG_SHIFT 26 +#define AC_UNSOL_RES_SUBTAG (0x1f<<21) +#define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ +#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ +#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ +#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ +#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */ + +/* Pin widget capabilies */ +#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ +#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ +#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */ +#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */ +#define AC_PINCAP_OUT (1<<4) /* output capable */ +#define AC_PINCAP_IN (1<<5) /* input capable */ +#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, + * but is marked reserved in the Intel HDA specification. + */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ +/* Note: The same bit as LR_SWAP is newly defined as HDMI capability + * in HD-audio specification + */ +#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */ +#define AC_PINCAP_DP (1<<24) /* DisplayPort pin, can + * coexist with AC_PINCAP_HDMI + */ +#define AC_PINCAP_VREF (0x37<<8) +#define AC_PINCAP_VREF_SHIFT 8 +#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ +#define AC_PINCAP_HBR (1<<27) /* High Bit Rate */ +/* Vref status (used in pin cap) */ +#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */ +#define AC_PINCAP_VREF_50 (1<<1) /* 50% */ +#define AC_PINCAP_VREF_GRD (1<<2) /* ground */ +#define AC_PINCAP_VREF_80 (1<<4) /* 80% */ +#define AC_PINCAP_VREF_100 (1<<5) /* 100% */ + +/* Amplifier capabilities */ +#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */ +#define AC_AMPCAP_OFFSET_SHIFT 0 +#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */ +#define AC_AMPCAP_NUM_STEPS_SHIFT 8 +#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB + * in 0.25dB + */ +#define AC_AMPCAP_STEP_SIZE_SHIFT 16 +#define AC_AMPCAP_MUTE (1<<31) /* mute capable */ +#define AC_AMPCAP_MUTE_SHIFT 31 + +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + +/* Connection list */ +#define AC_CLIST_LENGTH (0x7f<<0) +#define AC_CLIST_LONG (1<<7) + +/* Supported power status */ +#define AC_PWRST_D0SUP (1<<0) +#define AC_PWRST_D1SUP (1<<1) +#define AC_PWRST_D2SUP (1<<2) +#define AC_PWRST_D3SUP (1<<3) +#define AC_PWRST_D3COLDSUP (1<<4) +#define AC_PWRST_S3D3COLDSUP (1<<29) +#define AC_PWRST_CLKSTOP (1<<30) +#define AC_PWRST_EPSS (1U<<31) + +/* Power state values */ +#define AC_PWRST_SETTING (0xf<<0) +#define AC_PWRST_ACTUAL (0xf<<4) +#define AC_PWRST_ACTUAL_SHIFT 4 +#define AC_PWRST_D0 0x00 +#define AC_PWRST_D1 0x01 +#define AC_PWRST_D2 0x02 +#define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) + +/* Processing capabilies */ +#define AC_PCAP_BENIGN (1<<0) +#define AC_PCAP_NUM_COEF (0xff<<8) +#define AC_PCAP_NUM_COEF_SHIFT 8 + +/* Volume knobs capabilities */ +#define AC_KNBCAP_NUM_STEPS (0x7f<<0) +#define AC_KNBCAP_DELTA (1<<7) + +/* HDMI LPCM capabilities */ +#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ +#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */ +#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */ +#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */ +#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ +#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */ +#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */ +#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */ +#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ +#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */ +#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */ +#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */ +#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ +#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ + +/* Display pin's device list length */ +#define AC_DEV_LIST_LEN_MASK 0x3f +#define AC_MAX_DEV_LIST_LEN 64 + +/* + * Control Parameters + */ + +/* Amp gain/mute */ +#define AC_AMP_MUTE (1<<7) +#define AC_AMP_GAIN (0x7f) +#define AC_AMP_GET_INDEX (0xf<<0) + +#define AC_AMP_GET_LEFT (1<<13) +#define AC_AMP_GET_RIGHT (0<<13) +#define AC_AMP_GET_OUTPUT (1<<15) +#define AC_AMP_GET_INPUT (0<<15) + +#define AC_AMP_SET_INDEX (0xf<<8) +#define AC_AMP_SET_INDEX_SHIFT 8 +#define AC_AMP_SET_RIGHT (1<<12) +#define AC_AMP_SET_LEFT (1<<13) +#define AC_AMP_SET_INPUT (1<<14) +#define AC_AMP_SET_OUTPUT (1<<15) + +/* DIGITAL1 bits */ +#define AC_DIG1_ENABLE (1<<0) +#define AC_DIG1_V (1<<1) +#define AC_DIG1_VCFG (1<<2) +#define AC_DIG1_EMPHASIS (1<<3) +#define AC_DIG1_COPYRIGHT (1<<4) +#define AC_DIG1_NONAUDIO (1<<5) +#define AC_DIG1_PROFESSIONAL (1<<6) +#define AC_DIG1_LEVEL (1<<7) + +/* DIGITAL2 bits */ +#define AC_DIG2_CC (0x7f<<0) + +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<0) +#define AC_DIG3_KAE (1<<7) + +/* Pin widget control - 8bit */ +#define AC_PINCTL_EPT (0x3<<0) +#define AC_PINCTL_EPT_NATIVE 0 +#define AC_PINCTL_EPT_HBR 3 +#define AC_PINCTL_VREFEN (0x7<<0) +#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ +#define AC_PINCTL_VREF_50 1 /* 50% */ +#define AC_PINCTL_VREF_GRD 2 /* ground */ +#define AC_PINCTL_VREF_80 4 /* 80% */ +#define AC_PINCTL_VREF_100 5 /* 100% */ +#define AC_PINCTL_IN_EN (1<<5) +#define AC_PINCTL_OUT_EN (1<<6) +#define AC_PINCTL_HP_EN (1<<7) + +/* Pin sense - 32bit */ +#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) +#define AC_PINSENSE_PRESENCE (1<<31) +#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */ + +/* EAPD/BTL enable - 32bit */ +#define AC_EAPDBTL_BALANCED (1<<0) +#define AC_EAPDBTL_EAPD (1<<1) +#define AC_EAPDBTL_LR_SWAP (1<<2) + +/* HDMI ELD data */ +#define AC_ELDD_ELD_VALID (1<<31) +#define AC_ELDD_ELD_DATA 0xff + +/* HDMI DIP size */ +#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */ +#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */ + +/* HDMI DIP index */ +#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */ +#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */ + +/* HDMI DIP xmit (transmit) control */ +#define AC_DIPXMIT_MASK (0x3<<6) +#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */ +#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */ +#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */ + +/* HDMI content protection (CP) control */ +#define AC_CPCTRL_CES (1<<9) /* current encryption state */ +#define AC_CPCTRL_READY (1<<8) /* ready bit */ +#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */ +#define AC_CPCTRL_STATE (3<<0) /* current CP request state */ + +/* Converter channel <-> HDMI slot mapping */ +#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */ +#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */ + +/* configuration default - 32bit */ +#define AC_DEFCFG_SEQUENCE (0xf<<0) +#define AC_DEFCFG_DEF_ASSOC (0xf<<4) +#define AC_DEFCFG_ASSOC_SHIFT 4 +#define AC_DEFCFG_MISC (0xf<<8) +#define AC_DEFCFG_MISC_SHIFT 8 +#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) +#define AC_DEFCFG_COLOR (0xf<<12) +#define AC_DEFCFG_COLOR_SHIFT 12 +#define AC_DEFCFG_CONN_TYPE (0xf<<16) +#define AC_DEFCFG_CONN_TYPE_SHIFT 16 +#define AC_DEFCFG_DEVICE (0xf<<20) +#define AC_DEFCFG_DEVICE_SHIFT 20 +#define AC_DEFCFG_LOCATION (0x3f<<24) +#define AC_DEFCFG_LOCATION_SHIFT 24 +#define AC_DEFCFG_PORT_CONN (0x3<<30) +#define AC_DEFCFG_PORT_CONN_SHIFT 30 + +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + +/* device device types (0x0-0xf) */ +enum { + AC_JACK_LINE_OUT, + AC_JACK_SPEAKER, + AC_JACK_HP_OUT, + AC_JACK_CD, + AC_JACK_SPDIF_OUT, + AC_JACK_DIG_OTHER_OUT, + AC_JACK_MODEM_LINE_SIDE, + AC_JACK_MODEM_HAND_SIDE, + AC_JACK_LINE_IN, + AC_JACK_AUX, + AC_JACK_MIC_IN, + AC_JACK_TELEPHONY, + AC_JACK_SPDIF_IN, + AC_JACK_DIG_OTHER_IN, + AC_JACK_OTHER = 0xf, +}; + +/* jack connection types (0x0-0xf) */ +enum { + AC_JACK_CONN_UNKNOWN, + AC_JACK_CONN_1_8, + AC_JACK_CONN_1_4, + AC_JACK_CONN_ATAPI, + AC_JACK_CONN_RCA, + AC_JACK_CONN_OPTICAL, + AC_JACK_CONN_OTHER_DIGITAL, + AC_JACK_CONN_OTHER_ANALOG, + AC_JACK_CONN_DIN, + AC_JACK_CONN_XLR, + AC_JACK_CONN_RJ11, + AC_JACK_CONN_COMB, + AC_JACK_CONN_OTHER = 0xf, +}; + +/* jack colors (0x0-0xf) */ +enum { + AC_JACK_COLOR_UNKNOWN, + AC_JACK_COLOR_BLACK, + AC_JACK_COLOR_GREY, + AC_JACK_COLOR_BLUE, + AC_JACK_COLOR_GREEN, + AC_JACK_COLOR_RED, + AC_JACK_COLOR_ORANGE, + AC_JACK_COLOR_YELLOW, + AC_JACK_COLOR_PURPLE, + AC_JACK_COLOR_PINK, + AC_JACK_COLOR_WHITE = 0xe, + AC_JACK_COLOR_OTHER, +}; + +/* Jack location (0x0-0x3f) */ +/* common case */ +enum { + AC_JACK_LOC_NONE, + AC_JACK_LOC_REAR, + AC_JACK_LOC_FRONT, + AC_JACK_LOC_LEFT, + AC_JACK_LOC_RIGHT, + AC_JACK_LOC_TOP, + AC_JACK_LOC_BOTTOM, +}; +/* bits 4-5 */ +enum { + AC_JACK_LOC_EXTERNAL = 0x00, + AC_JACK_LOC_INTERNAL = 0x10, + AC_JACK_LOC_SEPARATE = 0x20, + AC_JACK_LOC_OTHER = 0x30, +}; +enum { + /* external on primary chasis */ + AC_JACK_LOC_REAR_PANEL = 0x07, + AC_JACK_LOC_DRIVE_BAY, + /* internal */ + AC_JACK_LOC_RISER = 0x17, + AC_JACK_LOC_HDMI, + AC_JACK_LOC_ATAPI, + /* others */ + AC_JACK_LOC_MOBILE_IN = 0x37, + AC_JACK_LOC_MOBILE_OUT, +}; + +/* Port connectivity (0-3) */ +enum { + AC_JACK_PORT_COMPLEX, + AC_JACK_PORT_NONE, + AC_JACK_PORT_FIXED, + AC_JACK_PORT_BOTH, +}; + +/* max. codec address */ +#define HDA_MAX_CODEC_ADDRESS 0x0f + +#endif /* __SOUND_HDA_VERBS_H */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 260b190adf74..3c484c27048b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -25,552 +25,7 @@ #include <sound/control.h> #include <sound/pcm.h> #include <sound/hwdep.h> - -/* - * nodes - */ -#define AC_NODE_ROOT 0x00 - -/* - * function group types - */ -enum { - AC_GRP_AUDIO_FUNCTION = 0x01, - AC_GRP_MODEM_FUNCTION = 0x02, -}; - -/* - * widget types - */ -enum { - AC_WID_AUD_OUT, /* Audio Out */ - AC_WID_AUD_IN, /* Audio In */ - AC_WID_AUD_MIX, /* Audio Mixer */ - AC_WID_AUD_SEL, /* Audio Selector */ - AC_WID_PIN, /* Pin Complex */ - AC_WID_POWER, /* Power */ - AC_WID_VOL_KNB, /* Volume Knob */ - AC_WID_BEEP, /* Beep Generator */ - AC_WID_VENDOR = 0x0f /* Vendor specific */ -}; - -/* - * GET verbs - */ -#define AC_VERB_GET_STREAM_FORMAT 0x0a00 -#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00 -#define AC_VERB_GET_PROC_COEF 0x0c00 -#define AC_VERB_GET_COEF_INDEX 0x0d00 -#define AC_VERB_PARAMETERS 0x0f00 -#define AC_VERB_GET_CONNECT_SEL 0x0f01 -#define AC_VERB_GET_CONNECT_LIST 0x0f02 -#define AC_VERB_GET_PROC_STATE 0x0f03 -#define AC_VERB_GET_SDI_SELECT 0x0f04 -#define AC_VERB_GET_POWER_STATE 0x0f05 -#define AC_VERB_GET_CONV 0x0f06 -#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07 -#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08 -#define AC_VERB_GET_PIN_SENSE 0x0f09 -#define AC_VERB_GET_BEEP_CONTROL 0x0f0a -#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d -#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ -#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f -/* f10-f1a: GPIO */ -#define AC_VERB_GET_GPIO_DATA 0x0f15 -#define AC_VERB_GET_GPIO_MASK 0x0f16 -#define AC_VERB_GET_GPIO_DIRECTION 0x0f17 -#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 -#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 -#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a -#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c -/* f20: AFG/MFG */ -#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 -#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d -#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e -#define AC_VERB_GET_HDMI_ELDD 0x0f2f -#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30 -#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31 -#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 -#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 -#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 -#define AC_VERB_GET_DEVICE_SEL 0xf35 -#define AC_VERB_GET_DEVICE_LIST 0xf36 - -/* - * SET verbs - */ -#define AC_VERB_SET_STREAM_FORMAT 0x200 -#define AC_VERB_SET_AMP_GAIN_MUTE 0x300 -#define AC_VERB_SET_PROC_COEF 0x400 -#define AC_VERB_SET_COEF_INDEX 0x500 -#define AC_VERB_SET_CONNECT_SEL 0x701 -#define AC_VERB_SET_PROC_STATE 0x703 -#define AC_VERB_SET_SDI_SELECT 0x704 -#define AC_VERB_SET_POWER_STATE 0x705 -#define AC_VERB_SET_CHANNEL_STREAMID 0x706 -#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707 -#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708 -#define AC_VERB_SET_PIN_SENSE 0x709 -#define AC_VERB_SET_BEEP_CONTROL 0x70a -#define AC_VERB_SET_EAPD_BTLENABLE 0x70c -#define AC_VERB_SET_DIGI_CONVERT_1 0x70d -#define AC_VERB_SET_DIGI_CONVERT_2 0x70e -#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f -#define AC_VERB_SET_GPIO_DATA 0x715 -#define AC_VERB_SET_GPIO_MASK 0x716 -#define AC_VERB_SET_GPIO_DIRECTION 0x717 -#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 -#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 -#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f -#define AC_VERB_SET_EAPD 0x788 -#define AC_VERB_SET_CODEC_RESET 0x7ff -#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d -#define AC_VERB_SET_HDMI_DIP_INDEX 0x730 -#define AC_VERB_SET_HDMI_DIP_DATA 0x731 -#define AC_VERB_SET_HDMI_DIP_XMIT 0x732 -#define AC_VERB_SET_HDMI_CP_CTRL 0x733 -#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 -#define AC_VERB_SET_DEVICE_SEL 0x735 - -/* - * Parameter IDs - */ -#define AC_PAR_VENDOR_ID 0x00 -#define AC_PAR_SUBSYSTEM_ID 0x01 -#define AC_PAR_REV_ID 0x02 -#define AC_PAR_NODE_COUNT 0x04 -#define AC_PAR_FUNCTION_TYPE 0x05 -#define AC_PAR_AUDIO_FG_CAP 0x08 -#define AC_PAR_AUDIO_WIDGET_CAP 0x09 -#define AC_PAR_PCM 0x0a -#define AC_PAR_STREAM 0x0b -#define AC_PAR_PIN_CAP 0x0c -#define AC_PAR_AMP_IN_CAP 0x0d -#define AC_PAR_CONNLIST_LEN 0x0e -#define AC_PAR_POWER_STATE 0x0f -#define AC_PAR_PROC_CAP 0x10 -#define AC_PAR_GPIO_CAP 0x11 -#define AC_PAR_AMP_OUT_CAP 0x12 -#define AC_PAR_VOL_KNB_CAP 0x13 -#define AC_PAR_DEVLIST_LEN 0x15 -#define AC_PAR_HDMI_LPCM_CAP 0x20 - -/* - * AC_VERB_PARAMETERS results (32bit) - */ - -/* Function Group Type */ -#define AC_FGT_TYPE (0xff<<0) -#define AC_FGT_TYPE_SHIFT 0 -#define AC_FGT_UNSOL_CAP (1<<8) - -/* Audio Function Group Capabilities */ -#define AC_AFG_OUT_DELAY (0xf<<0) -#define AC_AFG_IN_DELAY (0xf<<8) -#define AC_AFG_BEEP_GEN (1<<16) - -/* Audio Widget Capabilities */ -#define AC_WCAP_STEREO (1<<0) /* stereo I/O */ -#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */ -#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */ -#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */ -#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */ -#define AC_WCAP_STRIPE (1<<5) /* stripe */ -#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */ -#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */ -#define AC_WCAP_CONN_LIST (1<<8) /* connection list */ -#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */ -#define AC_WCAP_POWER (1<<10) /* power control */ -#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */ -#define AC_WCAP_CP_CAPS (1<<12) /* content protection */ -#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */ -#define AC_WCAP_DELAY (0xf<<16) -#define AC_WCAP_DELAY_SHIFT 16 -#define AC_WCAP_TYPE (0xf<<20) -#define AC_WCAP_TYPE_SHIFT 20 - -/* supported PCM rates and bits */ -#define AC_SUPPCM_RATES (0xfff << 0) -#define AC_SUPPCM_BITS_8 (1<<16) -#define AC_SUPPCM_BITS_16 (1<<17) -#define AC_SUPPCM_BITS_20 (1<<18) -#define AC_SUPPCM_BITS_24 (1<<19) -#define AC_SUPPCM_BITS_32 (1<<20) - -/* supported PCM stream format */ -#define AC_SUPFMT_PCM (1<<0) -#define AC_SUPFMT_FLOAT32 (1<<1) -#define AC_SUPFMT_AC3 (1<<2) - -/* GP I/O count */ -#define AC_GPIO_IO_COUNT (0xff<<0) -#define AC_GPIO_O_COUNT (0xff<<8) -#define AC_GPIO_O_COUNT_SHIFT 8 -#define AC_GPIO_I_COUNT (0xff<<16) -#define AC_GPIO_I_COUNT_SHIFT 16 -#define AC_GPIO_UNSOLICITED (1<<30) -#define AC_GPIO_WAKE (1<<31) - -/* Converter stream, channel */ -#define AC_CONV_CHANNEL (0xf<<0) -#define AC_CONV_STREAM (0xf<<4) -#define AC_CONV_STREAM_SHIFT 4 - -/* Input converter SDI select */ -#define AC_SDI_SELECT (0xf<<0) - -/* stream format id */ -#define AC_FMT_CHAN_SHIFT 0 -#define AC_FMT_CHAN_MASK (0x0f << 0) -#define AC_FMT_BITS_SHIFT 4 -#define AC_FMT_BITS_MASK (7 << 4) -#define AC_FMT_BITS_8 (0 << 4) -#define AC_FMT_BITS_16 (1 << 4) -#define AC_FMT_BITS_20 (2 << 4) -#define AC_FMT_BITS_24 (3 << 4) -#define AC_FMT_BITS_32 (4 << 4) -#define AC_FMT_DIV_SHIFT 8 -#define AC_FMT_DIV_MASK (7 << 8) -#define AC_FMT_MULT_SHIFT 11 -#define AC_FMT_MULT_MASK (7 << 11) -#define AC_FMT_BASE_SHIFT 14 -#define AC_FMT_BASE_48K (0 << 14) -#define AC_FMT_BASE_44K (1 << 14) -#define AC_FMT_TYPE_SHIFT 15 -#define AC_FMT_TYPE_PCM (0 << 15) -#define AC_FMT_TYPE_NON_PCM (1 << 15) - -/* Unsolicited response control */ -#define AC_UNSOL_TAG (0x3f<<0) -#define AC_UNSOL_ENABLED (1<<7) -#define AC_USRSP_EN AC_UNSOL_ENABLED - -/* Unsolicited responses */ -#define AC_UNSOL_RES_TAG (0x3f<<26) -#define AC_UNSOL_RES_TAG_SHIFT 26 -#define AC_UNSOL_RES_SUBTAG (0x1f<<21) -#define AC_UNSOL_RES_SUBTAG_SHIFT 21 -#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry - * (for DP1.2 MST) - */ -#define AC_UNSOL_RES_DE_SHIFT 15 -#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ -#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ -#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ -#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ -#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */ - -/* Pin widget capabilies */ -#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ -#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ -#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */ -#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */ -#define AC_PINCAP_OUT (1<<4) /* output capable */ -#define AC_PINCAP_IN (1<<5) /* input capable */ -#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ -/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, - * but is marked reserved in the Intel HDA specification. - */ -#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ -/* Note: The same bit as LR_SWAP is newly defined as HDMI capability - * in HD-audio specification - */ -#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */ -#define AC_PINCAP_DP (1<<24) /* DisplayPort pin, can - * coexist with AC_PINCAP_HDMI - */ -#define AC_PINCAP_VREF (0x37<<8) -#define AC_PINCAP_VREF_SHIFT 8 -#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ -#define AC_PINCAP_HBR (1<<27) /* High Bit Rate */ -/* Vref status (used in pin cap) */ -#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */ -#define AC_PINCAP_VREF_50 (1<<1) /* 50% */ -#define AC_PINCAP_VREF_GRD (1<<2) /* ground */ -#define AC_PINCAP_VREF_80 (1<<4) /* 80% */ -#define AC_PINCAP_VREF_100 (1<<5) /* 100% */ - -/* Amplifier capabilities */ -#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */ -#define AC_AMPCAP_OFFSET_SHIFT 0 -#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */ -#define AC_AMPCAP_NUM_STEPS_SHIFT 8 -#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB - * in 0.25dB - */ -#define AC_AMPCAP_STEP_SIZE_SHIFT 16 -#define AC_AMPCAP_MUTE (1<<31) /* mute capable */ -#define AC_AMPCAP_MUTE_SHIFT 31 - -/* driver-specific amp-caps: using bits 24-30 */ -#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ - -/* Connection list */ -#define AC_CLIST_LENGTH (0x7f<<0) -#define AC_CLIST_LONG (1<<7) - -/* Supported power status */ -#define AC_PWRST_D0SUP (1<<0) -#define AC_PWRST_D1SUP (1<<1) -#define AC_PWRST_D2SUP (1<<2) -#define AC_PWRST_D3SUP (1<<3) -#define AC_PWRST_D3COLDSUP (1<<4) -#define AC_PWRST_S3D3COLDSUP (1<<29) -#define AC_PWRST_CLKSTOP (1<<30) -#define AC_PWRST_EPSS (1U<<31) - -/* Power state values */ -#define AC_PWRST_SETTING (0xf<<0) -#define AC_PWRST_ACTUAL (0xf<<4) -#define AC_PWRST_ACTUAL_SHIFT 4 -#define AC_PWRST_D0 0x00 -#define AC_PWRST_D1 0x01 -#define AC_PWRST_D2 0x02 -#define AC_PWRST_D3 0x03 -#define AC_PWRST_ERROR (1<<8) -#define AC_PWRST_CLK_STOP_OK (1<<9) -#define AC_PWRST_SETTING_RESET (1<<10) - -/* Processing capabilies */ -#define AC_PCAP_BENIGN (1<<0) -#define AC_PCAP_NUM_COEF (0xff<<8) -#define AC_PCAP_NUM_COEF_SHIFT 8 - -/* Volume knobs capabilities */ -#define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<7) - -/* HDMI LPCM capabilities */ -#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ -#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */ -#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */ -#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */ -#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ -#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */ -#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */ -#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */ -#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ -#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */ -#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */ -#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */ -#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ -#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ - -/* Display pin's device list length */ -#define AC_DEV_LIST_LEN_MASK 0x3f -#define AC_MAX_DEV_LIST_LEN 64 - -/* - * Control Parameters - */ - -/* Amp gain/mute */ -#define AC_AMP_MUTE (1<<7) -#define AC_AMP_GAIN (0x7f) -#define AC_AMP_GET_INDEX (0xf<<0) - -#define AC_AMP_GET_LEFT (1<<13) -#define AC_AMP_GET_RIGHT (0<<13) -#define AC_AMP_GET_OUTPUT (1<<15) -#define AC_AMP_GET_INPUT (0<<15) - -#define AC_AMP_SET_INDEX (0xf<<8) -#define AC_AMP_SET_INDEX_SHIFT 8 -#define AC_AMP_SET_RIGHT (1<<12) -#define AC_AMP_SET_LEFT (1<<13) -#define AC_AMP_SET_INPUT (1<<14) -#define AC_AMP_SET_OUTPUT (1<<15) - -/* DIGITAL1 bits */ -#define AC_DIG1_ENABLE (1<<0) -#define AC_DIG1_V (1<<1) -#define AC_DIG1_VCFG (1<<2) -#define AC_DIG1_EMPHASIS (1<<3) -#define AC_DIG1_COPYRIGHT (1<<4) -#define AC_DIG1_NONAUDIO (1<<5) -#define AC_DIG1_PROFESSIONAL (1<<6) -#define AC_DIG1_LEVEL (1<<7) - -/* DIGITAL2 bits */ -#define AC_DIG2_CC (0x7f<<0) - -/* DIGITAL3 bits */ -#define AC_DIG3_ICT (0xf<<0) -#define AC_DIG3_KAE (1<<7) - -/* Pin widget control - 8bit */ -#define AC_PINCTL_EPT (0x3<<0) -#define AC_PINCTL_EPT_NATIVE 0 -#define AC_PINCTL_EPT_HBR 3 -#define AC_PINCTL_VREFEN (0x7<<0) -#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ -#define AC_PINCTL_VREF_50 1 /* 50% */ -#define AC_PINCTL_VREF_GRD 2 /* ground */ -#define AC_PINCTL_VREF_80 4 /* 80% */ -#define AC_PINCTL_VREF_100 5 /* 100% */ -#define AC_PINCTL_IN_EN (1<<5) -#define AC_PINCTL_OUT_EN (1<<6) -#define AC_PINCTL_HP_EN (1<<7) - -/* Pin sense - 32bit */ -#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) -#define AC_PINSENSE_PRESENCE (1<<31) -#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */ - -/* EAPD/BTL enable - 32bit */ -#define AC_EAPDBTL_BALANCED (1<<0) -#define AC_EAPDBTL_EAPD (1<<1) -#define AC_EAPDBTL_LR_SWAP (1<<2) - -/* HDMI ELD data */ -#define AC_ELDD_ELD_VALID (1<<31) -#define AC_ELDD_ELD_DATA 0xff - -/* HDMI DIP size */ -#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */ -#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */ - -/* HDMI DIP index */ -#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */ -#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */ - -/* HDMI DIP xmit (transmit) control */ -#define AC_DIPXMIT_MASK (0x3<<6) -#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */ -#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */ -#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */ - -/* HDMI content protection (CP) control */ -#define AC_CPCTRL_CES (1<<9) /* current encryption state */ -#define AC_CPCTRL_READY (1<<8) /* ready bit */ -#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */ -#define AC_CPCTRL_STATE (3<<0) /* current CP request state */ - -/* Converter channel <-> HDMI slot mapping */ -#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */ -#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */ - -/* configuration default - 32bit */ -#define AC_DEFCFG_SEQUENCE (0xf<<0) -#define AC_DEFCFG_DEF_ASSOC (0xf<<4) -#define AC_DEFCFG_ASSOC_SHIFT 4 -#define AC_DEFCFG_MISC (0xf<<8) -#define AC_DEFCFG_MISC_SHIFT 8 -#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) -#define AC_DEFCFG_COLOR (0xf<<12) -#define AC_DEFCFG_COLOR_SHIFT 12 -#define AC_DEFCFG_CONN_TYPE (0xf<<16) -#define AC_DEFCFG_CONN_TYPE_SHIFT 16 -#define AC_DEFCFG_DEVICE (0xf<<20) -#define AC_DEFCFG_DEVICE_SHIFT 20 -#define AC_DEFCFG_LOCATION (0x3f<<24) -#define AC_DEFCFG_LOCATION_SHIFT 24 -#define AC_DEFCFG_PORT_CONN (0x3<<30) -#define AC_DEFCFG_PORT_CONN_SHIFT 30 - -/* Display pin's device list entry */ -#define AC_DE_PD (1<<0) -#define AC_DE_ELDV (1<<1) -#define AC_DE_IA (1<<2) - -/* device device types (0x0-0xf) */ -enum { - AC_JACK_LINE_OUT, - AC_JACK_SPEAKER, - AC_JACK_HP_OUT, - AC_JACK_CD, - AC_JACK_SPDIF_OUT, - AC_JACK_DIG_OTHER_OUT, - AC_JACK_MODEM_LINE_SIDE, - AC_JACK_MODEM_HAND_SIDE, - AC_JACK_LINE_IN, - AC_JACK_AUX, - AC_JACK_MIC_IN, - AC_JACK_TELEPHONY, - AC_JACK_SPDIF_IN, - AC_JACK_DIG_OTHER_IN, - AC_JACK_OTHER = 0xf, -}; - -/* jack connection types (0x0-0xf) */ -enum { - AC_JACK_CONN_UNKNOWN, - AC_JACK_CONN_1_8, - AC_JACK_CONN_1_4, - AC_JACK_CONN_ATAPI, - AC_JACK_CONN_RCA, - AC_JACK_CONN_OPTICAL, - AC_JACK_CONN_OTHER_DIGITAL, - AC_JACK_CONN_OTHER_ANALOG, - AC_JACK_CONN_DIN, - AC_JACK_CONN_XLR, - AC_JACK_CONN_RJ11, - AC_JACK_CONN_COMB, - AC_JACK_CONN_OTHER = 0xf, -}; - -/* jack colors (0x0-0xf) */ -enum { - AC_JACK_COLOR_UNKNOWN, - AC_JACK_COLOR_BLACK, - AC_JACK_COLOR_GREY, - AC_JACK_COLOR_BLUE, - AC_JACK_COLOR_GREEN, - AC_JACK_COLOR_RED, - AC_JACK_COLOR_ORANGE, - AC_JACK_COLOR_YELLOW, - AC_JACK_COLOR_PURPLE, - AC_JACK_COLOR_PINK, - AC_JACK_COLOR_WHITE = 0xe, - AC_JACK_COLOR_OTHER, -}; - -/* Jack location (0x0-0x3f) */ -/* common case */ -enum { - AC_JACK_LOC_NONE, - AC_JACK_LOC_REAR, - AC_JACK_LOC_FRONT, - AC_JACK_LOC_LEFT, - AC_JACK_LOC_RIGHT, - AC_JACK_LOC_TOP, - AC_JACK_LOC_BOTTOM, -}; -/* bits 4-5 */ -enum { - AC_JACK_LOC_EXTERNAL = 0x00, - AC_JACK_LOC_INTERNAL = 0x10, - AC_JACK_LOC_SEPARATE = 0x20, - AC_JACK_LOC_OTHER = 0x30, -}; -enum { - /* external on primary chasis */ - AC_JACK_LOC_REAR_PANEL = 0x07, - AC_JACK_LOC_DRIVE_BAY, - /* internal */ - AC_JACK_LOC_RISER = 0x17, - AC_JACK_LOC_HDMI, - AC_JACK_LOC_ATAPI, - /* others */ - AC_JACK_LOC_MOBILE_IN = 0x37, - AC_JACK_LOC_MOBILE_OUT, -}; - -/* Port connectivity (0-3) */ -enum { - AC_JACK_PORT_COMPLEX, - AC_JACK_PORT_NONE, - AC_JACK_PORT_FIXED, - AC_JACK_PORT_BOTH, -}; - -/* max. codec address */ -#define HDA_MAX_CODEC_ADDRESS 0x0f +#include <sound/hda_verbs.h> /* * generic arrays -- 1.8.5

1 0

[alsa-devel] [PATCH 0/2] ARM: Rename ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY
by Laurent Pinchart 09 Dec '13

09 Dec '13

Hello, This patch series enables compilation of SH-Mobile drivers on both the non-multiplatform (ARCH_SHMOBILE) and the multiplatform (ARCH_SHMOBILE_MULTI) SH-Mobile architectures. To do so it renames ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY and creates a new ARCH_SHMOBILE symbol selected by both ARCH_SHMOBILE_MULTI and ARCH_SHMOBILE_LEGACY. The second patch adds a dependency on ARCH_SHMOBILE_LEGACY to the ASoC R-Car driver. It can be squashed with the first patch or applied separately to minimize the risk of conflicts if desired. Laurent Pinchart (2): ARM: Rename ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY ASoC: sh: rcar: Depend on ARCH_SHMOBILE_LEGACY arch/arm/Kconfig | 14 ++++++++------ arch/arm/Makefile | 1 - arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/dts/Makefile | 2 +- arch/arm/mach-shmobile/Kconfig | 8 ++++++-- drivers/Makefile | 2 +- sound/soc/sh/Kconfig | 1 + 7 files changed, 18 insertions(+), 12 deletions(-) -- Regards, Laurent Pinchart

4 7

[alsa-devel] issues with usb-audio and maudio quattro
by Patrick Shirkey 09 Dec '13

09 Dec '13

Hi, It seems that some changes were made to the usb-audio driver over the years that have resulted in partially broken behaviour with maudio quattro. We have run some tests on two different devices with different machines and found the following items: 1: There are no longer three pcm devices - this breaks previously working asoundrc files 2: pcm0 and pcm1 have different behaviours - this results in audio input working for pcm0 but white noise in pcm1 (probably due to endianess le vs be) 3: We can record 4 tracks with arecord. Tracks 1/2 have audio and track 3/4 have white noise 4: We can no longer run 4 playback tracks in JACK2 as it complains about the ports not matching (this might be a seperate issue with JACK2 and route/plug configs for multichannel devices but that would probably affect the delta 10/10, RME Hammerfall or other multichannel devices too.) jackd: ../common/JackGraphManager.cpp:44: void Jack::JackGraphManager::AssertPort(jack_port_id_t): Assertion `port_index < fPortMax' failed. So the main question is who made the decision to drop/consolidate the third pcm and break previously working asoundrc files? There must be a good reason in which case maybe we can get things working again relatively easily? Cheers -- Patrick Shirkey Boost Hardware Ltd

3 2

[alsa-devel] [PATCH][RESEND 0/8] Add and implement gen_pool_dma_alloc()
by Nicolin Chen 09 Dec '13

09 Dec '13

Previously, we don't have a specific gen_pool_alloc() for DMA usage; Instead, we need to use gen_pool_virt_to_phys() to convert the addr returned from gen_pool_alloc(). So each implementation of this has duplicated code. Thus add new helper function -- gen_pool_dma_alloc(). After gen_pool_dma_alloc() is introduced, we can replace the original combination of gen_pool_alloc()+gen_pool_virt_to_phys() with this new helper function. Thus this patch implement the helper function to all the current drivers which use gen_pool_virt_to_phys(). !!-------------------important------------------!! The later 7 patches need all related driver owner to test. We can here define a simple rule: 1, If one driver owner or maintainer doesn't like the mofication to his/her driver, just let me know. I would drop that patch. 2, If there's a bug and issue found after patch-testing, please reply the mail so that I can fix and refine the patch. 3, If one driver owner or maintainer is too busy and doesn't have bandwidth to test the patch, I would drop that patch from this series. We can reimplement it till there's someone test it. !!-------------------current progress-----------!! lib/genalloc: [Okay] ARM: davinci: [Untested] dma: mmp_tdma: [Untested] [media] coda: [Untested] uio: uio_pruss: [Untested] ALSA: memalloc: [Tested] by Nicolin Chen with i.MX6Q SabreSD ASoC: davinci: [Untested] ASoC: pxa: use [Untested] Nicolin Chen (8): lib/genalloc: add a helper function for DMA buffer allocation ARM: davinci: use gen_pool_dma_alloc() to sram.c dma: mmp_tdma: use gen_pool_dma_alloc() to allocate descriptor [media] coda: use gen_pool_dma_alloc() to allocate iram buffer uio: uio_pruss: use gen_pool_dma_alloc() to allocate sram memory ALSA: memalloc: use gen_pool_dma_alloc() to allocate iram buffer ASoC: davinci: use gen_pool_dma_alloc() in davinci-pcm.c ASoC: pxa: use gen_pool_dma_alloc() to allocate dma buffer arch/arm/mach-davinci/sram.c | 9 +-------- drivers/dma/mmp_tdma.c | 7 +------ drivers/media/platform/coda.c | 5 ++--- drivers/uio/uio_pruss.c | 6 ++---- include/linux/genalloc.h | 2 ++ lib/genalloc.c | 28 ++++++++++++++++++++++++++++ sound/core/memalloc.c | 6 +----- sound/soc/davinci/davinci-pcm.c | 3 +-- sound/soc/pxa/mmp-pcm.c | 3 +-- 9 files changed, 39 insertions(+), 30 deletions(-) -- 1.8.4

5 13

[alsa-devel] Resend: Incorrect sbits for snd-usb-audio
by Sander Jansen 08 Dec '13

08 Dec '13

Hi, I had send this out earlier back in September, but never got a response. I have a Focusrite Scarlett 2i4 USB Audio Interface. The audio format exposed by ALSA for this device is S32_LE. From the hardware specs and lsusb. it supposed to only have a 24 bit resolution (with the LSB bits set to 0): AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 When I query the sbits for this device, I expect it to return 24, but instead it returns 32. Looking through the usb audio driver code, it doesn't look like this information is actually exposed to userspace. The 'parse_audio_format_i_type' in usb/format.c merely returns the supported formats, but doesn't include any msbits info. (and it doesn't look like it can be easily changed to return the msbits info as well). Was this intentionally left out, or is this a missing feature in the driver? Thanks, Sander

2 4

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 08 Dec '13

08 Dec '13

> I just changed my RadeonHD 4670 for a 7870, and now when I type > "aplay -l" I get many HDMI codecs instead of just one like for the > previous you have to specify CONFIG_SND_DYNAMIC_MINORS=y when building the driver [ 4.637413] ALSA hda_codec.c:4506 Too many HDMI devices [ 4.637414] ALSA hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y [ 4.637644] ALSA control.c:352 control 0:0:0:HDMI/DP Jack:0 is already present [ 4.639220] hda_codec: cannot build controls for #0 (error -16) > can you post the system log if the graphic driver know hdmi, vga and two >> mini display ports are connected or not ? >> >> I cannot see anything in dmesg related, so attaching Xorg log. > >> - >> > 5.268590] [drm] Radeon Display Connectors [ 5.268592] [drm] Connector 0: [ 5.268593] [drm] DP-1 [ 5.268594] [drm] HPD4 [ 5.268595] [drm] DDC: 0x6530 0x6530 0x6534 0x6534 0x6538 0x6538 0x653c 0x653c [ 5.268596] [drm] Encoders: [ 5.268597] [drm] DFP1: INTERNAL_UNIPHY2 [ 5.268598] [drm] Connector 1: [ 5.268599] [drm] DP-2 [ 5.268600] [drm] HPD5 [ 5.268601] [drm] DDC: 0x6540 0x6540 0x6544 0x6544 0x6548 0x6548 0x654c 0x654c [ 5.268602] [drm] Encoders: [ 5.268603] [drm] DFP2: INTERNAL_UNIPHY2 [ 5.268603] [drm] Connector 2: [ 5.268604] [drm] HDMI-A-1 [ 5.268605] [drm] HPD1 [ 5.268606] [drm] DDC: 0x6550 0x6550 0x6554 0x6554 0x6558 0x6558 0x655c 0x655c [ 5.268607] [drm] Encoders: [ 5.268608] [drm] DFP3: INTERNAL_UNIPHY1 [ 5.268609] [drm] Connector 3: [ 5.268610] [drm] DVI-I-1 [ 5.268610] [drm] HPD6 [ 5.268612] [drm] DDC: 0x6580 0x6580 0x6584 0x6584 0x6588 0x6588 0x658c 0x658c [ 5.268612] [drm] Encoders: [ 5.268613] [drm] DFP4: INTERNAL_UNIPHY [ 5.268614] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 7.613] (II) RADEON(0): Output DisplayPort-0 has no monitor section [ 7.626] (II) RADEON(0): Output DisplayPort-1 has no monitor section [ 7.693] (II) RADEON(0): Output HDMI-0 has no monitor section [ 7.713] (II) RADEON(0): Output DVI-0 has no monitor section [ 7.726] (II) RADEON(0): EDID for output DisplayPort-0 [ 7.740] (II) RADEON(0): EDID for output DisplayPort-1 [ 7.801] (II) RADEON(0): EDID for output HDMI-0 [ 7.801] (II) RADEON(0): Manufacturer: DON Model: 26 Serial#: 16843009 [ 7.801] (II) RADEON(0): Year: 2011 Week: 0 [ 7.801] (II) RADEON(0): EDID Version: 1.3 [ 7.823] (II) RADEON(0): EDID for output DVI-0 [ 7.823] (II) RADEON(0): Output DisplayPort-0 disconnected [ 7.823] (II) RADEON(0): Output DisplayPort-1 disconnected [ 7.823] (II) RADEON(0): Output HDMI-0 connected [ 7.823] (II) RADEON(0): Output DVI-0 disconnected the graphic driver knew only HDMI-0 is connected and the others are not connected

2 1

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by John Ettedgui 08 Dec '13

08 Dec '13

and same logs but with -with-debug=detect instead of full.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Ignore small negative LPIB delay correction
by Takashi Iwai 06 Dec '13

06 Dec '13

Sometimes the hardware reports LPIB being advanced than POSBUF. When this happens, the driver adjusts to a positive value by adding the buffer size. Then the driver detects it as an error (greater than the period size), and stops the LPIB delay account from this point on. When I took a close look at these conditions, the values shown are all very small numbers, and it'd be better to just ignore these values instead of discontinuing the LPIB delay correction. In this patch, the driver checks a negative delay value and ignores if it's a significantly small error. Currently the threshold is set to 64 frames, but could be smaller. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: apply only when period size is greater than 64. sound/pci/hda/hda_intel.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14e8ecd..af86c71f27bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -431,6 +431,8 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + int delay_negative_threshold; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct mutex dsp_mutex; #endif @@ -2195,6 +2197,15 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + /* when LPIB delay correction gives a small negative value, + * we ignore it; currently set the threshold statically to + * 64 frames + */ + if (runtime->period_size > 64) + azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); + else + azx_dev->delay_negative_threshold = 0; + /* wallclk has 24Mhz clock source */ azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); @@ -2447,8 +2458,12 @@ static unsigned int azx_get_position(struct azx *chip, delay = pos - lpib_pos; else delay = lpib_pos - pos; - if (delay < 0) - delay += azx_dev->bufsize; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX "%s: Unstable LPIB (%d >= %d); " -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Ignore small negative LPIB delay correction
by Takashi Iwai 06 Dec '13

06 Dec '13

Sometimes the hardware reports LPIB being advanced than POSBUF. When this happens, the driver adjusts to a positive value by adding the buffer size. Then the driver detects it as an error (greater than the period size), and stops the LPIB delay account from this point on. When I took a close look at these conditions, the values shown are all very small numbers, and it'd be better to just ignore these values instead of discontinuing the LPIB delay correction. In this patch, the driver checks a negative delay value and ignores if it's a significantly small error. Currently the threshold is set to 64 frames, but could be smaller. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14e8ecd..fd4f2288a33f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -431,6 +431,8 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + int delay_negative_threshold; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct mutex dsp_mutex; #endif @@ -2195,6 +2197,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + /* when LPIB delay correction gives a small negative value, + * we ignore it; currently set the threshold statically to + * 64 frames + */ + azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); + /* wallclk has 24Mhz clock source */ azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); @@ -2447,8 +2455,12 @@ static unsigned int azx_get_position(struct azx *chip, delay = pos - lpib_pos; else delay = lpib_pos - pos; - if (delay < 0) - delay += azx_dev->bufsize; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX "%s: Unstable LPIB (%d >= %d); " -- 1.8.5

1 0

[alsa-devel] [PATCH 0/3] Enhance imx-wm8962 machine driver
by Nicolin Chen 06 Dec '13

06 Dec '13

Although there's no direct relationship between any of these three patches, I recommend to apply them in order so as to circumvent merging conflicts. Nicolin Chen (3): ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver ASoC: fsl: imx-wm8962: Grant hw_params/free() permission to control FLL ASoC: fsl: imx-wm8962: Add headphone and microphone jack plugin/out detection .../devicetree/bindings/sound/imx-audio-wm8962.txt | 8 + Documentation/devicetree/bindings/sound/wm8962.txt | 8 + sound/soc/fsl/imx-wm8962.c | 361 ++++++++++++++++++--- 3 files changed, 326 insertions(+), 51 deletions(-) -- 1.8.4

2 9

[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 06 Dec '13

06 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 7 +++++-- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 42 insertions(+), 45 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23a9893..4a88e36c82ec 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_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_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_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_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b494d4..ee23194f7ab8 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315a3eaf..fb09a1c5f75b 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..eab0050d4579 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256fc4b2c..6a5e4bf6ac96 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.5.1

1 1

[alsa-devel] [PATCH] ASoC mc13783 codec: correct names
by Lambrecht Jürgen 06 Dec '13

06 Dec '13

Correct names to reality and for consistency: - "Speaker Amp Switch"->"Speaker Amp" - "Earpiece Amp Switch"->"Speaker Amp Switch" - "Loudspeaker enable"->"Loudspeaker Amp Switch" - "..Capture.."->"..Playback" (Line in..., CODEC..., ...Mixin Switch) - move "PCM Capture Volume" to bottom with other Capture controls Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 08fb022..878e3e5 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,7 +534,7 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, &cdcout_ctl), - SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, @@ -590,7 +590,7 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "SP", "Speaker Amp", "Speaker Amp Source MUX"}, { "LSP", "Loudspeaker Amp", "Loudspeaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, @@ -609,26 +609,27 @@ static const struct soc_enum mc13783_enum_3d_mixer = mc13783_3d_mixer); static struct snd_kcontrol_new mc13783_control_list[] = { - SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), - SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), - SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Loudspeaker Amp Switch", MC13783_AUDIO_RX0, 5, 1, 0), SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), - SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), - SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), - SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + SOC_SINGLE("CODEC Playback Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Playback Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Playback Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Playback Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Playback Switch", MC13783_AUDIO_RX1, 0, 1, 0), - SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), - SOC_SINGLE("CODEC Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + SOC_SINGLE("Line in Playback Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Playback Switch", MC13783_AUDIO_RX1, 10, 1, 0), - SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), - SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), -- 1.7.10.4

2 2

Re: [alsa-devel] Intermittent cut out on the video
by Dolevo Jay 06 Dec '13

06 Dec '13

From: cmst(a)live.com To: clemens(a)ladisch.de Subject: RE: [alsa-devel] Intermittent cut out on the video Date: Mon, 2 Dec 2013 09:44:16 +0000 > Date: Thu, 28 Nov 2013 09:29:45 +0100 > From: clemens(a)ladisch.de > To: cmst(a)live.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Intermittent cut out on the video > > Dolevo Jay wrote: > > I have two sound cards on my system and I use the [multi plugin] to > > have audio from two jacks simultaneously. > > > > The audio coming from the jack on the pc is cutting out intermittently. > > It just cuts out for a fraction of a second and continues as normal > > and after some time it happens again. > > The two cards use different sample clocks, so it is not possible to feed > them reliably from the same stream. > > You would need software that measures that actual sample rate of both > devices and resamples its output appropriately. Try using PulseAudio. > > > Regards, > Clemens > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Hi, I have tried to apply pulseaudio but couldn't really manage it. Could you please tell me if I can fix it in the configuration file? I mean, for example, would it be solved if I resample the audio for the appropriate sound card in the asound.conf? What other option do I have other than pulseaudio? Thanks a lot.

2 5

[alsa-devel] [PATCH] ALSA: hda - Allow capture-only configuration
by Takashi Iwai 06 Dec '13

06 Dec '13

We have blindly assumed that all valid configurations should have either analog or digital playback, but there can be capture-only configurations. The parser shouldn't escape in such a case. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2729e881541d..5ba3106b9712 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4334,7 +4334,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, spec->no_analog = 1; goto dig_only; } - return 0; /* can't find valid BIOS pin config */ + if (!cfg->num_inputs && !cfg->dig_in_pin) + return 0; /* can't find valid BIOS pin config */ } if (!spec->no_primary_hp && -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - skip depop delay before D3 for Haswell and Valleyview2 display codec
by mengdong.lin＠intel.com 06 Dec '13

06 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch skips the default depop delay before D3 for Haswell (10 ms) and Valleyview2 (100 ms) display codec, to reduce codec suspend time. The analog part of display audio is implemented in the external display. Some displays have weak pop noise while others not when suspending, no matter there is the default delay or not. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2306645..e14359a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2022,6 +2022,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } + if (is_haswell(codec) || is_valleyview(codec)) { + codec->depop_delay = 0; + } + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove quirk for Dell Vostro 131
by David Henningsson 06 Dec '13

06 Dec '13

I've tested the old Dell Vostro 131 with the latest generic parser and it works just fine, and as a bonus we get better jack detection features in userspace. Therefore this quirk can be removed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1f2717f..3fbf288 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - do delayed suspend for HD-audio devices
by mengdong.lin＠intel.com 06 Dec '13

06 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> HD-Audio codecs tends to take a long time to supend and is time-consuming part in driver suspend. In this patch, codec suspending and the left operations are delayed to a worker thead, so that they can be done asynchronously in parallel with suspending of other devices. And azx_suspend_late() is defined as PM .suspend_late ops to wait for completion of the delayed suspending. Credit goes to Liam Girdwood, who suggested this workaround. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14..aed4cdf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -545,6 +545,11 @@ struct azx { struct work_struct probe_work; +#ifdef CONFIG_PM + struct workqueue_struct *pm_wq; /* workqueue to delay controller PM */ + struct work_struct suspend_work; /* task to delay controller suspend */ +#endif + /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; @@ -2908,20 +2913,11 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) /* * power management */ -static int azx_suspend(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct azx_pcm *p; - if (chip->disabled) - return 0; +static int azx_suspend_continue(struct azx *chip) +{ + struct pci_dev *pci = chip->pci; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - azx_clear_irq_pending(chip); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); @@ -2937,6 +2933,41 @@ static int azx_suspend(struct device *dev) pci_set_power_state(pci, PCI_D3hot); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); + + return 0; +} + +static void azx_suspend_work(struct work_struct *work) +{ + azx_suspend_continue(container_of(work, struct azx, suspend_work)); +} + + +static int azx_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + if (chip->disabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + + queue_work(chip->pm_wq, &chip->suspend_work); + return 0; +} + +static int azx_suspend_late(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + flush_workqueue(chip->pm_wq); + return 0; } @@ -3057,6 +3088,7 @@ static int azx_runtime_idle(struct device *dev) #ifdef CONFIG_PM static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + .suspend_late = azx_suspend_late, SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; @@ -3272,6 +3304,12 @@ static int azx_free(struct azx *chip) hda_display_power(false); hda_i915_exit(); } + +#ifdef CONFIG_PM + if (chip->pm_wq) + destroy_workqueue(chip->pm_wq); +#endif + kfree(chip); return 0; @@ -3515,6 +3553,9 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, }; struct azx *chip; int err; +#ifdef CONFIG_PM + char wqname[24]; +#endif *rchip = NULL; @@ -3581,6 +3622,18 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* continue probing in work context as may trigger request module */ INIT_WORK(&chip->probe_work, azx_probe_work); +#ifdef CONFIG_PM + sprintf(wqname, "hda-azx-pm-wq%d", card->number); + chip->pm_wq = create_workqueue("wqname"); + if (!chip->pm_wq) { + snd_printk(KERN_ERR "cannot create PM workqueue\n"); + azx_free(chip); + return -ENOMEM; + } + + INIT_WORK(&chip->suspend_work, azx_suspend_work); +#endif + *rchip = chip; return 0; -- 1.8.1.2

3 5

Re: [alsa-devel] [PATCH] usb-audio: fix warning
by Takashi Iwai 05 Dec '13

05 Dec '13

At Thu, 5 Dec 2013 14:32:43 -0500 (EST), Mikulas Patocka wrote: > > Fix the following warning when optimizing for size with gcc-4.6.4: > sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized] > > Signed-off-by: Mikulas Patocka <mpatocka(a)redhat.com> Thanks, applied. Takashi > > --- > sound/usb/mixer_quirks.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > Index: linux-3.12.3/sound/usb/mixer_quirks.c > =================================================================== > --- linux-3.12.3.orig/sound/usb/mixer_quirks.c 2013-12-05 20:21:18.000000000 +0100 > +++ linux-3.12.3/sound/usb/mixer_quirks.c 2013-12-05 20:21:23.000000000 +0100 > @@ -1520,7 +1520,7 @@ static int snd_microii_controls_create(s > return err; > } > > - return err; > + return 0; > } > > int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) >

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV
by Nicolin Chen 05 Dec '13

05 Dec '13

DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK, which would cause the calculation result from DSPCLK_DIV invalid since bit DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK while the driver won't calculate it again for the current instance. In this circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value. So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for calculation and then disables it afterward. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 07da601..97db3b4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); + /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK. + * So we here provisionally enable it and then disable it afterward + * if current bias_level hasn't reached SND_SOC_BIAS_ON. + */ + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); + + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, 0); + if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); return; -- 1.8.4

3 3

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Do FLL configuration in hw_params() and hw_free()
by Nicolin Chen 05 Dec '13

05 Dec '13

Previously, we couldn't use hw_params() and hw_free() to open and close FLL becuase there might be race between two simmultaneous substreams and cause FLL configuration being changed and accordingly mulfunction. So we adopted DAPM to control it since there won't be any race in DAPM. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level to reconfigure FLL. But after we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() and hw_free() to control FLL would allow us to support flexible use cases, 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k32bit.wav' for example. Thus this patch mainly moves FLL configuration from set_bias_level() to hw_params() and hw_free() so as to enchance the sound card's capability. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 147 +++++++++++++++++++++------------------------ 1 file changed, 70 insertions(+), 77 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 61e4885..df67a27 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -28,6 +28,9 @@ #define DAI_NAME_SIZE 32 +#define IMX_WM8962_DAI_FMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + struct imx_wm8962_data { struct snd_soc_dai_link dai; struct snd_soc_card card; @@ -49,92 +52,84 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; -static int sample_rate = 44100; -static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - sample_rate = params_rate(params); - sample_format = params_format(params); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + snd_pcm_format_t sample_format = params_format(params); + u32 pll_out, sample_rate = params_rate(params); + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, IMX_WM8962_DAI_FMT); + if (ret) { + dev_err(dev, "failed to set codec dai fmt: %d\n", ret); + return ret; + } + + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + int ret; + + /* Don't diable FLL if still having multiple substreams running */ + if (codec_dai->active != 1) + return 0; + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret < 0) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } return 0; } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, }; -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; - else - pll_out = sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - dapm->bias_level = level; - - return 0; -} - static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; @@ -248,8 +243,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + data->dai.dai_fmt = IMX_WM8962_DAI_FMT; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); @@ -264,7 +258,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { -- 1.8.4

2 6

[alsa-devel] [PATCH] ASoC: samsung: Ensure DMA data is initialised for secondary DAI
by Mark Brown 05 Dec '13

05 Dec '13

From: Mark Brown <broonie(a)linaro.org> While testing the upcoming dmaengine conversion for the driver Padma noticed that the DMA data for the secondard DAI is not being correctly initialised and provided a fix on top of the conversion. Also do this fix for mainline. Reported-by: Padmavathi Venna <padma.v(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- Compile tested only, I don't have hardware to test. sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..f9f2227421ea 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + snd_soc_dai_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { -- 1.8.5.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: add DT binding document for MAX98090 CODEC
by Stephen Warren 05 Dec '13

05 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This binding mainly serves to document the list of input and output pins that may be used in a sound card's audio routing table. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This series doesn't depend on anything; it could be applied anywhere in the ASoC tree. .../devicetree/bindings/sound/max98090.txt | 43 ++++++++++++++++++++++ 1 file changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98090.txt diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt new file mode 100644 index 000000000000..e4c8b36dcf89 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -0,0 +1,43 @@ +MAX98090 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98090". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Pins on the device (for linking into audio routes): + + * MIC1 + * MIC2 + * DMICL + * DMICR + * IN1 + * IN2 + * IN3 + * IN4 + * IN5 + * IN6 + * IN12 + * IN34 + * IN56 + * HPL + * HPR + * SPKL + * SPKR + * RCVL + * RCVR + * MICBIAS + +Example: + +audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; +}; -- 1.8.1.5

2 6

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 05 Dec '13

05 Dec '13

> >>>>>>>>> I just changed my RadeonHD 4670 for a 7870, and now when I type >>>>>>>>> "aplay -l" I get many HDMI codecs instead of just one like for the >>>>>>>>> previous card. >>>>>>>>> >>>>>>>> >>>>> How many hdmi , dvi or display port connectors do your card have ? >>>>> >>>>> 1 hdmi, 1 vga, 2 tiny ports that I cannot name. >>>> >>> are these mini display port ? http://en.wikipedia.org/wiki/Mini_DisplayPort > >>>>> Specification seem support up to 6 displays >>>>> >>>> I believe these are virtual HDMI ouputs, not physical ones and total >>>> for 6 in aplay -l and for 4 in aplay -L. >>>> >>> >>> are you using latest alsa-lib ? >>> >>> >>> http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=383912ef83cb2d583523… >>> >>> Good point, I'm on a 3.13-rc2 kernel but totally forgot about the lib >> part. >> I have the 1.0.27.2 lib, let me get the git version. >> > > Alright exactly the same behavior with -git. >

1 0

[alsa-devel] [PATCH] ASoC: samsung: Initialize the dma_data for secondary dai
by Padmavathi Venna 05 Dec '13

05 Dec '13

This patch initializes the dma_data for the secondary dai. Without this patch there is crash due to NULL pointer dereference during boot time on systems where secondary dai is initialized. Below is the crash log: [ 1.630000] smdk-audio-wm8894 sound.7: wm8994-aif1 <-> 3830000.i2s mapping ok [ 1.630000] Unable to handle kernel NULL pointer dereference at virtual address 00000010 [ 1.630000] pgd = c0004000 [ 1.630000] [00000010] *pgd=00000000 [ 1.630000] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.630000] Modules linked in: ... ... [<c0342d6c>] (dmaengine_pcm_new+0x284/0x2ac) from [<c0341ae8>] (soc_new_pcm+0x2fc/0x3f4) [<c0341ae8>] (soc_new_pcm+0x2fc/0x3f4) from [<c0338558>] (snd_soc_register_card+0x1898/0x18d8) [<c0338558>] (snd_soc_register_card+0x1898/0x18d8) from [<c034cc34>] (smdk_audio_probe+0xa8/0xe8) [<c034cc34>] (smdk_audio_probe+0xa8/0xe8) from [<c0252368>] (platform_drv_probe+0x18/0x1c) [<c0252368>] (platform_drv_probe+0x18/0x1c) from [<c02510a4>] (driver_probe_device+0xf4/0x204) [<c02510a4>] (driver_probe_device+0xf4/0x204) from [<c0251240>] (__driver_attach+0x8c/0x90) [<c0251240>] (__driver_attach+0x8c/0x90) from [<c024f70c>] (bus_for_each_dev+0x54/0x88) [<c024f70c>] (bus_for_each_dev+0x54/0x88) from [<c0250724>] (bus_add_driver+0xe0/0x2b8) [<c0250724>] (bus_add_driver+0xe0/0x2b8) from [<c0251868>] (driver_register+0x78/0xf4) [<c0251868>] (driver_register+0x78/0xf4) from [<c00086e8>] (do_one_initcall+0x34/0x150) [<c00086e8>] (do_one_initcall+0x34/0x150) from [<c0543c30>] (kernel_init_freeable+0x12c/0x1d0) [<c0543c30>] (kernel_init_freeable+0x12c/0x1d0) from [<c03ebf9c>] (kernel_init+0x8/0xe8) [<c03ebf9c>] (kernel_init+0x8/0xe8) from [<c000e978>] (ret_from_fork+0x14/0x3c) Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- This is made based on Mark Brown for-next branch of sound.git This patch is based on Mark Brown below patches for generic dma engine support for samsung audio IPs. ASoC: samsung: Use ASoC dmaengine code where possible ASoC: samsung: Provide helper for DMA init sound/soc/samsung/dmaengine.c | 3 ++- sound/soc/samsung/i2s.c | 5 ++++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index ad0a371..3f60479 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -67,7 +67,8 @@ EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); int samsung_asoc_dma_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + SND_DMAENGINE_PCM_FLAG_COMPAT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME); } EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 67d9fa9..0bcd7fb 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { -- 1.7.4.4

3 3

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 05 Dec '13

05 Dec '13

>>>> I just changed my RadeonHD 4670 for a 7870, and now when I type "aplay >>>> -l" I get many HDMI codecs instead of just one like for the previous card. >>>> >>> How many hdmi , dvi or display port connectors do your card have ? Specification seem support up to 6 displays >>> >>> Oh interesting, but I was just using the standard HDMI connector of the >> VGA card without connecting my card to the motherboard by more than PCI-E. >> Also, spdif would not get me 6 channels in LCPM, so I don't think that >> was it but I could be wrong. >> >>> >>> >>> Is there any reason for you to disable on-board hda controller and alc888 as you don't have any capture device anymore ? pactl list Do pulseaudio server select null sink or hdmi.monitor as the default source ? >>>> When I type "aplay -L" I get one more codec that I can see it in the >>>> pavucontrol profile (not sure which one as the names are irrelevant) >>>> >>>> >>>> if in default.pa I specify the alsa sink device=hw:0,3 directly, I do >>>> get sound, but then in pavucontrol I do not get the advanced sub-menu >>>> anymore which makes me think something is not working properly... >>>> >>>> For now I have settled on this hack instead: >>>> >>>> >>>> I changed the extra-hdmi.conf in >>>> /usr/share/pulseaudio/alsa-mixer/profile-sets file to : >>>> >>>> [Mapping hdmi-surround-extra0] >>>> description = Digital Surround 5.1 (HDMI 0) >>>> device-strings = hdmi:%f,0 >>>> paths-output = hdmi-output-0 >>>> channel-map = >>>> front-left,front-right,rear-left,rear-right,front-center,lfe >>>> priority = 1 >>>> direction = output >>>> >>>> [Mapping hdmi-surround-extra3] >>>> description = Digital Surround 5.1 (HDMI 3) >>>> device-strings = hdmi:%f,0 >>>> paths-output = hdmi-output-3 >>>> channel-map = >>>> front-left,front-right,rear-left,rear-right,front-center,lfe >>>> priority = 1 >>>> direction = output >>>> >>>> I only need one of these two, somehow they both work... >>>> >>>> Now everything works for me, but since it's not out of the box, I am >>>> guessing there might be something out there to fix. >>>> >>>> >>>> >>> >>> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >>> >>> Hmmm I am not sure what you meant by that. >> >> > Here is the alsa-info. > strange that all jacks return true control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value true comment { access read type BOOLEAN count 1 } } ... control.31 { iface CARD name 'HDMI/DP,pcm=11 Jack' value true comment { access read type BOOLEAN count 1 } } cat /proc/asound/card0/eld#*.0 does your hdmi montior name appear in one of the six eld [ 5.468413] [drm] Connector 2: [ 5.468414] [drm] HDMI-A-1 [ 5.468415] [drm] HPD1 Does the graphic driver show which hdmi monitor connect to which ports in system log ?

1 0

[alsa-devel] [PATCH] ALSA: hda - fix mic issues on Acer Aspire E-572
by Oleksij Rempel 05 Dec '13

05 Dec '13

This patch add quirk for Acer Aspire E-572: - fix external mic - limit mic boost for internal mic with maximal noise level of -24dB Signed-off-by: Oleksij Rempel <linux(a)rempel-privat.de> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59c36de..f3041a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3902,6 +3902,7 @@ enum { ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, @@ -4164,6 +4165,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK, + }, [ALC269_FIXUP_ACER_AC700] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4261,6 +4268,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.3.2

2 1

[alsa-devel] alsa/asoundlib.h file not found
by Appalayagari Sreedhar 05 Dec '13

05 Dec '13

Hi, I have integrated the codec driver and the kernel log shows the sound card is detected. now i want to test the codec. i tried the sample test code for capture and playback. i am getting a compilation errors due to the asoundlib.h. please find the error log for my application. main.c:2:23: error: asoundlib.h: No such file or directory main.c: In function 'main': main.c:7: error: 'snd_pcm_t' undeclared (first use in this function) main.c:7: error: (Each undeclared identifier is reported only once main.c:7: error: for each function it appears in.) main.c:7: error: 'pcm_handle' undeclared (first use in this function) main.c:10: error: 'snd_pcm_stream_t' undeclared (first use in this function) main.c:10: error: expected ';' before 'stream' main.c:15: error: 'snd_pcm_hw_params_t' undeclared (first use in this function) main.c:15: error: 'hwparams' undeclared (first use in this function) main.c:22: warning: 'return' with a value, in function returning void main.c:34: error: 'snd_pcm_uframes_t' undeclared (first use in this function) main.c:34: error: expected ';' before 'periodsize' main.c:46: error: 'SND_PCM_ACCESS_RW_INTERLEAVED' undeclared (first use in this function) main.c:48: warning: 'return' with a value, in function returning void main.c:52: error: 'SND_PCM_FORMAT_S16_LE' undeclared (first use in this function) main.c:54: warning: 'return' with a value, in function returning void main.c:62: warning: 'return' with a value, in function returning void main.c:65:19: warning: missing terminating " character main.c:65: error: missing terminating " character main.c:66: error: expected expression before '==' token main.c:66: error: stray '\' in program main.c:66:31: warning: missing terminating " character main.c:66: error: missing terminating " character main.c:67: error: expected ';' before '}' token main.c:72: warning: 'return' with a value, in function returning void main.c:78: warning: 'return' with a value, in function returning void main.c:86: error: 'periodsize' undeclared (first use in this function) main.c:88: warning: 'return' with a value, in function returning void main.c:99: warning: 'return' with a value, in function returning void main.c:112: warning: incompatible implicit declaration of built-in function 'malloc' main.c:115: error: 'num_frames' undeclared (first use in this function) main.c:5: warning: return type of 'main' is not 'int' please suggest what needs to be done to resolve this issue asap. Thank you, Sreedhar.

5 4

[alsa-devel] [PATCH] ASoC: imx-ssi: Check the return value from clk_prepare_enable()
by Fabio Estevam 05 Dec '13

05 Dec '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 6336757..df552fa 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -535,7 +535,9 @@ static int imx_ssi_probe(struct platform_device *pdev) ret); goto failed_clk; } - clk_prepare_enable(ssi->clk); + ret = clk_prepare_enable(ssi->clk); + if (ret) + goto failed_clk; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); ssi->base = devm_ioremap_resource(&pdev->dev, res); -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 0/8] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 04 Dec '13

04 Dec '13

Since the patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs got merged into Linux 3.12, it is a good time to migrate the platform to use this driver and drop the legacy DMA code. Due to changes scattered across different areas of kernel, patches are based on merged 4 branches: - for-next of Kukjin Kim's Samsung tree, - clk-next of Mike Turquette's clock tree, - for-next of Mark Brown's SPI tree, - for-next of Mark Brown's ASoC tree. To ease testing I have prepared a branch in my private tree for anyone willing to check the patches out: git://github.com/tom3q/linux.git s3c64xx-pl08x-patches On S3C64xx-based Mini6410 and SMDK6410 boards, with I2S audio playback and capture (including full duplex operation) and also SPI using spidev: Tested-by: Tomasz Figa <tomasz.figa(a)gmail.com> Changes since v1: - Dropped the now merged patches adding PL080s support amba-pl08x driver, - Rebased onto current for-next branches of respective trees. - Stopped hijacking pl330_filter() symbol to plug pl08_filter_id() as channel filtering callback on S3C64xx. Changes since RFC v2: - Added clkdev lookups to old clock driver. - Added patches removing the old DMA driver and any remaining code needed by it. - Fixed DMA support for SPI. - Added a word about PL080S to amba-pl08x.c file header. - Changed definition of LLI words from enums to macros. - Extended debugging messages to handle PL080S variant as well. - Little cleanup of LLI dumping code. - Added check for peripheral flow control, which is unsupported by PL080S to dma_set_runtime_config. - Corrected transfer size mask of PL080S. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. Tomasz Figa (8): spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled ASoC: Samsung: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 762 ------------------------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 ++---- arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++ arch/arm/plat-samsung/devs.c | 10 +- arch/arm/plat-samsung/dma-ops.c | 8 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/spi/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 12 files changed, 316 insertions(+), 875 deletions(-) delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.3.2

5 16

[alsa-devel] 3.13 regression on panda board
by Takashi Iwai 04 Dec '13

04 Dec '13

Hi Peter (or anyone who is responsible): could you update the bugzilla report below? https://bugzilla.kernel.org/show_bug.cgi?id=66451 thanks, Takashi

3 6

[alsa-devel] [PATCH 07/10] ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case
by Lee Jones 04 Dec '13

04 Dec '13

In this patch we do two things. Firstly, instead of open coding the store of DMA data in to the DAI for later use, we use the API provided. Secondly we create and store similar DMA data for the DT case, only this time we use 'struct snd_dmaengine_dai_dma_data' which is provided by the core for this very reason. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_dai.c | 42 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 40 insertions(+), 2 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index c6fb5cc..8d1b01f 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -17,12 +17,14 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/clk.h> +#include <linux/of.h> #include <linux/regulator/consumer.h> #include <linux/mfd/dbx500-prcmu.h> #include <linux/platform_data/asoc-ux500-msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> +#include <sound/dmaengine_pcm.h> #include "ux500_msp_i2s.h" #include "ux500_msp_dai.h" @@ -654,16 +656,52 @@ static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, return ret; } +static int ux500_msp_dai_of_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_dmaengine_dai_dma_data *playback_dma_data; + struct snd_dmaengine_dai_dma_data *capture_dma_data; + + playback_dma_data = devm_kzalloc(dai->dev, + sizeof(*playback_dma_data), + GFP_KERNEL); + if (!playback_dma_data) + return -ENOMEM; + + capture_dma_data = devm_kzalloc(dai->dev, + sizeof(*capture_dma_data), + GFP_KERNEL); + if (!capture_dma_data) + return -ENOMEM; + + playback_dma_data->addr = drvdata->msp->playback_dma_data.tx_rx_addr; + capture_dma_data->addr = drvdata->msp->capture_dma_data.tx_rx_addr; + + playback_dma_data->maxburst = 4; + capture_dma_data->maxburst = 4; + + snd_soc_dai_init_dma_data(dai, playback_dma_data, capture_dma_data); + + return 0; +} + static int ux500_msp_dai_probe(struct snd_soc_dai *dai) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct device_node *np = dai->dev->of_node; + int ret; - dai->playback_dma_data = &drvdata->msp->playback_dma_data; - dai->capture_dma_data = &drvdata->msp->capture_dma_data; + if (np) { + ret = ux500_msp_dai_of_probe(dai); + return ret; + } drvdata->msp->playback_dma_data.data_size = drvdata->slot_width; drvdata->msp->capture_dma_data.data_size = drvdata->slot_width; + snd_soc_dai_init_dma_data(dai, + &drvdata->msp->playback_dma_data, + &drvdata->msp->capture_dma_data); return 0; } -- 1.8.1.2

4 8

[alsa-devel] [PATCH 0/2] ASoC: sam9x5_wm8731: fix left right channel swap issue
by Bo Shen 04 Dec '13

04 Dec '13

This patch series fix sam9x5 wm8731 playback left right channel swap issue However, sam9x5 and wm8731 only can work in DSP A mode while can not work in I2S mode. Bo Shen (2): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode sound/soc/atmel/atmel_ssc_dai.c | 30 +++++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- 2 files changed, 30 insertions(+), 2 deletions(-) -- 1.7.9.5

4 8

[alsa-devel] ALSA - No sound on MBA2,1
by Bruno Prémont 04 Dec '13

04 Dec '13

Hi Takashi, Since I started using Linux on a MBA2,1 I never had sound working. recently searching for some solution to get sound working I saw that some users had sound with 3.2.x kernels and reported it stopped working somewhere between 3.2 and 3.4. Thus I tried bisecting this and found two steps of losing sound support, a first one being losing sound output on built-in speaker, a second point where sound through headphone jack stopped working. Even with 3.2 non of the mute options is working (I did not try recording via build-in microphone) First commit where speaker stops working: commit 3a93897ea37cbb8277f8a4232c12c0c18168a7db Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Oct 28 01:16:55 2011 +0200 ALSA: hda - Manage unsol tags in hda_jack.c Manage the tags assigned for unsolicited events dynamically together with the jack-detection routines. Basically this is almost same as what we've done in patch_sigmatel.c. Assign the new tag number for each new unsol event, associate with the given NID and the action type, etc. With this change, now all pins looked over in snd_hda_jack_add_kctls() are actually enabled for detection now even if the pins aren't used for jack-retasking by the driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> First commit where headphone stops working (while speaker already does not work anymore): commit 1a97b7f22774b454531f013638b181803fba470f Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Feb 21 11:11:48 2012 +0100 ALSA: hda/realtek - Remove the last static quirks for ALC882 Resitance is futile. The remaining static model quirks for Apple machines with ALC882-compatible codecs are converted to the auto-parser now. We can remove all alc*_quirks.c finally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Attached is the output of `alsa-info --with-devices` under current 3.12.2 kernel and under 3.2.53. As this all seems related to quirks being replaced with auto-parsing. Trying to revert the commits on top of 3.12.x looks unworkable. Is there some extra data I should try to provide or a workaround/fix I could try out? I'm booting in EFI mode using EFI stub (did backport it from 3.3 to 3.2 for bisecting/testing 3.2). Thanks, Bruno

2 7

[alsa-devel] [PATCH V2] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 04 Dec '13

04 Dec '13

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This is the initial codec driver of ALC286. I tested Headset recording and Speaker playback. I tested with snd_soc_register_codec because my kernel doesn't support devm_snd_soc_register_codec yet. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1367 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 89 +++ 5 files changed, 1481 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..6c0c1c9 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool irq_en; /*IRQ enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..755a39d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT286 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -319,6 +320,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT286 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b31615c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt286-objs := rt286.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..e7aef0d --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1367 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt286.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct rt286_platform_data pdata; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[TOTAL_NODE_ID + 1] = { + [NODE_ID_DAC_OUT1] = 0x7f7f, + [NODE_ID_DAC_OUT2] = 0x7f7f, + [NODE_ID_SPDIF] = 0x0000, + [NODE_ID_ADC_IN1] = 0x4343, + [NODE_ID_ADC_IN2] = 0x4343, + [NODE_ID_MIC1] = 0x0000, + [NODE_ID_MIXER_IN] = 0x000b, + [NODE_ID_MIXER_OUT1] = 0x0002, + [NODE_ID_MIXER_OUT2] = 0x0000, + [NODE_ID_SPK_OUT] = 0x0000, + [NODE_ID_HP_OUT] = 0x0000, + [NODE_ID_MIXER_IN1] = 0x0000, + [NODE_ID_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int rt286_read(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb; + unsigned int buf = 0x0; + + verb = (nid << 20) | (vid << 8) | data; + + regmap_read(rt286->regmap, verb, &buf); + return buf; +} + +static int rt286_write(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + reg = (nid << 20) | (vid << 8) | value; + + return regmap_write(rt286->regmap, reg, 0); +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + old = rt286_read(codec, vid | 0x800, nid, data); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + ret = rt286_write(codec, vid, nid, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = rt286_write(codec, AC_VERB_SET_PROC_COEF, + wid, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + return rt286_read(codec, AC_VERB_GET_PROC_COEF, + wid, index); +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + NODE_ID_DAC_OUT1, + NODE_ID_DAC_OUT2, + NODE_ID_ADC_IN1, + NODE_ID_ADC_IN2, + NODE_ID_MIC1, + NODE_ID_DMIC1, + NODE_ID_DMIC2, + NODE_ID_SPK_OUT, + NODE_ID_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { NODE_ID_DAC_OUT1, 0x7f7f }, + { NODE_ID_DAC_OUT2, 0x7f7f }, + { NODE_ID_SPDIF, 0x0000 }, + { NODE_ID_ADC_IN1, 0x4343 }, + { NODE_ID_ADC_IN2, 0x4343 }, + { NODE_ID_MIC1, 0x0000 }, + { NODE_ID_MIXER_IN, 0x000b }, + { NODE_ID_MIXER_OUT1, 0x0002 }, + { NODE_ID_MIXER_OUT2, 0x0000 }, + { NODE_ID_SPK_OUT, 0x0000 }, + { NODE_ID_HP_OUT, 0x0000 }, + { NODE_ID_MIXER_IN1, 0x0005 }, + { NODE_ID_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NODE_ID_DAC_OUT1: + case NODE_ID_DAC_OUT2: + case NODE_ID_ADC_IN1: + case NODE_ID_ADC_IN2: + case NODE_ID_MIC1: + case NODE_ID_SPDIF: + case NODE_ID_MIXER_IN: + case NODE_ID_MIXER_OUT1: + case NODE_ID_MIXER_OUT2: + case NODE_ID_DMIC1: + case NODE_ID_DMIC2: + case NODE_ID_SPK_OUT: + case NODE_ID_HP_OUT: + case NODE_ID_MIXER_IN1: + case NODE_ID_MIXER_IN2: + return true; + default: + return false; + } +} + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x03, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x4, 0x4); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x24); + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, AC_VERB_GET_PROC_COEF, + 0x20, 0x50); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, + AC_VERB_GET_PROC_COEF, 0x20, 0x50); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + rt286_write(codec, + AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x20); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x1001); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + + *mic = false; + } + } else { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + *mic = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_MIC1, 0); + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x33, 0x1, 0x1); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_jack_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int io = 0x8000; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0; + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x2000 | io) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, io) & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, io; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0x2000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x1000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x80); + ucontrol->value.integer.value[1] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", NODE_ID_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", NODE_ID_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", NODE_ID_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", NODE_ID_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", NODE_ID_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +static void hw_configure(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + rt286_write(codec, AC_VERB_SET_POWER_STATE, + rt286_support_power_controls[i], AC_PWRST_D3); + } + + if (rt286->pdata.irq_en) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x33, 0x0002, 0x0002); + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x50, 0x0000); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x19, 0x0816); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x20, 0x0000); + } + + mdelay(10); + + /* Set configuration default */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_MIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC2, 0x00); + + mdelay(10); + + if (rt286->pdata.cbj_en) + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0x5029); + else + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0xb029); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x0008, 0x0000); +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", NODE_ID_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", NODE_ID_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, NODE_ID_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, NODE_ID_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, NODE_ID_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spo_enum, NODE_ID_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, NODE_ID_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x40); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0002); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0000); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, NODE_ID_HP_OUT) & 0x8080; + switch (val) { + case 0x0: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb000); + break; + case 0x8000: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0x9000); + break; + case 0x0080: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xa000); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb080); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x20); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D3); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x40); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + } + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, e->reg, + snd_soc_read(codec, e->reg) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + e->reg, snd_soc_read(codec, e->reg) & 0x1); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", NODE_ID_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", NODE_ID_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", NODE_ID_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", NODE_ID_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, NODE_ID_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, NODE_ID_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_PGA_E("SPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("HPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", NULL, "SPK Mux"}, + {"HPO", NULL, "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_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_codec *codec = rtd->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0018, d_len_code << 3); + pr_debug("format val = 0x%lx\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + } else { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0100); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x63, 0x4, 0x4); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x8); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_OFF: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + hw_configure(codec); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt5640", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT33CA", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +static int __init rt286_modinit(void) +{ + return i2c_add_driver(&rt286_i2c_driver); +} + +module_init(rt286_modinit); + +static void __exit rt286_modexit(void) +{ + i2c_del_driver(&rt286_i2c_driver); +} + +module_exit(rt286_modexit); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..c579c53 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,89 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define NODE_ID_AUDIO_FUNCTION_GROUP 0x01 +#define NODE_ID_DAC_OUT1 0x02 +#define NODE_ID_DAC_OUT2 0x03 +#define NODE_ID_ADC_IN1 0x09 +#define NODE_ID_ADC_IN2 0x08 +#define NODE_ID_MIXER_IN 0x0b +#define NODE_ID_MIXER_OUT1 0x0c +#define NODE_ID_MIXER_OUT2 0x0d +#define NODE_ID_DMIC1 0x12 +#define NODE_ID_DMIC2 0x13 +#define NODE_ID_SPK_OUT 0x14 +#define NODE_ID_MIC1 0x18 +#define NODE_ID_LINE1 0x1a +#define NODE_ID_BEEP 0x1d +#define NODE_ID_SPDIF 0x1e +#define NODE_ID_VENDOR_REGISTERS 0x20 +#define NODE_ID_HP_OUT 0x21 +#define NODE_ID_MIXER_IN1 0x22 +#define NODE_ID_MIXER_IN2 0x23 +#define TOTAL_NODE_ID 0x23 + +#define CONNECTION_INDEX_MIC1 0X0 +#define CONNECTION_INDEX_DMIC 0X5 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] ASUS Phoebus (CMI8888) HDA Support
by Geoffrey McRae 04 Dec '13

04 Dec '13

Hi, I have spent the better part of the last two days trying to add support to the hda-intel driver for this card and I have finally got it functional, not through any extra code though :( To get it going I added the extra device info into the device list and specified it as a generic device with no msi and LPIB position fix. Initially this did not work, probing for codec info failed, the interrupt was getting received, but the DMA address contained no data (all 0x0). After messing around a heap trying to find out why there was no DMA data, I discovered that if I forced the device into single_cmd mode it would operate, with very nasty scratchy audio, seems the IRQs were not getting through and the circular buffer was under running. If I made did something involving IRQs such as cat /proc/interrupts, it would briefly fix the problem... if I wrote a while loop to do just that, I would get mostly clean audio. There is another thread in this mailing list where someone else has tried all of this and ended up with the same output. I decided to try to identify why the device was not returning RIRB data in the DMA responses, and out of desperation setup a KVM with windows to try to perform mmio tracing to identify where the hda-intel driver was going wrong. This did not work, windows would not detect the codecs attached to the device, I think it is due to a lack of MSI support for passthrough devices. A positive thing did come out of this though, to do this I enabled MMIO in my BIOS, which instantly fixed the missing RIRB DMA data issue... I do not know enough about MMIO to understand why this fixed this. The device now detects the codec without single_cmd mode, but alas, still no audio, but also no errors... further experimentation I found that if I started the device in single_cmd mode and then went back to normal, I would get perfect audio now. Since I did this, I get perfect audio even without entering single_cmd mode. I have dumped the registers under windows and can confirm that during its initialization of the HDA device it does indeed use single_cmd mode before switching over to IRQ/DMA mode, there might be some standard init the driver is missing here. So, to make this card work, very little extra code will be required, just need to understand why enabling MMIO fixed the DMA issue as this is not a viable fix for the problem, not everyone has MMIO. Here is playback of a stereo stream while the device is working: [ 66.697114] hda_intel: Disabling MSI [ 66.697116] ALSA hda_intel.c:2553 Using LPIB position fix [ 66.697146] snd_hda_intel 0000:06:00.0: setting latency timer to 64 [ 66.697151] ALSA hda_intel.c:2796 chipset global capabilities = 0x4300 [ 66.697159] ALSA hda_intel.c:1175 Clearing TCSEL [ 66.724049] ALSA hda_intel.c:1031 codec_mask = 0x2 [ 66.724181] ALSA hda_intel.c:1484 codec #1 probed OK [ 66.734419] ALSA hda_generic.c:586 AUD_IN = 8 [ 66.734422] ALSA hda_generic.c:239 UNMUTE IN: NID=0x8 IDX=0x0 [ 66.734879] ALSA hda_generic.c:239 UNMUTE IN: NID=0x12 IDX=0x0 [ 66.735565] ALSA hda_generic.c:619 [Capture Source] NID=0x8, #SRC=1 [ 66.735567] ALSA hda_generic.c:622 [Front Mic] IDX=0x0 [ 66.735568] ALSA hda_generic.c:291 AUD_OUT found 2 [ 66.735570] ALSA hda_generic.c:239 UNMUTE IN: NID=0xc IDX=0x0 [ 66.735903] ALSA hda_generic.c:224 UNMUTE OUT: NID=0xc [ 66.736136] ALSA hda_generic.c:224 UNMUTE OUT: NID=0xc [ 66.736139] ALSA hda_generic.c:291 AUD_OUT found 6 [ 66.736569] ALSA hda_generic.c:254 CONNECT: NID=0x8 IDX=0x0 [ 66.736572] ALSA hda_generic.c:699 [Capture Switch] NID=0x8, DIR=IN, IDX=0x0 [ 66.736575] ALSA hda_generic.c:725 [Capture Volume] NID=0x8, DIR=IN, IDX=0x0 [ 66.736578] ALSA hda_generic.c:710 [Master Playback Switch] NID=0x2, DIR=OUT [ 66.736580] ALSA hda_generic.c:734 [Master Playback Volume] NID=0x2, DIR=OUT [ 76.125606] ALSA hda_intel.c:1848 azx_pcm_prepare: bufsize=0x10000, format=0x31 [ 76.125615] ALSA hda_codec.c:1414 hda_codec_setup_stream: NID=0x2, stream=0x7, channel=0, format=0x31 [ 93.308666] ALSA hda_codec.c:1477 hda_codec_cleanup_stream: NID=0x2 - Geoffrey McRae

2 8

[alsa-devel] [PATCH] ASoC: rcar: use devm_clk_get() instead of clk_get()
by Kuninori Morimoto 04 Dec '13

04 Dec '13

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a821b14..85936ae 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -653,7 +653,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); - clk = clk_get(dev, name); + clk = devm_clk_get(dev, name); if (IS_ERR(clk)) return PTR_ERR(clk); @@ -714,7 +714,6 @@ void rsnd_ssi_remove(struct platform_device *pdev, int i; for_each_rsnd_ssi(ssi, priv, i) { - clk_put(ssi->clk); if (rsnd_ssi_dma_available(ssi)) rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/2] Only do snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 04 Dec '13

04 Dec '13

This series of patches mainly fixes an issue of snd_soc_dai_digital_mute() operation after playback/capture is done. The current system has two of them in a normal PCM instance while one of them has an invalid condition. Thus we first fix the condition and then drop the redundant one. Changelog: v2: * Bisected into two patches. * Fixed indentation in PATCH-1. Nicolin Chen (2): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in soc_pcm_close() sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) -- 1.8.4

2 4

[alsa-devel] ASoC fixes for v3.13
by Mark Brown 04 Dec '13

04 Dec '13

The following changes since commit dc1ccc48159d63eca5089e507c82c7d22ef60839: Linux 3.13-rc2 (2013-11-29 12:57:14 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc2 for you to fetch changes up to 29e248829dc7d44248c69bbd5d40eca152a50cab: Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus (2013-12-03 18:09:00 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A smattering of fixes here, some core ones for the rate combination issues for things other than simple bitmasks, for readback of byte controls and for updating the power of value muxes plus a bunch of driver fixes of varying severity. The warning fix in the i.MX FIQ driver is fixing a warning introduced by a previous fix. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel: sam9x5_wm8731: fix oops when unload module ASoC: wm8731: fix dsp mode configuration Charles Keepax (2): ASoC: wm5110: Remove output OSR and PGA volume controls ASoC: core: Use consistent byte ordering in snd_soc_bytes_get Fabio Estevam (1): ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable Jarkko Nikula (1): ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare Jean-Francois Moine (2): ASoC: kirkwood: Fix invalid S/PDIF format ASoC: kirkwood: Fix erroneous double output while playing Kuninori Morimoto (1): ASoC: rcar: select REGMAP Lars-Peter Clausen (2): ASoC: pcm: Fix rate_max calculation ASoC: pcm: Always honor DAI min and max sample rate constraints Mark Brown (4): ASoC: wm8990: Mark the register map as dirty when powering down Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus Oskar Schirmer (1): ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation Shawn Guo (1): ASoC: core: fix devres parameter in devm_snd_soc_register_card() Stephen Warren (1): ASoC: dapm: Use SND_SOC_DAPM_INIT_REG_VAL in SND_SOC_DAPM_MUX Wei Yongjun (1): ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe() include/sound/soc-dapm.h | 3 ++- sound/soc/atmel/sam9x5_wm8731.c | 2 ++ sound/soc/codecs/wm5110.c | 25 ------------------------- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8990.c | 2 ++ sound/soc/fsl/pcm030-audio-fabric.c | 3 ++- sound/soc/kirkwood/kirkwood-i2s.c | 22 +++++++++++++--------- sound/soc/omap/n810.c | 4 ++-- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 4 ++-- sound/soc/soc-devres.c | 4 ++-- sound/soc/soc-pcm.c | 18 ++++++++++++------ 12 files changed, 42 insertions(+), 50 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits
by Nicolin Chen 04 Dec '13

04 Dec '13

Since we introduced symmetric_channels and symmetric_samplebits, we implement these two features to fsl_ssi so as to drop some no-more-needed code and make the driver neat and clean. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 79 ++++++------------------------------------------- 1 file changed, 9 insertions(+), 70 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 90ff107..f9f4569 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -119,8 +119,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI - * @first_stream: pointer to the stream that was opened first - * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @cpu_dai: the CPU DAI for this device @@ -132,8 +130,6 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; - struct snd_pcm_substream *first_stream; - struct snd_pcm_substream *second_stream; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -438,54 +434,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; - /* - * If this is the first stream opened, then request the IRQ - * and initialize the SSI registers. + /* First, we only do fsl_ssi_setup() when SSI is going to be active. + * Second, fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ - if (!ssi_private->first_stream) { - ssi_private->first_stream = substream; - - /* - * fsl_ssi_setup was already called by ac97_init earlier if - * the driver is in ac97 mode. - */ - if (!ssi_private->imx_ac97) - fsl_ssi_setup(ssi_private); - } else { - if (synchronous) { - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - /* - * This is the second stream open, and we're in - * synchronous mode, so we need to impose sample - * sample size constraints. This is because STCCR is - * used for playback and capture in synchronous mode, - * so there's no way to specify different word - * lengths. - * - * Note that this can cause a race condition if the - * second stream is opened before the first stream is - * fully initialized. We provide some protection by - * checking to make sure the first stream is - * initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample - * rate or size. If the second stream is opened - * before the first stream has received its final - * parameters, then the second stream may be - * constrained to the wrong sample rate or size. - */ - if (first_runtime->sample_bits) { - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); - } - } - - ssi_private->second_stream = substream; - } + if (!dai->active && !ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); return 0; } @@ -615,23 +570,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -/** - * fsl_ssi_shutdown: shutdown the SSI - * - * Shutdown the SSI if there are no other substreams open. - */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - - if (ssi_private->first_stream == substream) - ssi_private->first_stream = ssi_private->second_stream; - - ssi_private->second_stream = NULL; -} - static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); @@ -647,7 +585,6 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, }; @@ -722,7 +659,6 @@ static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { .startup = fsl_ssi_startup, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_ac97_trigger, }; @@ -947,8 +883,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } /* Are the RX and the TX clocks locked? */ - if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); -- 1.8.4

2 4

[alsa-devel] [PATCH] Add no-mmu support for alsa-utils
by Aaron Wu 04 Dec '13

04 Dec '13

From: Aaron Wu <Aaron.wu(a)analog.com> Signed-off-by: Aaron Wu <Aaron.wu(a)analog.com> Add no-mmu support for alsa-utils mainly for Blackfin --- alsactl/init_utils_run.c | 4 ++++ alsaloop/alsaloop.c | 8 ++++++++ configure.in | 1 + 3 files changed, 13 insertions(+) diff --git a/alsactl/init_utils_run.c b/alsactl/init_utils_run.c index dde490b..ac074e6 100644 --- a/alsactl/init_utils_run.c +++ b/alsactl/init_utils_run.c @@ -89,7 +89,11 @@ int run_program0(struct space *space, argv[0] = program; } + #ifdef HAVE_FORK pid = fork(); + #else + pid = vfork(); + #endif switch(pid) { case 0: /* child closes parent ends of pipes */ diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 3b20682..1b1e540 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -861,14 +861,22 @@ int main(int argc, char *argv[]) logit(LOG_CRIT, "daemon() failed: %s\n", strerror(errno)); exit(EXIT_FAILURE); } + #ifdef HAVE_FORK i = fork(); + #else + i = vfork(); + #endif if (i < 0) { logit(LOG_CRIT, "fork() failed: %s\n", strerror(errno)); exit(EXIT_FAILURE); } if (i > 0) { /* wait(&i); */ + #ifdef HAVE_FORK exit(EXIT_SUCCESS); + #else + _exit(EXIT_SUCCESS); + #endif } } diff --git a/configure.in b/configure.in index 60da1fa..f289f9b 100644 --- a/configure.in +++ b/configure.in @@ -25,6 +25,7 @@ then AC_MSG_RESULT($CC) fi +AC_CHECK_FUNC([fork]) AC_PROG_CC dnl AC_PROG_CXX AC_PROG_INSTALL -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix missing ELD info when using jackpoll_ms parameter
by David Henningsson 04 Dec '13

04 Dec '13

In the case of using jackpoll_ms instead of unsol events, the jack was correctly detected, but ELD info was not refreshed on plug-in. And without ELD info, no proper restriction of pcm, which can in turn break sound output on some devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) @Takashi; feel free to add stable if you think it's worth it. (After all using the jackpoll_ms parameter is not something we usually do.) It's still a bug fix so sending to Linus tree should be okay, I believe. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4098196..f0a8ea8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct hdmi_spec *spec = codec->spec; + int pin_idx = pin_nid_to_pin_index(spec, jack->nid); + if (pin_idx < 0) + return; + + if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) + snd_hda_jack_report_sync(codec); +} + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pin_nid; - int pin_idx; struct hda_jack_tbl *jack; int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) return; - pin_nid = jack->nid; jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), + codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); - pin_idx = pin_nid_to_pin_index(spec, pin_nid); - if (pin_idx < 0) - return; - - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + jack_callback(codec, jack); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -2080,7 +2082,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hda_nid_t pin_nid = per_pin->pin_nid; hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid, + codec->jackpoll_interval > 0 ? jack_callback : NULL); } return 0; } -- 1.7.9.5

2 5

[alsa-devel] Independent of model for HP
by Kailang 04 Dec '13

04 Dec '13

Hi Takashi, HP Chrome Book, the headset jack was difference between other chrome book. It need to manual control Mic jack detect. Thanks. Kailang

2 6

Re: [alsa-devel] [pulseaudio-discuss] What-did-you-plugin-dialog
by Raymond Yau 04 Dec '13

04 Dec '13

: > > > can post the output of alsa-info.sh after jack retasking ? > > Before retasking: > > http://www.alsa-project.org/db/?f=fd97b8194e205b4f91127bca74ad7c9b2cd93ab9 > > After retasking blue line in, pin ID 0x1a as Line out (Side): > > http://www.alsa-project.org/db/?f=d56f9a6d0692c9a8d5c88f43100a6cb78407c163 > > Note that retasking did not work: attempting to play a 8-channel wav > file through hw:1,0 resulted in an "Invalid argument" error. > > FWIW, here is what hdajackretask printed to the console: > > 0x11 0x99430130 > 0x12 0x4037c540 > 0x14 0x01014010 > 0x15 0x01011012 > 0x16 0x01016011 > 0x17 0x411111f0 > 0x18 0x01a19050 > 0x19 0x02a19060 > 0x1a 0x01014413 > 0x1b 0x02214020 > 0x1c 0x411111f0 > 0x1d 0x4045e601 > 0x1e 0x01452140 > 0x1f 0x411111f0 > 1 > > you need root privilege for hdajackretask to write the firmware file to /lib/firmware and modify /etc/modprobe.d/alsa-lbase.conf to use the firmware file can you try the patch which add "channel mode" control which allow you to switch from 6ch to 8ch it is not clear that the "Line jack" control is used for detect the blue jack state or reflect the line In is no longer available after it is retasked as output jack

2 7

[alsa-devel] [PATCH - plugins] oss: avoid conflict between reclev and igain mixer ctls
by Luca Bayer 04 Dec '13

04 Dec '13

To fix the following error that happens with Intel HDA driver. $ amixer ALSA lib simple_none.c:1549:(simple_add1) helem (MIXER,'Capture Volume',0,0,0) appears twice or more amixer: Mixer default load error: Invalid argument Signed-off-by: Luca Bayer <kometto(a)web.de> diff --git a/oss/ctl_oss.c b/oss/ctl_oss.c index b8847b1..b05ba18 100644 --- a/oss/ctl_oss.c +++ b/oss/ctl_oss.c @@ -52,7 +52,7 @@ static const char *const vol_devices[SOUND_MIXER_NRDEVICES] = { [SOUND_MIXER_CD] = "CD Playback Volume", [SOUND_MIXER_IMIX] = "Monitor Mix Playback Volume", [SOUND_MIXER_ALTPCM] = "Headphone Playback Volume", - [SOUND_MIXER_RECLEV] = "Capture Volume", + [SOUND_MIXER_RECLEV] = "Master Capture Volume", [SOUND_MIXER_IGAIN] = "Capture Volume", [SOUND_MIXER_OGAIN] = "Playback Volume", [SOUND_MIXER_LINE1] = "Aux Playback Volume",

1 0

[alsa-devel] [PATCH] ASoC: dmaengine: remove SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME
by Stephen Warren 04 Dec '13

04 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> struct snd_dmaengine_pcm_config's new chan_names field implements the same feature, and nothing uses SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME, so remove it. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This follows on from the patch "ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config" which I posted earlier today. include/sound/dmaengine_pcm.h | 6 ------ sound/soc/soc-generic-dmaengine-pcm.c | 13 +------------ 2 files changed, 1 insertion(+), 18 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 87e9d481d7b6..634b0fca3afa 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,7 +61,6 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. - * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { @@ -70,7 +69,6 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; - const char *chan_name; unsigned int fifo_size; }; @@ -100,10 +98,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ac16259490ee..819d5997eece 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -188,8 +188,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; - struct device *dev = rtd->platform->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -210,13 +208,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) - pcm->chan[i] = dma_request_slave_channel(dev, - dma_data->chan_name); - if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -290,9 +281,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, unsigned int i; const char *name; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - !dev->of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (config->dma_dev) { -- 1.8.1.5

1 0

[alsa-devel] [PATCH 1/2] ASoC: jz4740-i2s: Use managed resources
by Lars-Peter Clausen 03 Dec '13

03 Dec '13

Makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 85 ++++++++--------------------------------- 1 file changed, 15 insertions(+), 70 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 4c849a4..a0b6a85 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -432,91 +432,36 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; + struct resource *mem; int ret; - i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); - + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!i2s->mem) { - ret = -ENOENT; - goto err_free; - } - - i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), - pdev->name); - if (!i2s->mem) { - ret = -EBUSY; - goto err_free; - } - - i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); - if (!i2s->base) { - ret = -EBUSY; - goto err_release_mem_region; - } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); - i2s->phys_base = i2s->mem->start; + i2s->phys_base = mem->start; - i2s->clk_aic = clk_get(&pdev->dev, "aic"); - if (IS_ERR(i2s->clk_aic)) { - ret = PTR_ERR(i2s->clk_aic); - goto err_iounmap; - } + i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) + return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); - if (IS_ERR(i2s->clk_i2s)) { - ret = PTR_ERR(i2s->clk_i2s); - goto err_clk_put_aic; - } + i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) + return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); - ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, - &jz4740_i2s_dai, 1); - - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI\n"); - goto err_clk_put_i2s; - } - - return 0; - -err_clk_put_i2s: - clk_put(i2s->clk_i2s); -err_clk_put_aic: - clk_put(i2s->clk_aic); -err_iounmap: - iounmap(i2s->base); -err_release_mem_region: - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); -err_free: - kfree(i2s); - - return ret; -} - -static int jz4740_i2s_dev_remove(struct platform_device *pdev) -{ - struct jz4740_i2s *i2s = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - - clk_put(i2s->clk_i2s); - clk_put(i2s->clk_aic); - iounmap(i2s->base); - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - - kfree(i2s); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4740_i2s_dai, 1); } static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, - .remove = jz4740_i2s_dev_remove, .driver = { .name = "jz4740-i2s", .owner = THIS_MODULE, -- 1.7.10.4

2 2

[alsa-devel] [PATCH - oxygen 1/1] Patch for the Xonar DG Patch by Roman Volkov
by Marius Graefe 03 Dec '13

03 Dec '13

I just tested Roman Volkovs oxygen patch for the xonar dg and it works like a charm, however I needed to change two lines in xonar_dg_mixer.c: 1) #include <linux/export.h> was missing which resulted in a warning when compiling. 2) I had to assign a .shortname in xonar_dg_model, otherwise my kernel (unmodified ubuntu 3.2.0-56-generic-pae) throws an oops, the backtrace led me to the missing string. Actually this is weird because the shortname is assigned in get_oxygen_model in oxygen.c, and alsamixer also shows that name later, but at some point in the kernel startup sequence it's NULL. Volkovs Xonar DG 4-Patch series is required! Signed-off-by: Marius Graefe <alsa(a)mgraefe.de> diff --git a/sound/pci/oxygen/xonar_dg_mixer.c b/sound/pci/oxygen/xonar_dg_mixer.c index 39a0da7..102cfce 100644 --- a/sound/pci/oxygen/xonar_dg_mixer.c +++ b/sound/pci/oxygen/xonar_dg_mixer.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/delay.h> +#include <linux/export.h> #include <sound/control.h> #include <sound/core.h> #include <sound/info.h> @@ -456,6 +457,7 @@ static int xonar_control_filter(struct snd_kcontrol_new *template) } struct oxygen_model model_xonar_dg = { + .shortname = "Xonar DG/DGX", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda - fixup ALC262 to skip depop delay before D3 on Intel BayleyBay
by mengdong.lin＠intel.com 03 Dec '13

03 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'. And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce codec suspend time. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Kailang Yang <kailang(a)realtek.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21dcb7b..69f11e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -374,8 +374,10 @@ static void alc_fixup_no_depop_delay(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PROBE) + if (action == HDA_FIXUP_ACT_PROBE) { spec->no_depop_delay = 1; + codec->depop_delay = 0; + } } static int alc_auto_parse_customize_define(struct hda_codec *codec) -- 1.8.1.2

2 1

[alsa-devel] [TINYCOMPRESS PATCH 1/2] crec: Initial version of a compressed capture utility
by Charles Keepax 03 Dec '13

03 Dec '13

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 425 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 448 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..d2fd873 --- /dev/null +++ b/crec.c @@ -0,0 +1,425 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + FILE *file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = fopen(name, "w+b"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error writing output file header: %d\n", + ferror(file)); + goto file_exit; + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = compress_get_alsa_rate(rate); + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = fwrite(buffer, 1, read, file); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %d\n", + ferror(file)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = fseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", stderror(errno)); + goto buf_exit; + } + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error updating output file header: %d\n", + ferror(file)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + fclose(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + fclose(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

2 3

[alsa-devel] [PATCH] ASoC: wm8731: fix dsp mode configuration
by Bo Shen 03 Dec '13

03 Dec '13

According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0. So, fix LRP for DSP mode as the datesheet specification. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c..bc7472c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: soc-pcm: Only do snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 03 Dec '13

03 Dec '13

This patch removed the redundant snd_soc_dai_digital_mute() in close() since it is better to mute in hw_free() which is slightly earlier and symmetrical for the case of reconfiguration, 'aplay 44100.wav 48000.wav' for example: open()->hw_params()->prepare(unmute)->playing->hw_free(mute)->hw_params()-> parepare(unmute)->playing->hw_free(mute)->close() This patch also corrects the condition of the snd_soc_dai_digital_mute() in hw_free() because it would not be executed as we only decrease codec->active in snd_soc_close(). Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb340a8..eac82858 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -474,11 +474,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); - if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -689,7 +684,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -707,7 +702,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - if (!codec->active) + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- 1.8.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: ux500_pcm: Differentiate between pdata and DT initialisation
by Lee Jones 03 Dec '13

03 Dec '13

If booting with full DT support (i.e. DMA too, the last piece of the puzzle), then we don't need to use the compatible_request_channel call back or require some of the historical bumph which probably isn't required by a platform data start-up now either. This will also be ripped out in upcoming commits. Cc: alsa-devel(a)alsa-project.org Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 2f1bdb7..ad0d420 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -134,15 +134,24 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .prepare_slave_config = ux500_pcm_prepare_slave_config, }; +static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = { + .prepare_slave_config = ux500_pcm_prepare_slave_config, +}; + int ux500_pcm_register_platform(struct platform_device *pdev) { + const struct snd_dmaengine_pcm_config *pcm_config; + struct device_node *np = pdev->dev.of_node; int ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, - &ux500_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_COMPAT | - SND_DMAENGINE_PCM_FLAG_NO_DT); + if (np) + pcm_config = &ux500_dmaengine_of_pcm_config; + else + pcm_config = &ux500_dmaengine_pcm_config; + + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", -- 1.8.3.2

3 7

[alsa-devel] [PATCH] ASoC: ssm2602: Use core for applying symmetry constraints
by Lars-Peter Clausen 03 Dec '13

03 Dec '13

Let the core take care of applying sample rate and sample bits constraints instead of open-coding this in the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 50 ++-------------------------------------------- 1 file changed, 2 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 480074d..c6dd485 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,8 +53,6 @@ enum ssm2602_type { struct ssm2602_priv { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; struct regmap *regmap; @@ -277,11 +275,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); unsigned int iface; - if (substream == ssm2602->slave_substream) { - dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - if (srate < 0) return srate; @@ -314,33 +307,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; if (ssm2602->sysclk_constraints) { snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -351,19 +317,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; -} - - static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); @@ -530,7 +483,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, @@ -551,6 +503,8 @@ static struct snd_soc_dai_driver ssm2602_dai = { .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, .ops = &ssm2602_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, }; static int ssm2602_suspend(struct snd_soc_codec *codec) -- 1.8.0

2 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa: Disable timer-scheduling for PCMs with the BATCH flag
by Raymond Yau 03 Dec '13

03 Dec '13

> > > > PCM Devices which have the BATCH flag set update the PCM pointer > only with > > period size granularity. Using timer based scheduling does not have > any > > advantage in this mode. For one devices which have that flag set > usually > > update > > the position pointer in software after getting the period interrupt. > So > > disabling the period interrupt is not possible for this kind of > devices. > > Furthermore writing to or reading from the buffer slice for the > current > > period > > is not possible since the position inside the buffer is not known. On > > the other > > hand the tsched algorithm seems to get easily confused for this kind > of > > hardware, which results in garbled audio output. This typically means > > that timer > > based scheduling needs to be manually disabled on systems with such > devices. > > Auto disabling tsched in this case allows these systems to run with > the > > default > > configuration. > > > > > > If the playback position is reported in steps instead of monotonic > increasing ? > > > > does this mean that you also need to increase rewind_safeguard to one > period > > or modifiy snd_pcm_rewindable ? > > > Yes that makes sense. The safeguard should probably be 1.5 periods or 1 > period + fixed value, since it will always take some time from when the > hardware reaches the next period to when userspace is notified about this. > > I'm unfortunately not that familiar with pulseaudio. Maybe there is a > better > way to fix this problem then to disable tsched. If anybody wants to > recreate > the issue it can easily be emulated by applying the following patch to your > kernel: > > does all alsa drivers with SNDRV_PCM_INFO_PATCH update the playback/capture position in period interrupt ? usb-audio seem update when each URB send/receive (in 1ms intervals) ymfpci driver also update hwptr in 5ms interval but no SNDRV_PCM_INFO_PATCH and no constraint in period time must be interval of 5 ms the rewind_safeguard is affected by the size of FIFO buffer as you cannot rewind any data in FIFO buffer of DSP/ sound controller

3 4

[alsa-devel] [PATCHv8] dmaengine: Add support for BCM2835
by Florian Meier 03 Dec '13

03 Dec '13

Add support for DMA controller of BCM2835 as used in the Raspberry Pi. Currently it only supports cyclic DMA. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- This version includes some more style improvements suggested in the previous thread. .../devicetree/bindings/dma/bcm2835-dma.txt | 57 ++ drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/bcm2835-dma.c | 727 +++++++++++++++++++++ 4 files changed, 791 insertions(+) create mode 100644 Documentation/devicetree/bindings/dma/bcm2835-dma.txt create mode 100644 drivers/dma/bcm2835-dma.c diff --git a/Documentation/devicetree/bindings/dma/bcm2835-dma.txt b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt new file mode 100644 index 0000000..e7c31c1 --- /dev/null +++ b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt @@ -0,0 +1,57 @@ +* BCM2835 DMA controller + +The BCM2835 DMA controller has 16 channels in total. +Only the lower 13 channels have an associated IRQ. +Some arbitrary channels are used by the firmware +(1,3,6,7 in the current firmware version). +The channels 0,2 and 3 have special functionality +and should not be used by the driver. + +Required properties: +- compatible: Should be "brcm,bcm2835-dma". +- reg: Should contain DMA registers location and length. +- interrupts: Should contain the DMA interrupts associated + to the DMA channels in ascending order. +- #dma-cells: Must be <1>, the cell in the dmas property of the + client device represents the DREQ number. +- brcm,dma-channel-mask: Bit mask representing the channels + not used by the firmware in ascending order, + i.e. first channel corresponds to LSB. + +Example: + +dma: dma@7e007000 { + compatible = "brcm,bcm2835-dma"; + reg = <0x7e007000 0xf00>; + interrupts = <1 16> + <1 17> + <1 18> + <1 19> + <1 20> + <1 21> + <1 22> + <1 23> + <1 24> + <1 25> + <1 26> + <1 27> + <1 28>; + + #dma-cells = <1>; + brcm,dma-channel-mask = <0x7f35>; +}; + +DMA clients connected to the BCM2835 DMA controller must use the format +described in the dma.txt file, using a two-cell specifier for each channel. + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = < 0x7e203000 0x20> + < 0x7e101098 0x02>; + + dmas = <&dma 2> + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/drivers/dma/Kconfig b/drivers/dma/Kconfig index c61a6ec..880e723 100644 --- a/drivers/dma/Kconfig +++ b/drivers/dma/Kconfig @@ -300,6 +300,12 @@ config DMA_OMAP select DMA_ENGINE select DMA_VIRTUAL_CHANNELS +config DMA_BCM2835 + tristate "BCM2835 DMA engine support" + depends on (ARCH_BCM2835 || MACH_BCM2708) + select DMA_ENGINE + select DMA_VIRTUAL_CHANNELS + config TI_CPPI41 tristate "AM33xx CPPI41 DMA support" depends on ARCH_OMAP diff --git a/drivers/dma/Makefile b/drivers/dma/Makefile index 0ce2da9..0a6f08e 100644 --- a/drivers/dma/Makefile +++ b/drivers/dma/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_EP93XX_DMA) += ep93xx_dma.o obj-$(CONFIG_DMA_SA11X0) += sa11x0-dma.o obj-$(CONFIG_MMP_TDMA) += mmp_tdma.o obj-$(CONFIG_DMA_OMAP) += omap-dma.o +obj-$(CONFIG_DMA_BCM2835) += bcm2835-dma.o obj-$(CONFIG_MMP_PDMA) += mmp_pdma.o obj-$(CONFIG_DMA_JZ4740) += dma-jz4740.o obj-$(CONFIG_TI_CPPI41) += cppi41.o diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c new file mode 100644 index 0000000..a9d34df --- /dev/null +++ b/drivers/dma/bcm2835-dma.c @@ -0,0 +1,727 @@ +/* + * BCM2835 DMA engine support + * + * This driver only supports cyclic DMA transfers + * as needed for the I2S module. + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * Based on + * OMAP DMAengine support by Russell King + * + * BCM2708 DMA Driver + * Copyright (C) 2010 Broadcom + * + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * MARVELL MMP Peripheral DMA Driver + * Copyright 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/spinlock.h> +#include <linux/of.h> +#include <linux/of_dma.h> + +#include "virt-dma.h" + +struct bcm2835_dmadev { + struct dma_device ddev; + spinlock_t lock; + void __iomem *base; + struct device_dma_parameters dma_parms; +}; + +struct bcm2835_dma_cb { + uint32_t info; + uint32_t src; + uint32_t dst; + uint32_t length; + uint32_t stride; + uint32_t next; + uint32_t pad[2]; +}; + +struct bcm2835_chan { + struct virt_dma_chan vc; + struct list_head node; + + struct dma_slave_config cfg; + bool cyclic; + unsigned int dreq; + + int ch; + struct bcm2835_desc *desc; + + void __iomem *chan_base; + int irq_number; +}; + +struct bcm2835_desc { + struct virt_dma_desc vd; + enum dma_transfer_direction dir; + + unsigned int control_block_size; + struct bcm2835_dma_cb *control_block_base; + dma_addr_t control_block_base_phys; + + unsigned int frames; + size_t size; +}; + +#define BCM2835_DMA_CS 0x00 +#define BCM2835_DMA_ADDR 0x04 +#define BCM2835_DMA_SOURCE_AD 0x0c +#define BCM2835_DMA_DEST_AD 0x10 +#define BCM2835_DMA_NEXTCB 0x1C + +/* DMA CS Control and Status bits */ +#define BCM2835_DMA_ACTIVE BIT(0) +#define BCM2835_DMA_INT BIT(2) +#define BCM2835_DMA_ISPAUSED BIT(4) /* Pause requested or not active */ +#define BCM2835_DMA_ISHELD BIT(5) /* Is held by DREQ flow control */ +#define BCM2835_DMA_ERR BIT(8) +#define BCM2835_DMA_ABORT BIT(30) /* Stop current CB, go to next, WO */ +#define BCM2835_DMA_RESET BIT(31) /* WO, self clearing */ + +#define BCM2835_DMA_INT_EN BIT(0) +#define BCM2835_DMA_D_INC BIT(4) +#define BCM2835_DMA_D_DREQ BIT(6) +#define BCM2835_DMA_S_INC BIT(8) +#define BCM2835_DMA_S_DREQ BIT(10) + +#define BCM2835_DMA_PER_MAP(x) ((x) << 16) + +#define BCM2835_DMA_DATA_TYPE_S8 1 +#define BCM2835_DMA_DATA_TYPE_S16 2 +#define BCM2835_DMA_DATA_TYPE_S32 4 +#define BCM2835_DMA_DATA_TYPE_S128 16 + +#define BCM2835_DMA_BULK_MASK BIT(0) +#define BCM2835_DMA_FIQ_MASK (BIT(2) | BIT(3)) + +/* Valid only for channels 0 - 14, 15 has its own base address */ +#define BCM2835_DMA_CHAN(n) ((n) << 8) /* Base address */ +#define BCM2835_DMA_CHANIO(base, n) ((base) + BCM2835_DMA_CHAN(n)) + +static inline struct bcm2835_dmadev *to_bcm2835_dma_dev(struct dma_device *d) +{ + return container_of(d, struct bcm2835_dmadev, ddev); +} + +static inline struct bcm2835_chan *to_bcm2835_dma_chan(struct dma_chan *c) +{ + return container_of(c, struct bcm2835_chan, vc.chan); +} + +static inline struct bcm2835_desc *to_bcm2835_dma_desc( + struct dma_async_tx_descriptor *t) +{ + return container_of(t, struct bcm2835_desc, vd.tx); +} + +static void bcm2835_dma_desc_free(struct virt_dma_desc *vd) +{ + struct bcm2835_desc *desc = container_of(vd, struct bcm2835_desc, vd); + dma_free_coherent(desc->vd.tx.chan->device->dev, + desc->control_block_size, + desc->control_block_base, + desc->control_block_base_phys); + kfree(desc); +} + +static int bcm2835_dma_abort(void __iomem *chan_base) +{ + unsigned long cs; + long int timeout = 10000; + + cs = readl(chan_base + BCM2835_DMA_CS); + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Write 0 to the active bit - Pause the DMA */ + writel(0, chan_base + BCM2835_DMA_CS); + + /* Wait for any current AXI transfer to complete */ + while ((cs & BCM2835_DMA_ISPAUSED) && --timeout) { + cpu_relax(); + cs = readl(chan_base + BCM2835_DMA_CS); + } + + /* We'll un-pause when we set of our next DMA */ + if (!timeout) + return -ETIMEDOUT; + + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Terminate the control block chain */ + writel(0, chan_base + BCM2835_DMA_NEXTCB); + + /* Abort the whole DMA */ + writel(BCM2835_DMA_ABORT | BCM2835_DMA_ACTIVE, + chan_base + BCM2835_DMA_CS); + + return 0; +} + +static void bcm2835_dma_start_desc(struct bcm2835_chan *c) +{ + struct virt_dma_desc *vd = vchan_next_desc(&c->vc); + struct bcm2835_desc *d; + + if (!vd) { + c->desc = NULL; + return; + } + + list_del(&vd->node); + + c->desc = d = to_bcm2835_dma_desc(&vd->tx); + + writel(d->control_block_base_phys, c->chan_base + BCM2835_DMA_ADDR); + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); +} + +static irqreturn_t bcm2835_dma_callback(int irq, void *data) +{ + struct bcm2835_chan *c = data; + struct bcm2835_desc *d; + unsigned long flags; + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT, c->chan_base + BCM2835_DMA_CS); + + d = c->desc; + + if (d) { + /* TODO Only works for cyclic DMA */ + vchan_cyclic_callback(&d->vd); + } + + /* Keep the DMA engine running */ + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return IRQ_HANDLED; +} + +static int bcm2835_dma_alloc_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + dev_dbg(c->vc.chan.device->dev, + "Allocating DMA channel %d\n", c->ch); + + return request_irq(c->irq_number, + bcm2835_dma_callback, 0, "DMA IRQ", c); +} + +static void bcm2835_dma_free_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + vchan_free_chan_resources(&c->vc); + free_irq(c->irq_number, c); + + dev_dbg(c->vc.chan.device->dev, "Freeing DMA channel %u\n", c->ch); +} + +static size_t bcm2835_dma_desc_size(struct bcm2835_desc *d) +{ + return d->size; +} + +static size_t bcm2835_dma_desc_size_pos(struct bcm2835_desc *d, dma_addr_t addr) +{ + unsigned int i; + size_t size; + + for (size = i = 0; i < d->frames; i++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[i]; + size_t this_size = control_block->length; + dma_addr_t dma; + + if (d->dir == DMA_DEV_TO_MEM) + dma = control_block->dst; + else + dma = control_block->src; + + if (size) + size += this_size; + else if (addr >= dma && addr < dma + this_size) + size += dma + this_size - addr; + } + + return size; +} + +static enum dma_status bcm2835_dma_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *txstate) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + struct virt_dma_desc *vd; + enum dma_status ret; + unsigned long flags; + + ret = dma_cookie_status(chan, cookie, txstate); + if (ret == DMA_SUCCESS || !txstate) + return ret; + + spin_lock_irqsave(&c->vc.lock, flags); + vd = vchan_find_desc(&c->vc, cookie); + if (vd) { + txstate->residue = + bcm2835_dma_desc_size(to_bcm2835_dma_desc(&vd->tx)); + } else if (c->desc && c->desc->vd.tx.cookie == cookie) { + struct bcm2835_desc *d = c->desc; + dma_addr_t pos; + + if (d->dir == DMA_MEM_TO_DEV) + pos = readl(c->chan_base + BCM2835_DMA_SOURCE_AD); + else if (d->dir == DMA_DEV_TO_MEM) + pos = readl(c->chan_base + BCM2835_DMA_DEST_AD); + else + pos = 0; + + txstate->residue = bcm2835_dma_desc_size_pos(d, pos); + } else { + txstate->residue = 0; + } + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return ret; +} + +static void bcm2835_dma_issue_pending(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + unsigned long flags; + + c->cyclic = true; /* Nothing else is implemented */ + + spin_lock_irqsave(&c->vc.lock, flags); + if (vchan_issue_pending(&c->vc) && !c->desc) + bcm2835_dma_start_desc(c); + + spin_unlock_irqrestore(&c->vc.lock, flags); +} + +static struct dma_async_tx_descriptor *bcm2835_dma_prep_dma_cyclic( + struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len, + size_t period_len, enum dma_transfer_direction direction, + unsigned long flags, void *context) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + enum dma_slave_buswidth dev_width; + struct bcm2835_desc *d; + dma_addr_t dev_addr; + unsigned int es, sync_type; + unsigned int frame; + + /* Grab configuration */ + if (!is_slave_direction(direction)) { + dev_err(chan->device->dev, "%s: bad direction?\n", __func__); + return NULL; + } + + if (direction == DMA_DEV_TO_MEM) { + dev_addr = c->cfg.src_addr; + dev_width = c->cfg.src_addr_width; + sync_type = BCM2835_DMA_S_DREQ; + } else { + dev_addr = c->cfg.dst_addr; + dev_width = c->cfg.dst_addr_width; + sync_type = BCM2835_DMA_D_DREQ; + } + + /* Bus width translates to the element size (ES) */ + switch (dev_width) { + case DMA_SLAVE_BUSWIDTH_4_BYTES: + es = BCM2835_DMA_DATA_TYPE_S32; + break; + default: + return NULL; + } + + /* Now allocate and setup the descriptor. */ + d = kzalloc(sizeof(*d), GFP_NOWAIT); + if (!d) + return NULL; + + d->dir = direction; + d->frames = buf_len / period_len; + + /* Allocate memory for control blocks */ + d->control_block_size = d->frames * sizeof(struct bcm2835_dma_cb); + d->control_block_base = dma_zalloc_coherent(chan->device->dev, + d->control_block_size, &d->control_block_base_phys, + GFP_NOWAIT); + + if (!d->control_block_base) { + kfree(d); + return NULL; + } + + /* + * Iterate over all frames, create a control block + * for each frame and link them together. + */ + for (frame = 0; frame < d->frames; frame++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[frame]; + + /* Setup adresses */ + if (d->dir == DMA_DEV_TO_MEM) { + control_block->info = BCM2835_DMA_D_INC; + control_block->src = dev_addr; + control_block->dst = buf_addr + frame * period_len; + } else { + control_block->info = BCM2835_DMA_S_INC; + control_block->src = buf_addr + frame * period_len; + control_block->dst = dev_addr; + } + + /* Enable interrupt */ + control_block->info |= BCM2835_DMA_INT_EN; + + /* Setup synchronization */ + if (sync_type != 0) + control_block->info |= sync_type; + + /* Setup DREQ channel */ + if (c->dreq != 0) + control_block->info |= + BCM2835_DMA_PER_MAP(c->dreq); + + /* Length of a frame */ + control_block->length = period_len; + d->size += control_block->length; + + /* + * Next block is the next frame. + * This DMA engine driver currently only supports cyclic DMA. + * Therefore, wrap around at number of frames. + */ + control_block->next = d->control_block_base_phys + + sizeof(struct bcm2835_dma_cb) + * ((frame + 1) % d->frames); + } + + return vchan_tx_prep(&c->vc, &d->vd, flags); +} + +static int bcm2835_dma_slave_config(struct bcm2835_chan *c, + struct dma_slave_config *cfg) +{ + if ((cfg->direction == DMA_DEV_TO_MEM && + cfg->src_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + (cfg->direction == DMA_MEM_TO_DEV && + cfg->dst_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + !is_slave_direction(cfg->direction)) { + return -EINVAL; + } + + c->cfg = *cfg; + + return 0; +} + +static int bcm2835_dma_terminate_all(struct bcm2835_chan *c) +{ + struct bcm2835_dmadev *d = to_bcm2835_dma_dev(c->vc.chan.device); + unsigned long flags; + int timeout = 10000; + LIST_HEAD(head); + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Prevent this channel being scheduled */ + spin_lock(&d->lock); + list_del_init(&c->node); + spin_unlock(&d->lock); + + /* + * Stop DMA activity: we assume the callback will not be called + * after bcm_dma_abort() returns (even if it does, it will see + * c->desc is NULL and exit.) + */ + if (c->desc) { + c->desc = NULL; + bcm2835_dma_abort(c->chan_base); + + /* Wait for stopping */ + while (--timeout) { + if (!(readl(c->chan_base + BCM2835_DMA_CS) & + BCM2835_DMA_ACTIVE)) + break; + + cpu_relax(); + } + + if (!timeout) + dev_err(d->ddev.dev, "DMA transfer could not be terminated\n"); + } + + vchan_get_all_descriptors(&c->vc, &head); + spin_unlock_irqrestore(&c->vc.lock, flags); + vchan_dma_desc_free_list(&c->vc, &head); + + return 0; +} + +static int bcm2835_dma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, + unsigned long arg) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + switch (cmd) { + case DMA_SLAVE_CONFIG: + return bcm2835_dma_slave_config(c, + (struct dma_slave_config *)arg); + + case DMA_TERMINATE_ALL: + return bcm2835_dma_terminate_all(c); + + default: + return -ENXIO; + } +} + +static int bcm2835_dma_chan_init(struct bcm2835_dmadev *d, int chan_id, int irq) +{ + struct bcm2835_chan *c; + + c = devm_kzalloc(d->ddev.dev, sizeof(*c), GFP_KERNEL); + if (!c) + return -ENOMEM; + + c->vc.desc_free = bcm2835_dma_desc_free; + vchan_init(&c->vc, &d->ddev); + INIT_LIST_HEAD(&c->node); + + d->ddev.chancnt++; + + c->chan_base = BCM2835_DMA_CHANIO(d->base, chan_id); + c->ch = chan_id; + c->irq_number = irq; + + return 0; +} + +static void bcm2835_dma_free(struct bcm2835_dmadev *od) +{ + while (!list_empty(&od->ddev.channels)) { + struct bcm2835_chan *c = list_first_entry(&od->ddev.channels, + struct bcm2835_chan, vc.chan.device_node); + + list_del(&c->vc.chan.device_node); + tasklet_kill(&c->vc.task); + } +} + +#if defined(CONFIG_OF) +static const struct of_device_id bcm2835_dma_of_match[] = { + { .compatible = "brcm,bcm2835-dma", }, + {}, +}; +MODULE_DEVICE_TABLE(of, bcm2835_dma_of_match); +#endif + +static struct dma_chan *bcm2835_dma_xlate(struct of_phandle_args *spec, + struct of_dma *ofdma) +{ + struct bcm2835_dmadev *d = ofdma->of_dma_data; + struct dma_chan *chan, *candidate; + +retry: + candidate = NULL; + + /* Walk the list of channels registered with the current instance and + * find one that is currently unused */ + list_for_each_entry(chan, &d->ddev.channels, device_node) + if (chan->client_count == 0) { + candidate = chan; + break; + } + + if (!candidate) + return NULL; + + /* dma_get_slave_channel will return NULL if we lost a race between + * the lookup and the reservation */ + chan = dma_get_slave_channel(candidate); + + if (!chan) + goto retry; + + /* Set DREQ from param */ + to_bcm2835_dma_chan(chan)->dreq = spec->args[0]; + + return chan; +} + +static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, + struct dma_slave_caps *caps) +{ + caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); + caps->cmd_pause = false; + caps->cmd_terminate = true; + + return 0; +} + +static int bcm2835_dma_probe(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od; + struct resource *res; + void __iomem *base; + int rc; + int i; + int irq; + uint32_t chans_available; + + if (!pdev->dev.dma_mask) + pdev->dev.dma_mask = &pdev->dev.coherent_dma_mask; + + rc = dma_set_mask(&pdev->dev, DMA_BIT_MASK(32)); + if (rc) + return rc; + dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); + + od = devm_kzalloc(&pdev->dev, sizeof(*od), GFP_KERNEL); + if (!od) + return -ENOMEM; + + pdev->dev.dma_parms = &od->dma_parms; + dma_set_max_seg_size(&pdev->dev, 0x3FFFFFFF); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + od->base = base; + + dma_cap_set(DMA_SLAVE, od->ddev.cap_mask); + dma_cap_set(DMA_CYCLIC, od->ddev.cap_mask); + od->ddev.device_alloc_chan_resources = bcm2835_dma_alloc_chan_resources; + od->ddev.device_free_chan_resources = bcm2835_dma_free_chan_resources; + od->ddev.device_tx_status = bcm2835_dma_tx_status; + od->ddev.device_issue_pending = bcm2835_dma_issue_pending; + od->ddev.device_slave_caps = bcm2835_dma_device_slave_caps; + od->ddev.device_prep_dma_cyclic = bcm2835_dma_prep_dma_cyclic; + od->ddev.device_control = bcm2835_dma_control; + od->ddev.dev = &pdev->dev; + INIT_LIST_HEAD(&od->ddev.channels); + spin_lock_init(&od->lock); + + platform_set_drvdata(pdev, od); + + /* Request DMA channel mask from device tree */ + if (of_property_read_u32(pdev->dev.of_node, + "brcm,dma-channel-mask", + &chans_available)) { + dev_err(&pdev->dev, "Failed to get channel mask\n"); + rc = -EINVAL; + goto err_no_dma; + } + + /* Do not use the FIQ and BULK channels, + * because they are used by the GPU. + */ + chans_available &= ~(BCM2835_DMA_FIQ_MASK | BCM2835_DMA_BULK_MASK); + + for (i = 0; i < pdev->num_resources; i++) { + irq = platform_get_irq(pdev, i); + if (irq < 0) + break; + + if (chans_available & (1 << i)) { + rc = bcm2835_dma_chan_init(od, i, irq); + if (rc) + goto err_no_dma; + } + } + + dev_dbg(&pdev->dev, "Initialized %i DMA channels\n", i); + + if (pdev->dev.of_node) { + /* Device-tree DMA controller registration */ + rc = of_dma_controller_register(pdev->dev.of_node, + bcm2835_dma_xlate, od); + if (rc) { + dev_err(&pdev->dev, "Failed to register DMA controller\n"); + goto err_no_dma; + } + } + + rc = dma_async_device_register(&od->ddev); + if (rc) { + dev_err(&pdev->dev, + "Failed to register slave DMA engine device: %d\n", rc); + goto err_no_dma; + } + + dev_dbg(&pdev->dev, "Load BCM2835 DMA engine driver\n"); + + return 0; + +err_no_dma: + bcm2835_dma_free(od); + return rc; +} + +static int bcm2835_dma_remove(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od = platform_get_drvdata(pdev); + + dma_async_device_unregister(&od->ddev); + bcm2835_dma_free(od); + + return 0; +} + +static struct platform_driver bcm2835_dma_driver = { + .probe = bcm2835_dma_probe, + .remove = bcm2835_dma_remove, + .driver = { + .name = "bcm2835-dma", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(bcm2835_dma_of_match), + }, +}; + +module_platform_driver(bcm2835_dma_driver); + +MODULE_ALIAS("platform:bcm2835-dma"); +MODULE_DESCRIPTION("BCM2835 DMA engine driver"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.8.3.2

3 2

[alsa-devel] How should SGTL5000 codec driver adapt to dummy regulator change?
by Shawn Guo 03 Dec '13

03 Dec '13

Mark, The sgtl5000_enable_regulators() function in SGTL5000 codec driver makes the decision whether an external VDDD supply is available by checking the return of regulator_bulk_get() call. It used to work because dummy regulator did not play in there, and if regulator_bulk_get() fails, most likely it's only because there is no external VDDD supply. But with commit 4ddfebd (regulator: core: Provide a dummy regulator with full constraints) merged, the regulator_bulk_get() will always succeed. It breaks the driver's assumption. What's your suggestion to the problem? Shawn

2 1

[alsa-devel] [PATCH 1/3] ASoC: ux500_pcm: Stop pretending that we support varying address widths
by Lee Jones 03 Dec '13

03 Dec '13

The Slave Config's addr_width attribute is populated by data_width of dma_cfg, which in turn is derived from dma_params' data_size attribute and that comes from the slot_width which is always 16 bits (2 Bytes). We're cutting out the middle man here and just setting the DMA Slave Config directly. Cc: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 6162f70..2f1bdb7 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -104,26 +104,25 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct ux500_msp_dma_params *dma_params; - struct stedma40_chan_cfg *dma_cfg; + struct snd_dmaengine_dai_dma_data *dma_params; int ret; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - dma_cfg = dma_params->dma_cfg; ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) return ret; slave_config->dst_maxburst = 4; - slave_config->dst_addr_width = dma_cfg->dst_info.data_width; slave_config->src_maxburst = 4; - slave_config->src_addr_width = dma_cfg->src_info.data_width; + + slave_config->src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - slave_config->dst_addr = dma_params->tx_rx_addr; + slave_config->dst_addr = dma_params->addr; else - slave_config->src_addr = dma_params->tx_rx_addr; + slave_config->src_addr = dma_params->addr; return 0; } -- 1.8.3.2

3 13

[alsa-devel] [PATCH] ALSA: hda - Add missing initialization of aamix paths
by Takashi Iwai 03 Dec '13

03 Dec '13

The loopback mixing paths aren't initialized correctly at init callback. Mostly this is harmless as codecs usually set the mute state as default, but we still should make sure. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c4671d00babd..28baeff99c62 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5138,6 +5138,23 @@ static void init_multi_io(struct hda_codec *codec) } } +static void init_aamix_paths(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + + if (!spec->have_aamix_ctl) + return; + update_aamix_paths(codec, spec->aamix_mode, spec->out_paths[0], + spec->aamix_out_paths[0], + spec->autocfg.line_out_type); + update_aamix_paths(codec, spec->aamix_mode, spec->hp_paths[0], + spec->aamix_out_paths[1], + AUTO_PIN_HP_OUT); + update_aamix_paths(codec, spec->aamix_mode, spec->speaker_paths[0], + spec->aamix_out_paths[2], + AUTO_PIN_SPEAKER_OUT); +} + /* set up input pins and loopback paths */ static void init_analog_input(struct hda_codec *codec) { @@ -5240,6 +5257,7 @@ int snd_hda_gen_init(struct hda_codec *codec) init_multi_out(codec); init_extra_out(codec); init_multi_io(codec); + init_aamix_paths(codec); init_analog_input(codec); init_input_src(codec); init_digital(codec); -- 1.8.5

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek)
by Takashi Iwai 03 Dec '13

03 Dec '13

At Tue, 3 Dec 2013 15:52:37 +0800, Kailang wrote: > > Hi David, > > The verb was ADC digital mute(bit 6 default 1). > Switch internal mic and headset mic will run alc_headset_mode_default. > The coef index 0x11 will set to 0x0041. > Because headset mode was fixed type. It doesn't need to run > alc_determine_headset_type. > So, the value still keep 0x0041. ADC was muted. > Your patch will solve this errors. OK, I added your comment in the changelog and add Cc to stable, then applied to for-linus branch now. thanks, Takashi > > Thanks. > Kailang > > > > By trial and error, I found this patch could work around an issue > > where the headset mic would stop working if you switch between the > > internal mic and the headset mic, and the internal mic was muted. > > > > It still takes a second or two before the headset mic actually starts > > working, but still better than nothing. > > > > I'd like to add stable to this patch, but I'd like Kailang's ack first > > because I don't know what the verbs really do. > > > > BugLink: https://bugs.launchpad.net/bugs/1256840 > > Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> > > --- > > sound/pci/hda/patch_realtek.c | 2 ++ > > 1 file changed, 2 insertions(+) > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > index 537991c..85d1c8b 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -3338,6 +3338,7 @@ static void alc_headset_mode_ctia(struct hda_codec > > *codec) > > alc_write_coef_idx(codec, 0x18, 0x7388); > > break; > > case 0x10ec0668: > > + alc_write_coef_idx(codec, 0x11, 0x0001); > > alc_write_coef_idx(codec, 0x15, 0x0d60); > > alc_write_coef_idx(codec, 0xc3, 0x0000); > > break; > > @@ -3366,6 +3367,7 @@ static void alc_headset_mode_omtp(struct hda_codec > > *codec) > > alc_write_coef_idx(codec, 0x18, 0x7388); > > break; > > case 0x10ec0668: > > + alc_write_coef_idx(codec, 0x11, 0x0001); > > alc_write_coef_idx(codec, 0x15, 0x0d50); > > alc_write_coef_idx(codec, 0xc3, 0x0000); > > break; > > -- > > 1.7.9.5 > > > > > > ------Please consider the environment before printing this e-mail. >

1 0

Re: [alsa-devel] compressed audio support (maruthi srinivas)
by anish singh 03 Dec '13

03 Dec '13

> Hi, > > I saw that compressed audio framework being used > with Asoc (alsa soc). > Can the framework be used with other than Asoc ? > I mean with ALSA pci based cards. Never worked on this so don't know. > I see the implementation in the following files. which one > is to be used ? > Does tinycompress in userspace can be used with either > approaches ? This will not be relevant after you see below answer. > >http://lxr.free-electrons.com/source/sound/core/compress_offload.c >http://lxr.free-electrons.com/source/sound/soc/soc-compress.c Both are compress framework driver files. compress_offload.c file is the one which interacts with userspace by using ioctl's, registers the compressed device with alsa framework(/dev/compress* gets created because of that), calls the corresponding functions such as trigger, pointer update, copy and others when userspace pauses, plays and stops the playback or recording. soc_compress.c file is the file which interacts with compress_offload.c file to create a 'dev' node for compress driver.Basically when you register your compress driver with compress framework,alsa core will come to know that your dai support compress operations and it will create the dev node by using soc_compress.c file.The functions in this file gets called by compress_offload.c functions and who inturn gets called by the userspace. userspace->compress_offload.c->soc_compress.c->your_compress_driver.

2 1

Re: [alsa-devel] [PATCH v2] ASoC: simple-card: add Device Tree support
by Mark Brown 03 Dec '13

03 Dec '13

On Thu, Oct 03, 2013 at 02:33:55AM -0700, Kuninori Morimoto wrote: > ping ? Don't send contentless pings. I'm waiting for review from the device tree guys but I see now that you didn't CC them so they're unlikely to respond - please resend CCing them.

4 31

[alsa-devel] [PATCH 37/39] sound: remove DEFINE_PCI_DEVICE_TABLE macro
by Jingoo Han 03 Dec '13

03 Dec '13

Don't use DEFINE_PCI_DEVICE_TABLE macro, because this macro is not preferred. Signed-off-by: Jingoo Han <jg1.han(a)samsung.com> --- sound/oss/kahlua.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 2 +- sound/pci/als4000.c | 2 +- sound/pci/asihpi/asihpi.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au8810.c | 2 +- sound/pci/au88x0/au8820.c | 2 +- sound/pci/au88x0/au8830.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/cmipci.c | 4 ++-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs5530.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/lola/lola.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sis7019.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- 68 files changed, 71 insertions(+), 71 deletions(-) diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 12be1fb..c4b0434 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -197,7 +197,7 @@ MODULE_LICENSE("GPL"); * 5530 only. The 5510/5520 decode is different. */ -static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { +static const struct pci_device_id id_tbl[] = { { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index b680d03..793cf1e 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1048,7 +1048,7 @@ snd_ad1889_remove(struct pci_dev *pci) snd_card_free(pci_get_drvdata(pci)); } -static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { +static const struct pci_device_id snd_ad1889_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index c6835a3..f7bdcc6 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -275,7 +275,7 @@ struct snd_ali { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_ali_ids) = { +static const struct pci_device_id snd_ali_ids[] = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 591efb6..7bfeed8 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -152,7 +152,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static DEFINE_PCI_DEVICE_TABLE(snd_als300_ids) = { +static const struct pci_device_id snd_als300_ids[] = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index ffc821b..6c9429e 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -116,7 +116,7 @@ struct snd_card_als4000 { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_als4000_ids) = { +static const struct pci_device_id snd_als4000_ids[] = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 5f2acd3..623f1ae 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2960,7 +2960,7 @@ static void snd_asihpi_remove(struct pci_dev *pci_dev) asihpi_adapter_remove(pci_dev); } -static DEFINE_PCI_DEVICE_TABLE(asihpi_pci_tbl) = { +static const struct pci_device_id asihpi_pci_tbl[] = { {HPI_PCI_VENDOR_ID_TI, HPI_PCI_DEV_ID_DSP6205, HPI_PCI_VENDOR_ID_AUDIOSCIENCE, PCI_ANY_ID, 0, 0, (kernel_ulong_t)HPI_6205}, diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index f6dec3e..144d594 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -286,7 +286,7 @@ struct atiixp { /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { +static const struct pci_device_id snd_atiixp_ids[] = { { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 289563e..c6d1148 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -261,7 +261,7 @@ struct atiixp_modem { /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { +static const struct pci_device_id snd_atiixp_ids[] = { { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index aa51cc7..1b2e340 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { +static const struct pci_device_id snd_vortex_ids[] = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index 2f321e7..74c53fa 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { +static const struct pci_device_id snd_vortex_ids[] = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index 279b78f..56f675a 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { +static const struct pci_device_id snd_vortex_ids[] = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 2925220..f05b843 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -160,7 +160,7 @@ MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); -static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { +static const struct pci_device_id snd_aw2_ids[] = { {PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0, 0, 0, 0}, {0} diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 1aef712..4d5b97b 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -376,7 +376,7 @@ struct snd_azf3328 { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_azf3328_ids) = { +static const struct pci_device_id snd_azf3328_ids[] = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 1880203..55a3d1e 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -789,7 +789,7 @@ fail: .driver_data = SND_BT87X_BOARD_ ## id } /* driver_data is the card id for that device */ -static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_ids) = { +static const struct pci_device_id snd_bt87x_ids[] = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ @@ -957,7 +957,7 @@ static void snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { +static const struct pci_device_id snd_bt87x_default_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index f4db558..bbf9f39 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1959,7 +1959,7 @@ static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume); #endif // PCI IDs -static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { +static const struct pci_device_id snd_ca0106_ids[] = { { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 2755ec5..fb73198 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2803,7 +2803,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static DEFINE_PCI_DEVICE_TABLE(snd_cmipci_ids) = { +static const struct pci_device_id snd_cmipci_ids[] = { {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, @@ -3025,7 +3025,7 @@ static int snd_cmipci_create(struct snd_card *card, struct pci_dev *pci, int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; - static DEFINE_PCI_DEVICE_TABLE(intel_82437vx) = { + static const struct pci_device_id intel_82437vx[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, { }, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 1dc793e..3eef727 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_cs4281_ids) = { +static const struct pci_device_id snd_cs4281_ids[] = { { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index b034983..8234fbb 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -64,7 +64,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static DEFINE_PCI_DEVICE_TABLE(snd_cs46xx_ids) = { +static const struct pci_device_id snd_cs46xx_ids[] = { { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index c6b82c8..1f86a42 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -66,7 +66,7 @@ struct snd_cs5530 { unsigned long pci_base; }; -static DEFINE_PCI_DEVICE_TABLE(snd_cs5530_ids) = { +static const struct pci_device_id snd_cs5530_ids[] = { {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0}, {0,} diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 902bebd..0276e12 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); -static DEFINE_PCI_DEVICE_TABLE(snd_cs5535audio_ids) = { +static const struct pci_device_id snd_cs5535audio_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index d464ad2..568b365 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -44,7 +44,7 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); module_param_array(subsystem, int, NULL, 0444); MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); -static DEFINE_PCI_DEVICE_TABLE(ct_pci_dev_ids) = { +static const struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), .driver_data = ATC20K1, diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index d47e72a..4632946 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -63,7 +63,7 @@ static const struct firmware card_fw[] = { {0, "darla20_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 413acf7..f81c839 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "darla24_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 1ec4edc..3a5346c 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -81,7 +81,7 @@ static const struct firmware card_fw[] = { {0, "3g_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */ {0,} }; diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index 039125b..9cb81c5 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "gina20_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index 5e966f6..35d3e6e 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -85,7 +85,7 @@ static const struct firmware card_fw[] = { {0, "gina24_361_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */ diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index c166b7e..8d91842 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */ {0,} }; diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index a3ef3b9..289cb96 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dj_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ {0,} }; diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index f516444..201688e 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_djx_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index c22c82f..405a3f2 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_io_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index 86cf2d0..e145b68 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_iox_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 6a027f3..b392dd7 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -76,7 +76,7 @@ static const struct firmware card_fw[] = { {0, "layla20_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index 96a5991..bc7f730 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -87,7 +87,7 @@ static const struct firmware card_fw[] = { {0, "layla24_2S_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index b8ce27e..27a9a6e 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -77,7 +77,7 @@ static const struct firmware card_fw[] = { {0, "mia_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */ {0,} diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index 1283bfb..3d13875 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -92,7 +92,7 @@ static const struct firmware card_fw[] = { {0, "mona_2_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */ diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 9e1bd0c..7feeb78 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -79,7 +79,7 @@ MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samp /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1_ids) = { +static const struct pci_device_id snd_emu10k1_ids[] = { { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 56ad9d6..6a80ab8 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1626,7 +1626,7 @@ static void snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { +static const struct pci_device_id snd_emu10k1x_ids[] = { { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 61262f3..f5963b6 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -446,7 +446,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_audiopci_ids) = { +static const struct pci_device_id snd_audiopci_ids[] = { #ifdef CHIP1370 { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 9213fb3..7eb87f2 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -243,7 +243,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_es1938_ids) = { +static const struct pci_device_id snd_es1938_ids[] = { { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index b0e3d92..cd1d947 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -570,7 +570,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { +static const struct pci_device_id snd_es1968_ids[] = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 45bc8a9..739f224 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -210,7 +210,7 @@ struct fm801 { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_fm801_ids) = { +static const struct pci_device_id snd_fm801_ids[] = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c6d2301..d1f11a5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3969,7 +3969,7 @@ static void azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { +static const struct pci_device_id azx_ids[] = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 28ec872..b8378e7 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -105,7 +105,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static DEFINE_PCI_DEVICE_TABLE(snd_ice1712_ids) = { +static const struct pci_device_id snd_ice1712_ids[] = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 5004717..4a3165b 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(snd_vt1724_ids) = { +static const struct pci_device_id snd_vt1724_ids[] = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 08d8733..e5cb94e 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -430,7 +430,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0_ids) = { +static const struct pci_device_id snd_intel8x0_ids[] = { { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 3573c11..ab7edfd 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -219,7 +219,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { +static const struct pci_device_id snd_intel8x0m_ids[] = { { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 9cf9829..9413b8a 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -418,7 +418,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal <gamal(a)alternex.com.br>"); -static DEFINE_PCI_DEVICE_TABLE(snd_korg1212_ids) = { +static const struct pci_device_id snd_korg1212_ids[] = { { .vendor = 0x10b5, .device = 0x906d, diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 0568540..a7beffe 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -762,7 +762,7 @@ static void lola_remove(struct pci_dev *pci) } /* PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(lola_ids) = { +static const struct pci_device_id lola_ids[] = { { PCI_VDEVICE(DIGIGRAM, 0x0001) }, { 0, } }; diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 5fcaaa6..a016e39 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -56,7 +56,7 @@ static const char card_name[] = "LX6464ES"; #define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 -static DEFINE_PCI_DEVICE_TABLE(snd_lx6464es_ids) = { +static const struct pci_device_id snd_lx6464es_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), .subvendor = PCI_VENDOR_ID_DIGIGRAM, .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index d541736..836ce95 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -800,7 +800,7 @@ struct snd_m3 { /* * pci ids */ -static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { +static const struct pci_device_id snd_m3_ids[] = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 1e0f6ee..c9fe3e2 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -61,7 +61,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_mixart_ids) = { +static const struct pci_device_id snd_mixart_ids[] = { { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index fe79fff..97eac98 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -262,7 +262,7 @@ struct nm256 { /* * PCI ids */ -static DEFINE_PCI_DEVICE_TABLE(snd_nm256_ids) = { +static const struct pci_device_id snd_nm256_ids[] = { {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ada6c25..74afb6b 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -97,7 +97,7 @@ enum { MODEL_XONAR_DGX, }; -static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { +static const struct pci_device_id oxygen_ids[] = { /* C-Media's reference design */ { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0217), .driver_data = MODEL_CMEDIA_REF }, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..ce33544 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -41,7 +41,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { +static const struct pci_device_id xonar_ids[] = { { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index d379b28..a4b05a2 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -102,7 +102,7 @@ enum { PCI_ID_LAST }; -static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { +static const struct pci_device_id pcxhr_ids[] = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 56cc891..d29fa04 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -508,7 +508,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { +static const struct pci_device_id snd_riptide_ids[] = { { PCI_DEVICE(0x127a, 0x4310) }, { PCI_DEVICE(0x127a, 0x4320) }, { PCI_DEVICE(0x127a, 0x4330) }, @@ -517,7 +517,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { }; #ifdef SUPPORT_JOYSTICK -static DEFINE_PCI_DEVICE_TABLE(snd_riptide_joystick_ids) = { +static const struct pci_device_id snd_riptide_joystick_ids[] = { { PCI_DEVICE(0x127a, 0x4312) }, { PCI_DEVICE(0x127a, 0x4322) }, { PCI_DEVICE(0x127a, 0x4332) }, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index cc26346..ef93838 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -226,7 +226,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static DEFINE_PCI_DEVICE_TABLE(snd_rme32_ids) = { +static const struct pci_device_id snd_rme32_ids[] = { {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 0236363..1211d36 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -263,7 +263,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static DEFINE_PCI_DEVICE_TABLE(snd_rme96_ids) = { +static const struct pci_device_id snd_rme96_ids[] = { { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index f59a321..fd348ce 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -603,7 +603,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static DEFINE_PCI_DEVICE_TABLE(snd_hdsp_ids) = { +static const struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e98dc00..c6bf321 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1077,7 +1077,7 @@ struct hdspm { }; -static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { +static const struct pci_device_id snd_hdspm_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index b96d9e1..18cf23d 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -313,7 +313,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static DEFINE_PCI_DEVICE_TABLE(snd_rme9652_ids) = { +static const struct pci_device_id snd_rme9652_ids[] = { { .vendor = 0x10ee, .device = 0x3fc4, diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index e413b4e..196df18 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -52,7 +52,7 @@ MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); module_param(codecs, int, 0444); MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)"); -static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { +static const struct pci_device_id snd_sis7019_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 2a46bf9..bc929f3 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -242,7 +242,7 @@ struct sonicvibes { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_sonic_ids) = { +static const struct pci_device_id snd_sonic_ids[] = { { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index b3b588b..29a0d32 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static DEFINE_PCI_DEVICE_TABLE(snd_trident_ids) = { +static const struct pci_device_id snd_trident_ids[] = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 5ae6f04..7029142 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -404,7 +404,7 @@ struct via82xx { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_ids) = { +static const struct pci_device_id snd_via82xx_ids[] = { /* 0x1106, 0x3058 */ { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index ca19028..7424c13 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -260,7 +260,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_modem_ids) = { +static const struct pci_device_id snd_via82xx_modem_ids[] = { { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index ab8a9b1..c4fa2f7 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static DEFINE_PCI_DEVICE_TABLE(snd_vx222_ids) = { +static const struct pci_device_id snd_vx222_ids[] = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e8932b2..3e14fd7 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); -static DEFINE_PCI_DEVICE_TABLE(snd_ymfpci_ids) = { +static const struct pci_device_id snd_ymfpci_ids[] = { { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ -- 1.7.10.4

1 0

[alsa-devel] Bug: plugging in a new sound card causes mixer state of other sound cards to be changed
by Angus McInnes 02 Dec '13

02 Dec '13

Hi, When I plug in my USB sound card, the mixer settings of the internal sound card are changed. I don't want this to happen. I believe the problem is with this commit: http://git.alsa-project.org/?p=alsa-utils.git;a=blobdiff;f=alsactl/90-alsa-… The card number $attr{number} comes from the card* device. Since KERNELS=="card*" was changed to KERNELS!="card*", alsactl restore is run with no card number attribute, so it restores the state of all cards. If I change the != back to ==, alsactl restore is run with a card number and the problem seems to be resolved. (By the way, your bug tracker at https://bugtrack.alsa-project.org/alsa-bug seems to be down.) Regards, Angus McInnes

1 0

[alsa-devel] [PATCH] ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare
by Jarkko Nikula 02 Dec '13

02 Dec '13

N810 audio driver has stopped working at some point. Probably when OMAP2 was converted to common clock framework since now call to clk_enable dumps the stack trace in drivers/clk/clk.c: __clk_enable() due clk->prepare_count is zero. Fix this by converting clk_enable/_disable calls to those that take care of clock prepare/unprepare. I'm not queueing this to linux-stable since OMAP2 common clock framework conversion in commit ed1ebc4948fd ("ARM: OMAP2: clock: Convert to common clk") happened before N810 was really usable in mainline and user base for N810 is anyway small. Potential linux-stable candidates are only those after commit 3d3a6d18abc6 ("watchdog: introduce retu_wdt driver"). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- sound/soc/omap/n810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6d216cb6c19b..3fde9e402710 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); n810_ext_control(&codec->dapm); - return clk_enable(sys_clkout2); + return clk_prepare_enable(sys_clkout2); } static void n810_shutdown(struct snd_pcm_substream *substream) { - clk_disable(sys_clkout2); + clk_disable_unprepare(sys_clkout2); } static int n810_hw_params(struct snd_pcm_substream *substream, -- 1.8.4.3

2 4

[alsa-devel] [PATCH v3] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v3: * Fixed conflict for git apply. v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fb8f52a..83e4ed8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -143,6 +143,7 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -354,14 +355,13 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -378,7 +378,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -508,6 +508,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -537,6 +538,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -649,14 +655,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

2 1

[alsa-devel] Why does `aplay -L' not show PCM's defined in ~/.asoundrc ?
by Gabriel M. Beddingfield 02 Dec '13

02 Dec '13

Hi guys, Why does `aplay -L' not list PCM's that are defined in ~/.asoundrc (i.e. PCM's implemented by plugins)? How do you get a listing that includes these? I was helping a user today who needed to set up a device using the `multi' plugin -- and the fact that the new virtual pcm device didn't show up in any standard listing made things very confusing. -gabriel

2 2

[alsa-devel] [PATCH] ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 02 Dec '13

02 Dec '13

The snd_soc_dai_digital_mute() here will be never executed because we only decrease codec->active in snd_soc_close(). Thus correct it. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 22af038..7d19117 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -689,7 +689,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - if (!codec->active) + if (codec->active == 1) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- 1.8.4

3 4

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f43be6d..cba5c4d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -144,6 +144,7 @@ struct fsl_ssi_private { bool imx_ac97; bool use_dma; bool use_dual_fifo; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -325,14 +326,13 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -349,7 +349,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -517,6 +517,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -546,6 +547,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -658,14 +664,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda/analog - Handle inverted EAPD properly in vmaster hook
by Takashi Iwai 02 Dec '13

02 Dec '13

ad_vmaster_eapd_hook() needs to handle the inverted EAPD case properly, too. Otherwise the output gets broken on Lenovo N100 with AD1986A codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 34d86ec5d3dd..f6351b8e0f5d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) if (!spec->eapd_nid) return; + if (codec->inv_eapd) + enabled = !enabled; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); -- 1.8.4.4

1 1