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[alsa-devel] Bad sound quality
by Stanislav Kučera 31 Jan '11

31 Jan '11

Hi, In linux compared to windows the sound quality is much worse. I have tried different sound cards, different distributions during past 3 years and the sound is still not good. In linux the sound is somewhat towards mids, less bass and is overall dirtier compared to windows (xp or 7). Problem is not driver specific as I have tried different sound cards on different computers. I can hear the difference even on laptop speakers. When using jack server sounds is little better, but still no way same as in windows. For me it is major obstacle not using linux as I would love to, because I listen to music so much with quality hifi system. Sound cards which were tested are: onboard VIA, onboard Intel ACL889, Sound Blaster Live, Echo Gina20. Thank. Standa

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[alsa-devel] [Patch] sound: silent echo'ed messages in Makefile
by Amerigo Wang 31 Jan '11

31 Jan '11

Silent these echo's, please. Signed-off-by: WANG Cong <amwang(a)redhat.com> --- diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 96f14dc..90ffb99 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y) $(obj)/bin2hex pss_synth < $< > $@ else $(obj)/pss_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * pss_synth = NULL;'; \ echo 'static int pss_synthLen = 0;'; \ ) > $@ @@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y) $(obj)/hex2hex -i trix_boot < $< > $@ else $(obj)/trix_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * trix_boot = NULL;'; \ echo 'static int trix_boot_len = 0;'; \ ) > $@

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[alsa-devel] snd-via82xx setting
by Benjamin Hovey 31 Jan '11

31 Jan '11

FYI, the below system required the following non-default settings for ALSA to play any SPDIF output correctly. Prior to finding this setting, output from different sources, WAV, MP3, CD, etc, were played at a slightly faster speed than they should have. snd_via82xx: dxs_support=3 When installed, the default setting was dxs_support=5. Hope this helps. I am new to Linux and it took me quite a while to find out how to set this setting (I put it in /etc/modprobe.d/alsa-base.conf) Cheers upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Mon Jan 31 02:37:03 UTC 2011 !!Linux Distribution !!------------------ Ubuntu 10.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 10.04.1 LTS" !!DMI Information !!--------------- Manufacturer: VIA Technologies, Inc. Product Name: KT600-8237 Product Version: !!Kernel Information !!------------------ Kernel release: 2.6.32-28-generic Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.23 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_via82xx snd_emu10k1 !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [V8237 ]: VIA8237 - VIA 8237 VIA 8237 with VT1616i at 0xe400, irq 22 1 [Live ]: EMU10K1 - SB Live! Platinum [CT4760P] SB Live! Platinum [CT4760P] (rev.8, serial:0x80401102) at 0xb000, irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:09.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1 (rev 08) 00:0a.0 Multimedia controller: Sigma Designs, Inc. REALmagic Hollywood Plus DVD Decoder (rev 02) 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 60) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:09.0 0401: 1102:0002 (rev 08) Subsystem: 1102:8040 -- 00:11.5 0401: 1106:3059 (rev 60) Subsystem: 147b:1408 !!Modprobe options (Sound related) !!-------------------------------- snd_via82xx: index=0 snd_emu10k1: index=1 snd_via82xx: dxs_support=4 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd_via82xx: index=0 snd_emu10k1: index=1 snd_via82xx: dxs_support=3 snd_via82xx: ac97_clock=48000

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[alsa-devel] alsamixer don´t start
by kvoigtlaender 31 Jan '11

31 Jan '11

I described my problem here (in german): http://www.vdr-portal.de/board/thread.php?postid=974850#post974850 Mobo Gigabyte M68M-S2P with nVidia chip and ALC883 Tried with headphone on rear and front socket. Error "unknown symbol" in dmesg. Card is not recognized.

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Re: [alsa-devel] Any OSS changes from kernel 2.6.21 to 2.6.23? Something broke.
by Raymond Yau 31 Jan '11

31 Jan '11

> Clemens Ladisch wrote: >> Timur Tabi wrote: >>> It turns out that ALSA (when using mplayer to play a divx video file via OSS >>> emulation) is rapidly sending back-to-back SNDRV_PCM_TRIGGER_STOP and >>> SNDRV_PCM_TRIGGER_START commands. Why would it do that? >> >> To recover from underruns. >I typically get the STOP command just milliseconds before the period ends. ALSA >is pretty impatient! How does ALSA (in OSS emulation mode) know that an >underrun has occurred? >> There shouldn't be any difference, but maybe the buffer size of the >> driver has changed. Could you show the output of mplayer with the "-v" >> option for both kernels? >Unfortunately, I can't seem to get the original kernel to work now, so it's >broken on all kernels. Anywhere, here's the mplayer output on out 2.6.23-based >kernel: >Linux MPC8610HPCD 2.6.23-6 #20 Thu Nov 8 10:02:21 CST 2007 ppc GNU/Linux: ># mplayer -v -ao oss Cars480_1\[1\].5M.divx >MPlayer 1.0rc2-3.3.5 (C) 2000-2007 MPlayer Team >CommandLine: '-v' '-ao' 'oss' 'Cars480_1[1].5M.divx' >========================== >Found 'wf', 30 bytes of 18 >======= WAVE Format ======= >Format Tag: 85 (0x55) >Channels: 2 >Samplerate: 44100 >avg byte/sec: 16000 >Block align: 1 >bits/sample: 0 >cbSize: 12 >mp3.wID=256 >mp3.fdwFlags=0x0 >mp3.nBlockSize=41729 >mp3.nFramesPerBlock=256 >mp3.nCodecDelay=28933 >Opening audio decoder: [mp3lib] MPEG layer-2, layer-3 >dec_audio: Allocating 4608 + 65536 = 70144 bytes for output buffer. >mp3lib: using AltiVec optimized decore! >MP3lib: init layer2&3 finished, tables done >MPEG 1.0, Layer III, 44100 Hz 128 kbit Joint-Stereo, BPF: 418 >Channels: 2, copyright: No, original: No, CRC: No, emphasis: 0 >AUDIO: 44100 Hz, 2 ch, s16be, 128.0 kbit/9.07% (ratio: 16000->176400) >Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3) >========================================================================== >Building audio filter chain for 44100Hz/2ch/s16be -> 0Hz*/0ch/*??... >[libaf] Adding filter dummy >[dummy] Was reinitialized: 44100Hz/2ch/s16be >[dummy] Was reinitialized: 44100Hz/2ch/s16be >ao2: 44100 Hz 2 chans s16be >audio_setup: using '/dev/dsp' dsp device >audio_setup: using '/dev/mixer' mixer device >audio_setup: using 'pcm' mixer device >audio_setup: sample format: s16be (requested: s16be) >audio_setup: using 2 channels (requested: 2) >audio_setup: using 44100 Hz samplerate (requested: 44100) >audio_setup: frags: 2/2 (4104 bytes/frag) free: 8208 >AO: [oss] 44100Hz 2ch s16be (2 bytes per sample) >AO: Description: OSS/ioctl audio output >AO: Author: A'rpi >Building audio filter chain for 44100Hz/2ch/s16be -> 44100Hz/2ch/s16be... >[dummy] Was reinitialized: 44100Hz/2ch/s16be >[dummy] Was reinitialized: 44100Hz/2ch/s16be >Starting playback... I also get shuttering with mplayer SVN-r32829-4.3.2 after mplayer display "Increasing filtered audio buffer size from 65536 to 67584" using oss or alsa with sound card which has max buffer bytes 65536 bytes , but it does not happen with "pulse" or hda driver with max buffer bytes > 65536 bytes dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. audio_setup: frags: 16/16 (4096 bytes/frag) free: 65536 2048 bytes is only half of the frag size 4096 bytes why mplayer increasing filtered audio buffer size by half frag or half period size ? it look like related to R24920 Change decode_audio() interface Rewrite decode_audio to better deal with filters that handle input in large blocks. It now always places output in sh_audio->a_out_buffer (which was always given as a parameter before) and reallocates the buffer if needed. After the changes filters can return arbitrarily large blocks of data without some of it being lost. The new version also allows simplifying some code. mplayer -ao oss stereo.wav ======= WAVE Format ======= Format Tag: 1 (0x1) Channels: 2 Samplerate: 44100 avg byte/sec: 176400 Block align: 4 bits/sample: 16 cbSize: 0 ========================================================================== demux_audio: audio data 0x2C - 0x308002C Audio only file format detected. Load subtitles in ./ get_path('sub/') -> '/home/raymond/.mplayer/sub/' ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'oss', options '[none]' ao2: 44100 Hz 2 chans s16le audio_setup: using '/dev/dsp' dsp device audio_setup: using '/dev/mixer' mixer device audio_setup: using 'pcm' mixer device audio_setup: sample format: s16le (requested: s16le) audio_setup: using 2 channels (requested: 2) audio_setup: using 44100 Hz samplerate (requested: 44100) audio_setup: frags: 16/16 (4096 bytes/frag) free: 65536 AO: [oss] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: OSS/ioctl audio output AO: Author: A'rpi Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 65536 A: 127.7 (02:07.6) of 288.0 (04:48.0) 0.1% Increasing filtered audio buffer size from 65536 to 67584 A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) ds_fill_buffer: EOF reached (stream: audio) A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) A: 288.0 (04:47.9) of 288.0 (04:48.0) 0.1% EOF code: 1 Uninit audio filters... [libaf] Removing filter dummy Uninit audio: pcm mplayer -ao alsa:device=hw=0 stereo.wav ======= WAVE Format ======= Format Tag: 1 (0x1) Channels: 2 Samplerate: 44100 avg byte/sec: 176400 Block align: 4 bits/sample: 16 cbSize: 0 ========================================================================== demux_audio: audio data 0x2C - 0x308002C Audio only file format detected. Load subtitles in ./ get_path('sub/') -> '/home/raymond/.mplayer/sub/' ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options 'device=hw=0' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.23 alsa-init: setup for 1/2 channel(s) alsa-init: using device hw:0 alsa-init: pcm opened in blocking mode alsa-init: got buffersize=65536 alsa-init: got period size 1024 alsa: 44100 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under development Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 65536 A: 3.0 (02.9) of 288.0 (04:48.0) 0.1% Increasing filtered audio buffer size from 65536 to 67584 A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) ds_fill_buffer: EOF reached (stream: audio) A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) A: 288.0 (04:47.9) of 288.0 (04:48.0) 0.1% EOF code: 1 Uninit audio filters... [libaf] Removing filter dummy Uninit audio: pcm alsa-uninit: pcm closed mplayer -ao alsa:device=pulse stereo.wav ======= WAVE Format ======= Format Tag: 1 (0x1) Channels: 2 Samplerate: 44100 avg byte/sec: 176400 Block align: 4 bits/sample: 16 cbSize: 0 ========================================================================== demux_audio: audio data 0x2C - 0x308002C Audio only file format detected. Load subtitles in ./ get_path('sub/') -> '/home/raymond/.mplayer/sub/' ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options 'device=pulse' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.23 alsa-init: setup for 1/2 channel(s) alsa-init: using device pulse alsa-init: pcm opened in blocking mode alsa-init: got buffersize=88200 alsa-init: got period size 1378 alsa: 44100 Hz/2 channels/4 bpf/88200 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under development Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 65536 A: 287.7 (04:47.7) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) ds_fill_buffer: EOF reached (stream: audio) A: 287.7 (04:47.7) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) A: 287.8 (04:47.7) of 288.0 (04:48.0) 0.1% EOF code: 1 Uninit audio filters... [libaf] Removing filter dummy Uninit audio: pcm alsa-uninit: pcm closed

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[alsa-devel] VIA VT1828S: crash in alsamixer, no mic input
by Wolfram Nyaa~ 29 Jan '11

29 Jan '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I've found crash in alsamixer and described it in comment 0023300 in the following issue thread at bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5249 Here is copypaste from it: > Also I've found crash in alsamixer: when I'm trying to unmute "Smart 5.1" control it crashes with message "alsamixer: hcontrol.c:764: snd_hctl_handle_event: Assertion `elem' failed.", but control really unmutes. Muting control doesn't cause crash. > > (gdb) bt > #0 0x00007ffff6d45335 in raise () from /lib/libc.so.6 > #1 0x00007ffff6d465cf in abort () from /lib/libc.so.6 > #2 0x00007ffff6d3e4c1 in __assert_fail () from /lib/libc.so.6 > #3 0x00007ffff72d00d3 in snd_hctl_handle_events () from /usr/lib/libasound.so.2 > #4 0x00007ffff72d98c9 in snd_mixer_handle_events () from /usr/lib/libasound.so.2 > #5 0x0000000000405af1 in ?? () > #6 0x0000000000405096 in ?? () > #7 0x00007ffff6d31d2d in __libc_start_main () from /lib/libc.so.6 > #8 0x0000000000404679 in ?? () > #9 0x00007fffffffd688 in ?? () > #10 0x000000000000001c in ?? () > #11 0x0000000000000001 in ?? () > #12 0x00007fffffffda39 in ?? () > #13 0x0000000000000000 in ?? () Also could anybody give me some hints how to get microphone working with my soundcard (problem is described in thread)? -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.16 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEbBAEBAgAGBQJNQ5rfAAoJEN9tENx5d7sA1gUH9itsA4Hnly22lkei9FRRM22d 0ztVKKFiJq2kz4FAHgvUKr97OlW30b7O5ojbAXU3s6+//w8g//wt4+UsLHdWVKfC RvFIb5sEUh2kdiwqcfAbNqrU27favRmKX9+ZjBXnw4Z6RNGiY8v/StwiuK/uB82h xmC2Bx79Y5jHUnhSSS/kmSMgtiTxFqZrqxYyd/zA/yvth1CLIBgqKrl9oni29Y1n VRFn45NB5DMbNREi4qS17lr37iUZ1j/VB6c9Odd4IWSnTCDn30QPZb8WZJNj2hHn vqHemKdFKnPsMavs3OCjxcEplpoea3kQQjpfXjElPg+O3ZErcGvYqojsSW80fg== =KPqa -----END PGP SIGNATURE-----

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[alsa-devel] Fixed sampling freq UAC2 doubt
by Jassi Brar 29 Jan '11

29 Jan '11

Hi, While testing my under-development generic USB Audio Class 2.0 _gadget_ driver with ALSA's USB Audio Class driver, I came across an apparent discrepency. Or so do I think. Page-98 Section 5.2.5.1.1 of 'Audio 20 final.pdf' specifies :- [In many cases, the Clock Source Entity represents a crystal oscillator based generator with a single fixed frequency. In that case, the Set request is not supported.] Here 'not supported' means setup requests for CUR/RANGE CS_SAM_FREQ_CONTROL should stall. Right ? The snd_usb_hw_params in sound/usb/pcm.c apparently attempts to do that with disregard to Clock-Frequency-Control bitmask in bmControls field of the Clock Source Descriptor. The confusion is, subs->cur_rate is initialized only if snd_usb_init_sample_rate succeeds. Which does SET control tranfer, albeit with the supported frequency. Couldn't a fixed sampling freq UAC2 compliant device refuse to work in that case, thereby failing any attempt to use the USB card? Could someone please point where I got it wrong and/or if such a card works with ALSA USB? Thanks.

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[alsa-devel] [PATCH] sound, hda_eld: suspicious sizeof() in call to snd_print_pcm_bits()
by Jesper Juhl 29 Jan '11

29 Jan '11

In hdmi_show_short_audio_desc() we call snd_print_pcm_bits() like this: snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); that last argument is supposed to be the size of the buffer that snd_print_pcm_bits() writes to, so I believe it should be this: snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) Compile tested only. I don't have relevant hardware. diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 4a66347..74b0560 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) snd_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) - snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); + snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); else if (a->max_bitrate) snprintf(buf2, sizeof(buf2), ", max bitrate = %d", a->max_bitrate); -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/~esr/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] stereo mix of the input source of ad1988 and snd_hda_input_mux_put
by Raymond Yau 29 Jan '11

29 Jan '11

Refer to http://www.analog.com/en/audiovideo-products/audio-codecs/ad1988a/products/… 20 Analog Mixer Audio mixer Mixes individual gain analog inputs 21 Mixer Output Attenuator Audio selector Attenuates the mixer output to drive the port mixers patch_analog.c create "Analog Mix Playback"volume control/switch for node 0x21 when any one of the three [Audio selector] Input source conrol select "mix" ( node 0x20) it seem need to mute the node 0x21 and umute "Line Playback Switch" to get the signal from 0x20 But when using "Line Playback volume" (route the input from. blue jack to green jack ) when you plug the output of other audio device (e.g. TV card ) into blue jack, "Analog Mix Playback Switch" need to be unmute and "Analog Mix Playback Volume" to be set at level selected by user Does this mean that ad1988 may need to implement its a specific snd_hda_input_mux_put() to handle the above requirement ? Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' 'CD' 'Mix' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' 'CD' 'Mix' Item0: 'Front Mic' Simple mixer control 'Input Source',2 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' 'CD' 'Mix' Item0: 'Front Mic' Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=0, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x27 0x27] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=1, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x0e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=2, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=2, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=2, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=6, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=6, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x39 0x33 0x38 0x3d 0x34 0x3b 0x18 0x1a Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Analog Mix Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Analog Mix Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x20

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[alsa-devel] [patch] DMX 6Fire USB - firmware loader update
by Torsten Schenk 28 Jan '11

28 Jan '11

Hello all, I got in contact with a guy for whom the firmware loading didn't work with the current version and I could fix my driver so that firmware loading for all device versions should be supported by now. Here the patch for that. Also I included some instructions how to install the firmware. Thanks, Torsten

1 0

[alsa-devel] [PATCH 3/3] ASoC: sn95031 codec - adding jack detection/reporting
by Harsha Priya 28 Jan '11

28 Jan '11

This patch adds the jack detection logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/sn95031.c | 197 +++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/sn95031.h | 45 ++++++++++ 2 files changed, 232 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 593632c..3fd2289 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -27,6 +27,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/input.h> #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -38,12 +39,130 @@ #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +/* adc helper functions */ + +/* enables mic bias voltage */ +void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2)); +} + +/* Enable/Disable the ADC depending on the argument */ +static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) { + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + value &= (~ADC_NO_LOOP); + } else + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + /* - * todo: - * capture paths - * jack detection - * PM functions + * finds an empty channel for conversion + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * */ +static int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/* Initialize the ADC for reading thermistor values. Can sleep. */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, AUDIO_DETECT_CODE | 0x10); + + chnl_addr = ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/* reads the ADC registers and gets the mic bias value in mV. */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + /* convesion into mV */ + mic_bias = (ADC_ONE_LSB_MULTIPLIER * mic_bias) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); + +/*end - adc helper functions */ + +/* codec read/write functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, unsigned int reg) @@ -69,6 +188,8 @@ static inline int sn95031_write(struct snd_soc_codec *codec, return ret; } +/* end - codec read/write functions */ + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -373,6 +494,62 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +int sn95031_get_headset_state(struct snd_soc_codec *codec) +{ + unsigned int mic_bias = sn95031_get_mic_bias(codec); + int jack_type = snd_soc_jack_get_type(mic_bias); + + if (mic_bias >= SND_SOC_JACK_AM_HS_START && + mic_bias < SND_SOC_JACK_AM_HS_END) + snd_soc_update_bits(codec, AUDIO_GPIO_CTRL, BIT(1), BIT(1)); + else + snd_soc_update_bits(codec, AUDIO_GPIO_CTRL, BIT(1), BIT(0)); + + pr_debug("jack type detected = %d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) + enable_jack_btn(codec); + return jack_type; +} +EXPORT_SYMBOL_GPL(sn95031_get_headset_state); + +void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg) +{ + unsigned int status, mask; + static int jack_state = SND_JACK_HEADSET; + struct snd_soc_codec *codec = jack_msg->codec; + + if (!codec) { + pr_err("codec is null\n"); + return; + } + pr_debug("interrupt id read in sram = 0x%x\n", jack_msg->intr_id); + if (jack_msg->intr_id & 0x1) { + pr_debug("short_push detected\n"); + mask = status = SND_JACK_BTN_0; + } else if (jack_msg->intr_id & 0x2) { + pr_debug("long_push detected\n"); + mask = status = SND_JACK_BTN_1; + } else if (jack_msg->intr_id & 0x4) { + mask = status = sn95031_get_headset_state(codec); + jack_state = mask; + } else if (jack_msg->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + mask = jack_state; + disable_jack_btn(codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(mfld_jack, status, mask); + /*button pressed and released */ + if (status == SND_JACK_BTN_0 || status == SND_JACK_BTN_1) + snd_soc_jack_report(mfld_jack, 0, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); + /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { @@ -443,18 +620,18 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sn95031_codec = { - .probe = sn95031_codec_probe, - .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, +struct snd_soc_codec_driver sn95031_soc_codec = { + .probe = sn95031_codec_probe, + .remove = sn95031_codec_remove, + .read = sn95031_read, + .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, }; static int __devinit sn95031_device_probe(struct platform_device *pdev) { pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); - return snd_soc_register_codec(&pdev->dev, &sn95031_codec, + return snd_soc_register_codec(&pdev->dev, &sn95031_soc_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d9..d808cff 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -1,3 +1,4 @@ + /* * sn95031.h - TI sn95031 Codec driver * @@ -25,6 +26,7 @@ */ #ifndef _SN95031_H #define _SN95031_H +#include <sound/jack.h> /*register map*/ #define SN95031_VAUD 0xDB @@ -96,4 +98,47 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +/* ADC registers */ + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 +#define ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define ADC_LOOP_MAX (ADC_CHANLS_MAX - 1) +#define ADC_NO_LOOP 0x07 + +/* ADC channel code values */ +#define AUDIO_DETECT_CODE 0x06 +#define AUDIO_GPIO_CTRL 0x070 +/* ADC base addresses */ +#define ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ +/* multipier to convert to mV */ +#define ADC_ONE_LSB_MULTIPLIER 2346 + +struct mfld_jack_msg_wq { + u8 intr_id; + struct snd_soc_codec *codec; + struct work_struct wq; +}; + +static inline void disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +extern void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg); + #endif -- 1.7.3.4

1 0

[alsa-devel] [PATCH 2/3] ASoC: mid-x86 machine driver - add jack detect support
by Harsha Priya 28 Jan '11

28 Jan '11

This patch adds the jack detection in mid-86 machine driver Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 175 ++++++++++++++++++++++++++++++++------ 1 files changed, 148 insertions(+), 27 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be..d708da8 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -27,6 +27,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -35,10 +36,33 @@ #define MID_MONO 1 #define MID_STEREO 2 #define MID_MAX_CAP 5 +#define IRQ_BASE 0xFFFF7FCD +#define IRQ_SIZE 1 static unsigned int hs_switch; static unsigned int lo_dac; +struct mfld_mc_private { + struct platform_device *socdev; + struct workqueue_struct *mfld_jack_wq; + void __iomem *int_base; + struct snd_soc_codec *codec; +} mc_drv_ctx; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -67,13 +91,11 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) { pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } snd_soc_dapm_sync(&codec->dapm); @@ -91,12 +113,10 @@ static void lo_enable_out_pins(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } } @@ -130,8 +150,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, case 1: pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); break; @@ -162,12 +181,36 @@ static const struct snd_kcontrol_new mfld_snd_controls[] = { lo_get_switch, lo_set_switch), }; +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + static int mfld_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; int ret_val; + /* Add jack sense widgets */ + snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); + + /* Set up the map */ + snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + snd_soc_dapm_sync(dapm); + + mc_drv_ctx.codec = codec; + ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); if (ret_val) { @@ -175,14 +218,32 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; - return snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + + /* Headset jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + disable_jack_btn(codec); + return ret_val; } struct snd_soc_dai_link mfld_msic_dailink[] = { @@ -231,37 +292,97 @@ static struct snd_soc_card snd_soc_card_mfld = { .num_links = ARRAY_SIZE(mfld_msic_dailink), }; +void snd_mfld_jack_detection(struct work_struct *work) +{ + struct mfld_jack_msg_wq *jack_msg = + container_of(work, struct mfld_jack_msg_wq, wq); + + sn95031_jack_detection(&mfld_jack, jack_msg); + kfree(jack_msg); +} + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_jack_msg_wq *jack_msg; + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + jack_msg = kzalloc(sizeof(*jack_msg), GFP_ATOMIC); + memcpy_fromio(&jack_msg->intr_id, + ((void *)(mc_private->int_base)), + sizeof(u8)); + jack_msg->codec = mc_private->codec; + pr_debug("snd_mfld_jack_intr_handler called value %x\n", + jack_msg->intr_id); + INIT_WORK(&jack_msg->wq, snd_mfld_jack_detection); + queue_work(mc_private->mfld_jack_wq, &jack_msg->wq); + + return IRQ_HANDLED; +} + static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) { - struct platform_device *socdev; - int ret_val = 0; + int ret_val = 0, irq; pr_debug("snd_mfld_mc_probe called\n"); - socdev = platform_device_alloc("soc-audio", -1); - if (!socdev) { + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx.int_base = ioremap_nocache(IRQ_BASE, IRQ_SIZE); + if (!mc_drv_ctx.int_base) { + pr_err("Mapping of cache failed\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* create botoom half handler for interrupt */ + mc_drv_ctx.mfld_jack_wq = create_workqueue("mfld_jack_wq"); + if (!mc_drv_ctx.mfld_jack_wq) { + pr_err("cannot create jack workqueue\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* register for interrupt */ + ret_val = request_irq(irq, snd_mfld_jack_intr_handler, + IRQF_SHARED, pdev->dev.driver->name, + &mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + goto destroy_wq; + } + /* create soc device */ + mc_drv_ctx.socdev = platform_device_alloc("soc-audio", -1); + if (!mc_drv_ctx.socdev) { pr_err("soc-audio device allocation failed\n"); - return -ENOMEM; + ret_val = -ENOMEM; + goto destroy_wq; } - platform_set_drvdata(socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(socdev); + platform_set_drvdata(mc_drv_ctx.socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(mc_drv_ctx.socdev); if (ret_val) { pr_err("Unable to add soc-audio device, err %d\n", ret_val); - platform_device_put(socdev); + goto unregister; } - - platform_set_drvdata(pdev, socdev); - + platform_set_drvdata(pdev, &mc_drv_ctx); pr_debug("successfully exited probe\n"); return ret_val; + +unregister: + platform_device_put(mc_drv_ctx.socdev); +destroy_wq: + destroy_workqueue(mc_drv_ctx.mfld_jack_wq); +unalloc: + return ret_val; } static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) { - struct platform_device *socdev = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); + struct mfld_mc_private *mc_private_data = platform_get_drvdata(pdev); - platform_device_unregister(socdev); + pr_debug("snd_mfld_mc_remove called\n"); + free_irq(platform_get_irq(pdev, 0), mc_private_data); + destroy_workqueue(mc_private_data->mfld_jack_wq); + platform_device_unregister(mc_private_data->socdev); platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.3.4

1 0

[alsa-devel] [PATCH 3/3] ASoC: mid-x86: Add support for capture in machine driver
by Harsha Priya 28 Jan '11

28 Jan '11

This configures the capture unused pins Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be..7925851 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -182,6 +182,9 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); return snd_soc_dapm_sync(dapm); } -- 1.7.3.4

1 0

[alsa-devel] [PATCH 2/3] ASoC: sst_platform: add support for capture stream on headset dai
by Harsha Priya 28 Jan '11

28 Jan '11

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/sst_platform.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index a4e3fa3..0d8c430 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -73,6 +73,12 @@ struct snd_soc_dai_driver sst_platform_dai[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, }, { .name = "Speaker-cpu-dai", -- 1.7.2.3

1 0

[alsa-devel] [PATCH 1/3] ASoC: sn95031: add capture support
by Harsha Priya 28 Jan '11

28 Jan '11

This patch adds the support for capture path in sn95031 codec. This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs are connected through a MUX to ADC. The TX paths can be assigned to any of the ADCs or DMICs. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/codecs/sn95031.c | 281 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 281 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 593632c..40e285d 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -33,6 +33,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -145,6 +146,129 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, return 0; } +static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk_dir = 0, data_dir = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ldo = BIT(5)|BIT(4); + clk_dir = BIT(0); + data_dir = BIT(7); + } + /* program DMIC LDO, clock and set clock */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir); + return 0; +} + +static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk_dir = 0, data_dir = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ldo = BIT(5)|BIT(4); + clk_dir = BIT(2); + data_dir = BIT(1); + } + /* program DMIC LDO, clock and set clock */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir); + snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir); + return 0; +} + +static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ldo = BIT(7)|BIT(6); + + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); + return 0; +} + +/* mux controls */ +static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; + +static const struct soc_enum sn95031_micl_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micl_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micl_enum); + +static const struct soc_enum sn95031_micr_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micr_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micr_enum); + +static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", + "DMIC4", "DMIC5", "DMIC6", + "ADC Left", "ADC Right" }; + +static const struct soc_enum sn95031_input1_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input1_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input1_enum); + +static const struct soc_enum sn95031_input2_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input2_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input2_enum); + +static const struct soc_enum sn95031_input3_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input3_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input3_enum); + +static const struct soc_enum sn95031_input4_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input4_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input4_enum); + +/* capture path controls */ + +static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; + +/* 0dB to 30dB in 10dB steps */ +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); + +static const struct soc_enum sn95031_micmode1_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); +static const struct soc_enum sn95031_micmode2_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); + +static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; + +static const struct soc_enum sn95031_dmic12_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic34_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic56_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); + +static const struct snd_kcontrol_new sn95031_snd_controls[] = { + SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), + SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum), + SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum), + SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum), + SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum), + SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1, + 2, 4, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2, + 2, 4, 0, mic_tlv), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { @@ -159,6 +283,36 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIB1OUT"), SND_SOC_DAPM_OUTPUT("VIB2OUT"), + SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */ + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + SND_SOC_DAPM_INPUT("DMIC5"), + SND_SOC_DAPM_INPUT("DMIC6"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + + SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0), + SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0), + SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0), + SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0), + SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0), + + SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0, + sn95031_dmic12_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0, + sn95031_dmic34_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0, + sn95031_dmic56_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, sn95031_vhs_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -209,6 +363,40 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SN95031_VIB1C5, 1, 0), SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", SN95031_VIB2C5, 1, 0), + + /* capture widgets */ + SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2, + 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0), + + /* ADC have null stream as they will be turned ON by TX path */ + SND_SOC_DAPM_ADC("ADC Left", NULL, + SN95031_ADCCONFIG, 0, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, + SN95031_ADCCONFIG, 2, 0), + + SND_SOC_DAPM_MUX("Mic_InputL Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control), + SND_SOC_DAPM_MUX("Mic_InputR Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control), + + SND_SOC_DAPM_MUX("Txpath1 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control), + SND_SOC_DAPM_MUX("Txpath2 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control), + SND_SOC_DAPM_MUX("Txpath3 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control), + SND_SOC_DAPM_MUX("Txpath4 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control), + }; static const struct snd_soc_dapm_route sn95031_audio_map[] = { @@ -250,6 +438,87 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Lineout Right Playback", NULL, "Headset Right Filter"}, { "Lineout Right Playback", NULL, "Speaker Right Filter"}, { "Lineout Right Playback", NULL, "Vibra2 DAC"}, + + /* Headset (AMIC1) mic */ + { "AMIC1Bias", NULL, "AMIC1"}, + { "MIC1 Enable", NULL, "AMIC1Bias"}, + { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"}, + + /* AMIC2 */ + { "AMIC2Bias", NULL, "AMIC2"}, + { "MIC2 Enable", NULL, "AMIC2Bias"}, + { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"}, + + + /* Linein */ + { "LineIn Enable Left", NULL, "LINEINL"}, + { "LineIn Enable Right", NULL, "LINEINR"}, + { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"}, + { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"}, + + /* ADC connection */ + { "ADC Left", NULL, "Mic_InputL Capture Route"}, + { "ADC Right", NULL, "Mic_InputR Capture Route"}, + + /*DMIC connections */ + { "DMIC1", NULL, "DMIC12supply"}, + { "DMIC2", NULL, "DMIC12supply"}, + { "DMIC3", NULL, "DMIC34supply"}, + { "DMIC4", NULL, "DMIC34supply"}, + { "DMIC5", NULL, "DMIC56supply"}, + { "DMIC6", NULL, "DMIC56supply"}, + + { "DMIC12Bias", NULL, "DMIC1"}, + { "DMIC12Bias", NULL, "DMIC2"}, + { "DMIC34Bias", NULL, "DMIC3"}, + { "DMIC34Bias", NULL, "DMIC4"}, + { "DMIC56Bias", NULL, "DMIC5"}, + { "DMIC56Bias", NULL, "DMIC6"}, + + /*TX path inputs*/ + { "Txpath1 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath2 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath3 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath4 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath1 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath1 Capture Route", NULL, "DMIC1"}, + { "Txpath2 Capture Route", NULL, "DMIC1"}, + { "Txpath3 Capture Route", NULL, "DMIC1"}, + { "Txpath4 Capture Route", NULL, "DMIC1"}, + { "Txpath1 Capture Route", NULL, "DMIC2"}, + { "Txpath2 Capture Route", NULL, "DMIC2"}, + { "Txpath3 Capture Route", NULL, "DMIC2"}, + { "Txpath4 Capture Route", NULL, "DMIC2"}, + { "Txpath1 Capture Route", NULL, "DMIC3"}, + { "Txpath2 Capture Route", NULL, "DMIC3"}, + { "Txpath3 Capture Route", NULL, "DMIC3"}, + { "Txpath4 Capture Route", NULL, "DMIC3"}, + { "Txpath1 Capture Route", NULL, "DMIC4"}, + { "Txpath2 Capture Route", NULL, "DMIC4"}, + { "Txpath3 Capture Route", NULL, "DMIC4"}, + { "Txpath4 Capture Route", NULL, "DMIC4"}, + { "Txpath1 Capture Route", NULL, "DMIC5"}, + { "Txpath2 Capture Route", NULL, "DMIC5"}, + { "Txpath3 Capture Route", NULL, "DMIC5"}, + { "Txpath4 Capture Route", NULL, "DMIC5"}, + { "Txpath1 Capture Route", NULL, "DMIC6"}, + { "Txpath2 Capture Route", NULL, "DMIC6"}, + { "Txpath3 Capture Route", NULL, "DMIC6"}, + { "Txpath4 Capture Route", NULL, "DMIC6"}, + + /* tx path */ + { "TX1 Enable", NULL, "Txpath1 Capture Route"}, + { "TX2 Enable", NULL, "Txpath2 Capture Route"}, + { "TX3 Enable", NULL, "Txpath3 Capture Route"}, + { "TX4 Enable", NULL, "Txpath4 Capture Route"}, + { "PCM_Out", NULL, "TX1 Enable"}, + { "PCM_Out", NULL, "TX2 Enable"}, + { "PCM_Out", NULL, "TX3 Enable"}, + { "PCM_Out", NULL, "TX4 Enable"}, + }; /* speaker and headset mutes, for audio pops and clicks */ @@ -339,6 +608,13 @@ struct snd_soc_dai_driver sn95031_dais[] = { .rates = SN95031_RATES, .formats = SN95031_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, .ops = &sn95031_headset_dai_ops, }, { .name = "SN95031 Speaker", @@ -390,6 +666,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32); /* pcm port setting * This sets the pcm port to slave and clock at 19.2Mhz which * can support 6slots, sampling rate set per stream in hw-params @@ -423,6 +701,9 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); + snd_soc_add_controls(codec, sn95031_snd_controls, + ARRAY_SIZE(sn95031_snd_controls)); + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, ARRAY_SIZE(sn95031_dapm_widgets)); if (ret) -- 1.7.3.4

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[alsa-devel] [PATCH 0/3] ASoC: patch series to add support for jack detection in mid-x86 and sn95031 codec
by Harsha, Priya 28 Jan '11

28 Jan '11

Hi Mark, This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. Also adds a helper function in soc-jack.c for converting mic bias values to jack types. The comments from previous submission are addressed. [PATCH 1/3] ASoC: soc_jack - add function to determine jack type This patch adds a helper function that determines the jack type given the mic bias value. For any codec that would give the mic bias value on a jack insertion, this function would return the type of jack based on mic bias range [PATCH 2/3] ASoC: mid-x86 machine driver - add jack detect support This patch adds the jack detection in mid-86 machine driver [PATCH 3/3] ASoC: sn95031 codec - adding jack detection/reporting This patch adds the jack detection logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected -Harsha

1 0

[alsa-devel] [PATCH] ASoC: Fix typo in snd_soc_dapm_put_volsw
by Stephen Warren 28 Jan '11

28 Jan '11

This fixes a regression introduced by: 97404f2e0386ac147cec00fc5d89ea475b04bd78 ASoC: Do DAPM control updates in the middle of DAPM sequences Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0f94fd0..d0342aa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1779,7 +1779,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val_mask; + unsigned int val; int connect, change; struct snd_soc_dapm_update update; @@ -1787,13 +1787,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - val_mask = mask << shift; + mask = mask << shift; val = val << shift; mutex_lock(&widget->codec->mutex); widget->value = val; - change = snd_soc_test_bits(widget->codec, reg, val_mask, val); + change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { if (val) /* new connection */ -- 1.7.1

3 4

[alsa-devel] [PATCH] ASoC: Add card driver data
by Mark Brown 28 Jan '11

28 Jan '11

Provide driver data for cards within the card structure. To simplify the implementation of the PM operations we don't use the struct device driver data as this is used by the core to retrieve the card in callbacks from the device model and PM core. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ecdaef..4b6c0a8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -705,6 +705,8 @@ struct snd_soc_card { struct dentry *debugfs_pop_time; #endif u32 pop_time; + + void *drvdata; }; /* SoC machine DAI configuration, glues a codec and cpu DAI together */ @@ -756,6 +758,17 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec, /* device driver data */ +static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, + void *data) +{ + card->drvdata = data; +} + +static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card) +{ + return card->drvdata; +} + static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, void *data) { -- 1.7.2.3

3 2

[alsa-devel] ASoC: Configuring codec PCM port question
by Babu, Ramesh 28 Jan '11

28 Jan '11

Hi My codec has 2 ports, PCM1 and PCM2 ports. PCM2 port is used for playback and capture operations and PCM1 is connected to Modem. The ASoC driver has to configure, enable and disable PCM1 port during voice call activity. I am thinking of following two options to configure PCM1 port: 1. AlSA Mixer control: Create mixer control and provide options like VOICE_HS, VOICE_IHF, VOICE_EP and VOICE_DISABLE. Application will use one of the option to Configure the PCM1 port. 2. IOCTL : Application invokes IOCTL in hardware specific driver code and then it enables PCM1 port (cross-path DAPM ?) Which is the best way to implement it? Are there any other existing way to implement it? Thanks Ramesh

2 1

[alsa-devel] ALSA mixer query
by Irfan Shaikh 28 Jan '11

28 Jan '11

Hello all, I am rendering audio to a PCM playback device (hw:0,0). I also need to render the same audio data to second PCM playback device (hw:0,1). 1) Can i use ALSA mixer APIs so that i give PCM audio data to 1 playback device or some common interface, but the PCM audio data is rendered to both devices (hw:0,0 and hw:0,1) Regards Irfan SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 1

[alsa-devel] [PATCH 0/2] ASoC: Refcount fixes
by Jarkko Nikula 28 Jan '11

28 Jan '11

Hi Nothing new, just yesterday's refcount fix for-2.6.38 and another minor for-2.6.39 rebased on top of for-2.6.39 and sent together. -- Jarkko

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[alsa-devel] Strange control interactions on Tegra Harmony
by Stephen Warren 28 Jan '11

28 Jan '11

Mark, I performed some testing on the patch to enable Harmony's internal speaker that I recently posted, and found a few strange interactions between the mixer controls. The relevant controls that appear are in alsamixer are: Headphone (mute + volume) [default: unmuted, high volume] Speaker (mute + volume) [default: unmuted, high volume] Left/Right Speaker Mixer DACL/DACR/Left Bypass/Right Bypass (switch, default muted/off) Now, if I start playing something, and go enable Right Speaker Mixer DACR, I hear the sound on the speaker. Muting this control turns off the speaker. However, unmuting again doesn't turn the speaker back on. This exact same behavior is seen for Right Speaker Mixer DACL/Left Bypass. For Left Bypass at least, that's odd, since the ADC path shouldn't be active to Capture anything and bypass it back, and there's no mic attached. The Right Speaker Right Bypass control works the first and all subsequent times. Perhaps that's something to do with it being the last entry in wm8903.c's right_speaker_mixer[] array? Finally, the muting either the Headphone and Speaker controls actually affects both the headphone and speaker outputs in all cases, rather than just the output the control is named for. I briefly started to debug this, and notice that the SND_SOC_DAPM_SPK event function isn't being called at all in the cases where muting doesn't toggle as expected. I didn't manage to trace the code through the entire ASoc code-base yet, but initial indications were that some register writes for the speaker mixer routing setup weren't sticking - i.e. I'd toggle a control to unmuted, exit alsamixer, restart alsamixer, and the control wouldn't be in the state I'd placed it. Can you comment on this? I can certainly continue to debug it, but I figured it might be immediately obvious to you what the problem was e.g. if the same thing had happened on other platforms/codecs. Thanks. -- nvpublic

2 4

[alsa-devel] [PATCH] ASoC: Use card as parameter to exported PM functions
by Stephen Warren 28 Jan '11

28 Jan '11

If a machine driver is going to use the exported snd_soc_suspend/resume, at present, the platform device's drvdata must be the snd_soc_card. This prevents the machine driver from placing its own custom drvdata into the platform device. This patch reworks the exported PM functions to take the card instead of the dev, thus allowing machine drivers to use a custom drvdata type. The disadvantage is that every machine driver that does this must then implement something similar to the soc_suspend/resume_dev functions in this patch. An alternative might be to add a drvdata field to struct snd_soc_card, accompanied by snd_soc_set/get_drvdata functions. Another alternative would be to wrap the snd_soc_card stored in the drvdata within another type, and use container_of whenever the machine driver needed the data in the wrapper type. This patch is hence more of a question of "do we want to do it this way", rather than a concrete proposed patch. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- include/sound/soc.h | 6 +++--- sound/soc/soc-core.c | 46 ++++++++++++++++++++++++++++++++-------------- 2 files changed, 35 insertions(+), 17 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ecdaef..96ae31d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -260,9 +260,9 @@ enum snd_soc_compress_type { int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); -int snd_soc_suspend(struct device *dev); -int snd_soc_resume(struct device *dev); -int snd_soc_poweroff(struct device *dev); +int snd_soc_suspend(struct snd_soc_card *card); +int snd_soc_resume(struct snd_soc_card *card); +int snd_soc_poweroff(struct snd_soc_card *card); int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e9f81c5..b72110c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -972,9 +972,8 @@ static struct snd_pcm_ops soc_pcm_ops = { #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ -int snd_soc_suspend(struct device *dev) +int snd_soc_suspend(struct snd_soc_card *card) { - struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; int i; @@ -1088,6 +1087,13 @@ int snd_soc_suspend(struct device *dev) } EXPORT_SYMBOL_GPL(snd_soc_suspend); +static int soc_suspend_dev(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + return snd_soc_suspend(card); +} + /* deferred resume work, so resume can complete before we finished * setting our codec back up, which can be very slow on I2C */ @@ -1192,9 +1198,8 @@ static void soc_resume_deferred(struct work_struct *work) } /* powers up audio subsystem after a suspend */ -int snd_soc_resume(struct device *dev) +int snd_soc_resume(struct snd_soc_card *card) { - struct snd_soc_card *card = dev_get_drvdata(dev); int i; /* AC97 devices might have other drivers hanging off them so @@ -1205,21 +1210,28 @@ int snd_soc_resume(struct device *dev) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; if (cpu_dai->driver->ac97_control) { - dev_dbg(dev, "Resuming AC97 immediately\n"); + dev_dbg(card->dev, "Resuming AC97 immediately\n"); soc_resume_deferred(&card->deferred_resume_work); } else { - dev_dbg(dev, "Scheduling resume work\n"); + dev_dbg(card->dev, "Scheduling resume work\n"); if (!schedule_work(&card->deferred_resume_work)) - dev_err(dev, "resume work item may be lost\n"); + dev_err(card->dev, "resume work item may be lost\n"); } } return 0; } EXPORT_SYMBOL_GPL(snd_soc_resume); + +static int soc_resume_dev(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + return snd_soc_resume(card); +} #else -#define snd_soc_suspend NULL -#define snd_soc_resume NULL +#define soc_suspend_dev NULL +#define soc_resume_dev NULL #endif static struct snd_soc_dai_ops null_dai_ops = { @@ -1926,9 +1938,8 @@ static int soc_remove(struct platform_device *pdev) return 0; } -int snd_soc_poweroff(struct device *dev) +int snd_soc_poweroff(struct snd_soc_card *card) { - struct snd_soc_card *card = dev_get_drvdata(dev); int i; if (!card->instantiated) @@ -1947,10 +1958,17 @@ int snd_soc_poweroff(struct device *dev) } EXPORT_SYMBOL_GPL(snd_soc_poweroff); +static int soc_poweroff_dev(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + return snd_soc_poweroff(card); +} + const struct dev_pm_ops snd_soc_pm_ops = { - .suspend = snd_soc_suspend, - .resume = snd_soc_resume, - .poweroff = snd_soc_poweroff, + .suspend = soc_suspend_dev, + .resume = soc_resume_dev, + .poweroff = soc_poweroff_dev, }; /* ASoC platform driver */ -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: fix kernel panic due to uninitialized platform_data
by Manjunathappa, Prakash 28 Jan '11

28 Jan '11

This patch fixes the Kernel panic issue on accessing davinci_vc in cq93vc_probe function. struct davinci_vc is part of platform device's private driver data(codec->dev->p->driver_data) and this is populated by DaVinci Voice Codec MFD driver. Signed-off-by: Manjunathappa, Prakash <prakash.pm(a)ti.com> --- This patch applies to Linus's tree at [1]. [1] http://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git --- sound/soc/codecs/cq93vc.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd0..347a567 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; -- 1.7.1

3 2

[alsa-devel] configure sampling rate using alsa-utils
by subashchandra.bose＠wipro.com 28 Jan '11

28 Jan '11

Hi All, we are planning to write a driver for GenADC device in ALSA framework. The hardware device is capable of working in different sampling rate for voice channel, High Speed Channel and low Speed Channel. I went into ALSA home page, using alsa-utils aplay can be used to configure the sampling rate(eg: aplay -r SNDRV_PCM_RATE_8000) Can anybody config if this can be used to configure the GenADC device for voice/HighSpeed/Low Speed channels? Thanks & Regards, -Subash Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

Re: [alsa-devel] [PATCH - [hdspm] RayDAT/AIO 1/2] ALSA: Add support for RME RayDAT and AIO
by Takashi Iwai 27 Jan '11

27 Jan '11

At Wed, 26 Jan 2011 19:32:14 +0100, Adrian Knoth wrote: > > diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h > index 81990b2..c3f1819 100644 > --- a/include/sound/hdspm.h > +++ b/include/sound/hdspm.h > @@ -131,4 +225,175 @@ typedef struct hdspm_version hdspm_version_t; > typedef struct hdspm_channelfader snd_hdspm_channelfader_t; > typedef struct hdspm_mixer hdspm_mixer_t; > > -#endif /* __SOUND_HDSPM_H */ > +/* These tables map the ALSA channels 1..N to the channels that we > + need to use in order to find the relevant channel buffer. RME > + refers to this kind of mapping as between "the ADAT channel and > + the DMA channel." We index it using the logical audio channel, > + and the value is the DMA channel (i.e. channel buffer number) > + where the data for that channel can be read/written from/to. > +*/ > + > +char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = { Err, no, array definitions should be never in a header file. It doesn't matter whether it's static or not. The definition should be in *.c file where it's used. So, simply move these array definitions to hdspm.c. thanks, Takashi

4 8

[alsa-devel] [PATCH] ASoC: Add subsequence information to seq_notify callbacks
by Mark Brown 27 Jan '11

27 Jan '11

Allows drivers to distinguish which subsequence is being notified when they get called back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- sound/soc/soc-dapm.c | 5 +++-- 3 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6a25e69..979ed84 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -501,7 +501,7 @@ struct snd_soc_dapm_context { struct snd_soc_dapm_update *update; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type); + enum snd_soc_dapm_type, int); struct device *dev; /* from parent - for debug */ struct snd_soc_codec *codec; /* parent codec */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 49a14da..ab0d5cd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -554,7 +554,7 @@ struct snd_soc_codec_driver { enum snd_soc_bias_level level); void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type); + enum snd_soc_dapm_type, int); }; /* SoC platform interface */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 37b376f..0f94fd0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -899,7 +899,8 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i); + i, + cur_subseq); } INIT_LIST_HEAD(&pending); @@ -968,7 +969,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i); + i, cur_subseq); } } -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Make cache status available via debugfs
by Mark Brown 27 Jan '11

27 Jan '11

Could just as well live in sysfs but sysfs doesn't have the simple value export helpers debugfs does. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 5 +++++ 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4fd79e0..49a14da 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -494,14 +494,14 @@ struct snd_soc_codec { struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; unsigned int cache_bypass:1; /* Suppress access to the cache */ - unsigned int cache_only:1; /* Suppress writes to hardware */ - unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ unsigned int suspended:1; /* Codec is in suspend PM state */ unsigned int probed:1; /* Codec has been probed */ unsigned int ac97_registered:1; /* Codec has been AC97 registered */ unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int sysfs_registered:1; /* codec has been sysfs registered */ unsigned int cache_init:1; /* codec cache has been initialized */ + u32 cache_only; /* Suppress writes to hardware */ + u32 cache_sync; /* Cache needs to be synced to hardware */ /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5dffc7a4..8af47f7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -235,6 +235,11 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) return; } + debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + &codec->cache_sync); + debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, + &codec->cache_only); + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->debugfs_codec_root, codec, &codec_reg_fops); -- 1.7.2.3

2 1

[alsa-devel] [PATCH - [hdspmixer] 1/1] hdspmixer: Don't use channelmap from hdspm.h
by Adrian Knoth 27 Jan '11

27 Jan '11

The channel mapping has been moved to hdspm.c, so it's no longer available to userspace tools. For now, let's simply copy (duplicate) the data and wait for a way to query this information from the driver. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/hdspmixer/src/HDSPMixerCard.h b/hdspmixer/src/HDSPMixerCard.h index 9303d21..cfb2402 100644 --- a/hdspmixer/src/HDSPMixerCard.h +++ b/hdspmixer/src/HDSPMixerCard.h @@ -28,6 +28,7 @@ #include <alsa/sound/hdsp.h> #include <alsa/sound/hdspm.h> #include "defines.h" +#include "channelmap.h" #include "HDSPMixerWindow.h" class HDSPMixerWindow; diff --git a/hdspmixer/src/channelmap.h b/hdspmixer/src/channelmap.h new file mode 100644 index 0000000..de89140 --- /dev/null +++ b/hdspmixer/src/channelmap.h @@ -0,0 +1,167 @@ +#ifndef channelmap_H +#define channelmap_H + +static char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, 2, 3, 4, 5, 6, 7, + 8, 9, 10, 11, 12, 13, 14, 15, + 16, 17, 18, 19, 20, 21, 22, 23, + 24, 25, 26, 27, 28, 29, 30, 31, + 32, 33, 34, 35, 36, 37, 38, 39, + 40, 41, 42, 43, 44, 45, 46, 47, + 48, 49, 50, 51, 52, 53, 54, 55, + 56, 57, 58, 59, 60, 61, 62, 63 +}; + +static char channel_map_unity_ds[HDSPM_MAX_CHANNELS] = { + 0, 2, 4, 6, 8, 10, 12, 14, + 16, 18, 20, 22, 24, 26, 28, 30, + 32, 34, 36, 38, 40, 42, 44, 46, + 48, 50, 52, 54, 56, 58, 60, 62, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_unity_qs[HDSPM_MAX_CHANNELS] = { + 0, 4, 8, 12, 16, 20, 24, 28, + 32, 36, 40, 44, 48, 52, 56, 60, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_raydat_ss[HDSPM_MAX_CHANNELS] = { + 4, 5, 6, 7, 8, 9, 10, 11, /* ADAT 1 */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT 2 */ + 20, 21, 22, 23, 24, 25, 26, 27, /* ADAT 3 */ + 28, 29, 30, 31, 32, 33, 34, 35, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_raydat_ds[HDSPM_MAX_CHANNELS] = { + 4, 5, 6, 7, /* ADAT 1 */ + 8, 9, 10, 11, /* ADAT 2 */ + 12, 13, 14, 15, /* ADAT 3 */ + 16, 17, 18, 19, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_raydat_qs[HDSPM_MAX_CHANNELS] = { + 4, 5, /* ADAT 1 */ + 6, 7, /* ADAT 2 */ + 8, 9, /* ADAT 3 */ + 10, 11, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in, */ + 10, 11, /* spdif in */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ + -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ + 6, 7, /* phone out */ + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 14, 16, 18, /* adat in */ + -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 14, 16, 18, /* adat out */ + 6, 7, /* phone out */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 16, /* adat in */ + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 16, /* adat out */ + 6, 7, /* phone out */ + -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +#endif /* channelmap_H */ -- 1.7.2.3

2 1

[alsa-devel] [PATCH 1/4] ASoC: soc-core: Increment codec and platform driver refcounts before probing
by Jarkko Nikula 27 Jan '11

27 Jan '11

Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec and platform driver refcount increments from soc_bind_dai_link to more appropriate places. Adjust a little them so that refcounts are incremented before executing the driver probe functions. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Harsha Priya <priya.harsha(a)intel.com> --- I don't know any regression with this so probably 2.6.39 material. --- sound/soc/soc-core.c | 21 +++++++++++++-------- 1 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 14861f9..4da185f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1408,26 +1408,31 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->dapm.card = card; soc_set_name_prefix(card, codec); + if (!try_module_get(codec->dev->driver->owner)) + return -ENODEV; + if (codec->driver->probe) { ret = codec->driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", codec->name, ret); - return ret; + goto err_probe; } } soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); list_add(&codec->dapm.list, &card->dapm_list); + return 0; + +err_probe: + module_put(codec->dev->driver->owner); + return ret; } @@ -1545,19 +1550,19 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the platform */ if (!platform->probed) { + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + if (platform->driver->probe) { ret = platform->driver->probe(platform); if (ret < 0) { printk(KERN_ERR "asoc: failed to probe platform %s\n", platform->name); + module_put(platform->dev->driver->owner); return ret; } } /* mark platform as probed and add to card platform list */ - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - platform->probed = 1; list_add(&platform->card_list, &card->platform_dev_list); } -- 1.7.2.3

3 11

[alsa-devel] Triggering a DAPM widget event
by Harrod, John 27 Jan '11

27 Jan '11

Hi all, I have a program that uses the ALSA PCM interface to play sounds on an embedded system. Does anyone know how alsa-lib triggers a SND_SOC_DAPM_PRE_PMD widget event in DAPM core? On my system the event only gets called when the pcm is closed with snd_pcm_close(). Is there another function that can cause a widget shutdown? Or is opening and closing the PCM every time I want to play a sound a good practice for saving power? Regards, John ________________________________ The information contained in this message may be confidential and legally protected under applicable law. The message is intended solely for the addressee(s). If you are not the intended recipient, you are hereby notified that any use, forwarding, dissemination, or reproduction of this message is strictly prohibited and may be unlawful. If you are not the intended recipient, please contact the sender by return e-mail and destroy all copies of the original message.

3 7

[PATCH] ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
by David Henningsson 27 Jan '11

27 Jan '11

BugLink: http://bugs.launchpad.net/bugs/708521 This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9867afc..7e1ca43 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3120,6 +3120,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ -- 1.7.1 --------------020306060702000000060003--

1 0

[alsa-devel] [PATCH 2/2] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
by David Henningsson 27 Jan '11

27 Jan '11

BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 8

[alsa-devel] [PATCH 2.6.38] ASoC: Fix module refcount for auxiliary devices
by Jarkko Nikula 26 Jan '11

26 Jan '11

Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec driver refcount increments from soc_bind_dai_link into soc_probe_codec. However, the commit didn't remove try_module_get from soc_probe_aux_dev so the auxiliary device reference counts are incremented twice as the soc_probe_codec is called from soc_probe_aux_dev too. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/soc-core.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291..c4b6061 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) goto out; found: - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - ret = soc_probe_codec(card, codec); if (ret < 0) return ret; -- 1.7.2.3

3 2

[alsa-devel] [PATCH 1/3] ASoC: Use card rather than soc-audio device to card PM functions
by Mark Brown 26 Jan '11

26 Jan '11

The platform device for the card is tied closely to the soc-audio implementation which we're currently trying to remove in favour of allowing cards to have their own devices. Begin removing it by replacing it with the card in the suspend and resume callbacks we give to cards, also taking the opportunity to remove the legacy suspend types which are currently hard coded anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 8 ++++---- sound/soc/pxa/raumfeld.c | 4 ++-- sound/soc/pxa/zylonite.c | 5 ++--- sound/soc/soc-core.c | 9 ++++----- 4 files changed, 12 insertions(+), 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 00c6fba..d1cc1a2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -655,10 +655,10 @@ struct snd_soc_card { /* the pre and post PM functions are used to do any PM work before and * after the codec and DAI's do any PM work. */ - int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); - int (*suspend_post)(struct platform_device *pdev, pm_message_t state); - int (*resume_pre)(struct platform_device *pdev); - int (*resume_post)(struct platform_device *pdev); + int (*suspend_pre)(struct snd_soc_card *card); + int (*suspend_post)(struct snd_soc_card *card); + int (*resume_pre)(struct snd_soc_card *card); + int (*resume_post)(struct snd_soc_card *card); /* callbacks */ int (*set_bias_level)(struct snd_soc_card *, diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 0fd60f4..db1dd56 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +static int raumfeld_line_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct platform_device *pdev) +static int raumfeld_line_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index b222a7d..7b72901 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -226,8 +226,7 @@ static int zylonite_remove(struct platform_device *pdev) return 0; } -static int zylonite_suspend_post(struct platform_device *pdev, - pm_message_t state) +static int zylonite_suspend_post(struct snd_soc_card *card) { if (clk_pout) clk_disable(pout); @@ -235,7 +234,7 @@ static int zylonite_suspend_post(struct platform_device *pdev, return 0; } -static int zylonite_resume_pre(struct platform_device *pdev) +static int zylonite_resume_pre(struct snd_soc_card *card) { int ret = 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 14861f9..446838e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1011,7 +1011,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_pre) - card->suspend_pre(pdev, PMSG_SUSPEND); + card->suspend_pre(card); for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; @@ -1078,7 +1078,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_post) - card->suspend_post(pdev, PMSG_SUSPEND); + card->suspend_post(card); return 0; } @@ -1090,7 +1090,6 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); - struct platform_device *pdev = to_platform_device(card->dev); struct snd_soc_codec *codec; int i; @@ -1104,7 +1103,7 @@ static void soc_resume_deferred(struct work_struct *work) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2); if (card->resume_pre) - card->resume_pre(pdev); + card->resume_pre(card); /* resume AC97 DAIs */ for (i = 0; i < card->num_rtd; i++) { @@ -1179,7 +1178,7 @@ static void soc_resume_deferred(struct work_struct *work) } if (card->resume_post) - card->resume_post(pdev); + card->resume_post(card); dev_dbg(card->dev, "resume work completed\n"); -- 1.7.2.3

2 3

Re: [alsa-devel] [[PATCH] - hdspm 1/2] Add support for RME RayDAT and AIO
by Takashi Iwai 26 Jan '11

26 Jan '11

At Wed, 26 Jan 2011 16:58:29 +0100, Adrian Knoth wrote: > > diff --git a/include/hdspm.h b/include/hdspm.h > index 81990b2..0caf472 100644 > --- a/include/hdspm.h > +++ b/include/hdspm.h ... > struct hdspm_version { > + uint8_t card_type; /* enum hdspm_io_type */ > + char cardname[20]; > + unsigned int serial; > unsigned short firmware_rev; > + int addons; > }; > > #define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, struct hdspm_version) Changing the existing ioctl isn't so pretty. If any, I'd vote for changing the ioctl number. > +/* These tables map the ALSA channels 1..N to the channels that we > + need to use in order to find the relevant channel buffer. RME > + refers to this kind of mapping as between "the ADAT channel and > + the DMA channel." We index it using the logical audio channel, > + and the value is the DMA channel (i.e. channel buffer number) > + where the data for that channel can be read/written from/to. > +*/ > + > +static char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = { > + 0, 1, 2, 3, 4, 5, 6, 7, > + 8, 9, 10, 11, 12, 13, 14, 15, > + 16, 17, 18, 19, 20, 21, 22, 23, > + 24, 25, 26, 27, 28, 29, 30, 31, > + 32, 33, 34, 35, 36, 37, 38, 39, > + 40, 41, 42, 43, 44, 45, 46, 47, > + 48, 49, 50, 51, 52, 53, 54, 55, > + 56, 57, 58, 59, 60, 61, 62, 63 > +}; Please don't put static arrays in the public header files. Otherwise it looks OK. It's huge, and maybe there are something to be fixed. But We can fix things after merging. However, the problem is that this patch isn't applicable cleanly to the latest sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git I can change the patch path easily, but the contents get rejected at this time, and I have no gut to fix manually. Could you rebase your patch to the git tree above? Or, you can use alsa-driver-snapshot tarball if you don't want to use the whole kernel git tree. thanks, Takashi

2 1

[alsa-devel] [hdspm: Add RME RayDAT/AIO 0/2] Reworked support for RayDAT/AIO
by Adrian Knoth 26 Jan '11

26 Jan '11

Hi! Finally, here's the patch that adds RME RayDAT and AIO support to hdpsm. Code was taken from Florian Faber's http://wiki.linuxproaudio.org/index.php/Driver:hdspe revision (2010-12-25) and heavily modified to (almost) comply with the coding style. I also enabled S/PDIF input and augmented the code to allow for all period sizes between 64 and 4096 samples on RayDATs/AIOs. As pointed out by Takashi earlier this month, this version now doesn't use typedefs or enums in ioctl structs. Obviously, the patch is extremely long, it adds MMAP access, support for the additional TCO hardware module, more MIDI ports and probably a lot more. Cheers PS: This is the third version of the patch, now without conflicting ioctls (hopefully) and without static arrays in the header file. Adrian Knoth (2): ALSA: Add support for RME RayDAT and AIO ALSA: Add RayDAT and AIO strings to Kconfig include/sound/hdspm.h | 349 ++++- sound/pci/Kconfig | 6 +- sound/pci/rme9652/hdspm.c | 4231 +++++++++++++++++++++++++++++++++------------ 3 files changed, 3427 insertions(+), 1159 deletions(-) -- 1.7.2.3

1 1

[alsa-devel] [PATCH] ASoC: EDB93xx machine sound driver with CS4271
by Alexander Sverdlin 26 Jan '11

26 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for EDB93xx sound with CS4271 CODEC. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Limitations: - No DAPM support Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/Kconfig | 9 +++ sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 153 insertions(+), 0 deletions(-) diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 5742904..91a28de 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE help Say Y or M here if you want to add support for AC97 audio on the Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile index 8e7977f..5514146 100644 --- a/sound/soc/ep93xx/Makefile +++ b/sound/soc/ep93xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support snd-soc-snappercl15-objs := snappercl15.o snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c new file mode 100644 index 0000000..b270085 --- /dev/null +++ b/sound/soc/ep93xx/edb93xx.c @@ -0,0 +1,142 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include "ep93xx-pcm.h" + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + unsigned int rate = params_rate(params); + /* + * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, + * because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + * MCLK = SCLK * 4 is the best recommended value, + * but we have to fall back to ratio 2 for higher + * sample rates. + */ + unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static struct platform_device *edb93xx_snd_device; + +static int __init edb93xx_init(void) +{ + int ret; + + if (!edb93xx_has_audio()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + edb93xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!edb93xx_snd_device) { + ret = -ENOMEM; + goto free_i2s; + } + + platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); + ret = platform_device_add(edb93xx_snd_device); + if (ret) + goto device_put; + + return 0; + +device_put: + platform_device_put(edb93xx_snd_device); +free_i2s: + ep93xx_i2s_release(); + return ret; +} +module_init(edb93xx_init); + +static void __exit edb93xx_exit(void) +{ + platform_device_unregister(edb93xx_snd_device); + ep93xx_i2s_release(); +} +module_exit(edb93xx_exit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL");

4 3

[alsa-devel] [hdspm: Add RME RayDAT/AIO 0/2] Reworked support for RayDAT/AIO
by Adrian Knoth 26 Jan '11

26 Jan '11

Hi! Finally, here's the patch that adds RME RayDAT and AIO support to hdpsm. Code was taken from Florian Faber's http://wiki.linuxproaudio.org/index.php/Driver:hdspe revision (2010-12-25) and heavily modified to (almost) comply with the coding style. I also enabled S/PDIF input and augmented the code to allow for all period sizes between 64 and 4096 samples on RayDATs/AIOs. As pointed out by Takashi earlier this month, this version now doesn't use typedefs or enums in ioctl structs. I hope that I caught all upstream changes that happened in the last three years. ;) Obviously, the patch is extremely long, it adds MMAP access, support for the additional TCO hardware module, more MIDI ports and probably a lot more. It's more like a new generation than an evolution, and if you think it's too much, you could still call it hdspe and make it a new driver instead, however, given that OSX and Win32 only have one hdsp driver for all cards, it feels wrong to have three on Linux. In theory, all this can be merged with the hdsp driver, but that's for some other time. ;) Cheers Adrian Knoth (2): Add support for RME RayDAT and AIO Add RayDAT and AIO strings to Kconfig include/hdspm.h | 347 ++++- pci/Kconfig | 6 +- pci/rme9652/hdspm.c | 4227 +++++++++++++++++++++++++++++++++++++-------------- 3 files changed, 3424 insertions(+), 1156 deletions(-) -- 1.7.2.3

1 1

[alsa-devel] [RFC][PATCH 0/3] ASoC: sh: fsi: sound format selection
by Kuninori Morimoto 26 Jan '11

26 Jan '11

Dear Mark, Liam, Paul These are FSI-codecs sound format selection patches. Kuninori Morimoto(3) ASoC: sh: fsi: move chan_num from fsi_stream to fsi_priv ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_info ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format FSI-codecs sound format should be selected on fsi-xxx.c for ALSA. But FSI sound format is still selected from FSI platform data for now. 3rd patch modify it. (1st and 2nd are needed from 3rd patch) But current FSI can use I2S/PCM and SPDIF today. I2S/PCM can be selected by snd_soc_dai_set_fmt from fsi-xxx, but can not select SPDIF today. So, 3rd patch add 2 modes to FSI. DAI mode and SPDIF mode. If DAI mode (default), I2S/PCM format can be selected from fsi-xxx by snd_soc_dai_set_fmt. If SPDIF mode, FSI become SPDIF format. But I'm not sure it is OK for SPDIF. So, I set these patches as [RFC] Best regards -- Kuninori Morimoto

3 6

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 26 Jan '11

26 Jan '11

The following changes since commit 20a4e7fc7e213365ea3771d7bf1e10a6bab853be: ASoC: Handle low measured DC offsets for wm_hubs devices (2011-01-21 18:20:16 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Dmitry Eremin-Solenikov (1): ASoC: correct link specifications for corgi, poodle and spitz Lars-Peter Clausen (2): ASoC: Fix codec device id format used by some dai_links ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- sound/soc/samsung/neo1973_gta02_wm8753.c | 6 +++--- sound/soc/samsung/neo1973_wm8753.c | 6 +++--- sound/soc/samsung/s3c24xx_simtec_hermes.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 ++-- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 10 files changed, 18 insertions(+), 18 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: HDA: Fix automute on Thinkpad L412/L512
by David Henningsson 26 Jan '11

26 Jan '11

More Thinkpad machines with invalid SKU found, that disables automute between speakers and headphones on these machines. I'm considering putting in a quirk mask instead, but I'm not exactly sure on how wide I should make it...any suggestion? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
by David Henningsson 26 Jan '11

26 Jan '11

Backport of commit a1d6906e2d2b4655e248f490ab088c27876a600a (currently in linux/kernel/git/tiwai/sound-2.6.git tree), meant for kernel 2.6.38. BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 41 ++++++++++++++++++++++++- 2 files changed, 40 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 16ae430..0caf77e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5066 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60 + asus Asus K52JU, Lenovo G560 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9bb030a..32bdd23 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -127,6 +127,7 @@ struct conexant_spec { unsigned int ideapad:1; unsigned int thinkpad:1; unsigned int hp_laptop:1; + unsigned int asus:1; unsigned int ext_mic_present; unsigned int recording; @@ -2312,6 +2313,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_asus_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 0); +} + + /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_hp_laptop_automic(struct hda_codec *codec) { @@ -2433,6 +2447,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } /* unsolicited event for jack sensing */ +static void cxt5066_asus_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_asus: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_asus_automic(codec); + break; + } +} + +/* unsolicited event for jack sensing */ static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); @@ -3053,6 +3081,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); + else if (spec->asus) + cxt5066_asus_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3080,6 +3110,7 @@ enum { CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3091,6 +3122,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", }; @@ -3103,6 +3135,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3112,6 +3145,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3167,13 +3201,16 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_ASUS: case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + codec->patch_ops.unsol_event = board_config == CXT5066_ASUS ? + cxt5066_asus_event : cxt5066_hp_laptop_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; - spec->hp_laptop = 1; + spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; + spec->asus = board_config == CXT5066_ASUS; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ -- 1.7.1 --------------080403010602060706040708--

1 0

[alsa-devel] [PATCH] ASoC: Staticise non-exported symbols in cs4271
by Mark Brown 26 Jan '11

26 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 5357ec5..9c5b7db 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -386,7 +386,7 @@ static struct snd_soc_dai_ops cs4271_dai_ops = { .digital_mute = cs4271_digital_mute, }; -struct snd_soc_dai_driver cs4271_dai = { +static struct snd_soc_dai_driver cs4271_dai = { .name = "cs4271-hifi", .playback = { .stream_name = "Playback", @@ -526,7 +526,7 @@ static int cs4271_remove(struct snd_soc_codec *codec) return 0; }; -struct snd_soc_codec_driver soc_codec_dev_cs4271 = { +static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .probe = cs4271_probe, .remove = cs4271_remove, .suspend = cs4271_soc_suspend, -- 1.7.2.3

2 1

[alsa-devel] [QUERY] Download calibration data to co-processor
by Patrick Lai 26 Jan '11

26 Jan '11

Hi, I am looking for ways to download audio calibration data to a co-processor under ALSA framework. I would like to know if there is already a precedent which handles this use case. I considered request_firmware() approach but there are few drawbacks to this approach. The audio calibration data size can be substantial as number of audio use cases grows. It also requires co-processor perform extensive parsing of data. Furthermore, host processor and co-processor need to establish common use case definitions so co-processor know what chunk of calibration data to apply in certain scenario such as device switch. If there is no precedent, my idea is to apply calibration data through sysfs. After introduction of ASoC multi-component architecture, PCM stream created from dailink is registered as device. So, there is syfs entry for each dailink. If ALSA ASoC can provide API for drivers(platform/CPU) to register additional attributes to the PCM device, this approach gives user-space application the ability to push down new calibration as audio situation changes. I believe it's a viable solution to my problem and perhaps can be useful to other systems as well. Your feedback is appreciated. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

[alsa-devel] [PATCH v2 0/4] Tegra: Harmony: Add internal speaker support
by Stephen Warren 26 Jan '11

26 Jan '11

Harmony has headers on the board for an internal speaker and internal mic. This patch series adds various infra-structure, and enables usage of the internal speaker. The patch series includes both changes to code within the ASoC subsystem and Tegra subsystem, the latter provided only for context when reading the ASoC changes. Would you prefer I separated these so that the ASoC maintainers can grab a whole series and apply it, rather than just a subset of the patches? While patch 2 is physically located in the Tegra subsystem, I'd consider it part of the ASoC driver. Can such a change be checked into the ASoC tree? This new header file will be required to build to ASoC driver once the following patches are in. I was considering placing this in include/sound, following the example of various codecs, but it doesn't seem a good idea to pollute that directory with platform-specific files. The current location for the file follows the example of s3c24xx/simtec. Perhaps it should be in sound/soc/tegra? Stephen Warren (4): ASoC: WM8903: Expose GPIOs through gpiolib ARM: tegra: Add Harmony sound platform data type ARM: tegra: Platform data fixes for ASoC driver updates ASoC: tegra: Harmony: Support the internal speaker arch/arm/mach-tegra/board-harmony.c | 31 ++++++ arch/arm/mach-tegra/gpio-names.h | 2 + arch/arm/mach-tegra/include/mach/harmony_audio.h | 19 ++++ include/sound/wm8903.h | 7 +- sound/soc/codecs/wm8903.c | 117 +++++++++++++++++++++- sound/soc/tegra/harmony.c | 89 ++++++++++++++-- 6 files changed, 251 insertions(+), 14 deletions(-) create mode 100644 arch/arm/mach-tegra/include/mach/harmony_audio.h

3 12

[PATCH] ALSA: HDA: Fix automute on Thinkpad L412/L512
by David Henningsson 26 Jan '11

26 Jan '11

BugLink: http://bugs.launchpad.net/bugs/707902 More Thinkpad machines with invalid SKU found, that disables automute between speakers and headphones on these machines. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index be4df4c..2fa9ed9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14954,9 +14954,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), - SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), {} -- 1.7.1 --------------050307010508090001070500--

1 0

[alsa-devel] hdspmixer
by Jaroslav Kysela 26 Jan '11

26 Jan '11

Hi, working on new ALSA release, but it seems that alsa-tools/hdspmixer is out of sync with current linux/sound/hdsp.h / hdspm.h after this commit: commit e24e56795ea57e3dd7da45063ab71f04e231192d Author: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Date: Sun Jan 9 19:42:59 2011 +0100 Add support for RME MADI, RayDAT and AIO Merged the work by Florian Faber that's distributed separately as hdspmixer64. Code taken from http://wiki.linuxproaudio.org/index.php/App:hdspmixer_64 Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Any clue what to do to get things fixed? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

5 7

[alsa-devel] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 675220 bytes (3827 ms).
by support 26 Jan '11

26 Jan '11

alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 675220 bytes (3827 ms). alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_emu10k1'. Please report this issue to the ALSA developers. http://www.alsa-project.org/db/?f=641d9cb90af2e47e527c2912bee4e34bff41b87b # rpm -qa | grep alsa alsa-lib-devel-1.0.23-2.fc14.i686 alsa-plugins-pulseaudio-1.0.22-1.fc13.i686 alsa-utils-1.0.23-3.fc14.i686 alsa-lib-1.0.23-2.fc14.i686 # uname -a Linux wp.san.ru 2.6.35.10-74.fc14.i686 #1 SMP Thu Dec 23 16:17:40 UTC 2010 i686 i686 i386 GNU/Linux -- _________________________________________ Системный администратор Участок телематических служб Цеха передачи данных г. Саратов, OАО "ВолгаТелеком", саратовский филиал тел : (8452) 272727 e-mail: support(a)san.ru

2 2

[PATCH] OMAP4: hwmod data: Add entries for DMIC
by Benoit Cousson 26 Jan '11

26 Jan '11

Adds HWMOD entries for the OMAP DMIC. The HWMOD entires define the system resource requirements for the driver such as DMA addresses, channels, and IRQ's. Placing this information in the HWMOD database allows for more generic drivers to be written and having the specific implementation details defined in HWMOD. Signed-off-by: Benoit Cousson <b-cousson(a)ti.com> Signed-off-by: David Lambert <dlambert(a)ti.com> [b-cousson(a)ti.com: Change the wrong hwmod name, add memory name, add missing flag and re-order structures] --- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 96 +++++++++++++++++++++++++++- 1 files changed, 95 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 6665922..703f3d4 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -519,7 +519,6 @@ static struct omap_hwmod omap44xx_mpu_private_hwmod = { * ctrl_module_pad_wkup * ctrl_module_wkup * debugss - * dmic * efuse_ctrl_cust * efuse_ctrl_std * elm @@ -646,6 +645,98 @@ static struct omap_hwmod omap44xx_dma_system_hwmod = { }; /* + * 'dmic' class + * digital microphone controller + */ + +static struct omap_hwmod_class_sysconfig omap44xx_dmic_sysc = { + .rev_offs = 0x0000, + .sysc_offs = 0x0010, + .sysc_flags = (SYSC_HAS_EMUFREE | SYSC_HAS_RESET_STATUS | + SYSC_HAS_SIDLEMODE | SYSC_HAS_SOFTRESET), + .idlemodes = (SIDLE_FORCE | SIDLE_NO | SIDLE_SMART | + SIDLE_SMART_WKUP), + .sysc_fields = &omap_hwmod_sysc_type2, +}; + +static struct omap_hwmod_class omap44xx_dmic_hwmod_class = { + .name = "dmic", + .sysc = &omap44xx_dmic_sysc, +}; + +/* dmic */ +static struct omap_hwmod omap44xx_dmic_hwmod; +static struct omap_hwmod_irq_info omap44xx_dmic_irqs[] = { + { .irq = 114 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_dmic_sdma_reqs[] = { + { .dma_req = 66 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_dmic_addrs[] = { + { + .name = "mpu", + .pa_start = 0x4012e000, + .pa_end = 0x4012e07f, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> dmic */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__dmic = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_dmic_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_dmic_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_dmic_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_dmic_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x4902e000, + .pa_end = 0x4902e07f, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> dmic (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__dmic_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_dmic_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_dmic_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_dmic_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* dmic slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_dmic_slaves[] = { + &omap44xx_l4_abe__dmic, + &omap44xx_l4_abe__dmic_dma, +}; + +static struct omap_hwmod omap44xx_dmic_hwmod = { + .name = "dmic", + .class = &omap44xx_dmic_hwmod_class, + .mpu_irqs = omap44xx_dmic_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_dmic_irqs), + .sdma_reqs = omap44xx_dmic_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_dmic_sdma_reqs), + .main_clk = "dmic_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_DMIC_CLKCTRL, + }, + }, + .slaves = omap44xx_dmic_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_dmic_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* * 'dsp' class * dsp sub-system */ @@ -3895,6 +3986,9 @@ static __initdata struct omap_hwmod *omap44xx_hwmods[] = { /* dma class */ &omap44xx_dma_system_hwmod, + /* dmic class */ + &omap44xx_dmic_hwmod, + /* dsp class */ &omap44xx_dsp_hwmod, &omap44xx_dsp_c0_hwmod, -- 1.7.0.4

1 0

[alsa-devel] UCM questions
by pl bossart 25 Jan '11

25 Jan '11

Hello all, I've been looking at the UCM code in git://git.alsa-project.org/alsa-lib.git branch ucm. Looks both simple/powerfl/useful, still I have a couple of questions (most likely for Liam and Mark): - how would a USB device be handled? An audio use case is defined by a verb and device parameter. My understanding is for a USB headset used for music playback, we would still use the 'HiFi' verb, but then we would need a 'USB' device added in include/use-case.h ? Then how would I make a difference between a USB headset and USB speakers? - same question for remote displays/display port. - how useful is the notion of 'card list'? Seems to me that the verbs are the main entry points, and the notion of card is abstracted away. For a given use case, the application can query the string containing the device name. Or I am misled and verbs are only related to a given card? - If I want to enable a camcorder use case, possibly with multiple microphones, do I still use the 'HiFi' verb? Or do we need a new verb for capture cases? I don't see any mics as capture devices hence the question. - how would IEC-formatted data be handled for HDMI/SPDIF? It could be a different physical device for PCM and IEC-formatted data, and you would need to enable IEC-related switches in the alsa controls. Or would a modifier be more appropriate to provide additional information on the content type? - has anyone generated a typical configuration file for USB and HDAudio? I only see an OMAP conf. Thanks for your feedback -Pierre

5 15

[alsa-devel] [GIT PULL] Branch to add harmony_audio.h to Tegra and ASoC
by Stephen Warren 25 Jan '11

25 Jan '11

Mark, Colin, Please pull: http://avon.wwwdotorg.org/downloads/kernel.git tegra_harmony_asoc_platform_data This is a branch that's suitable for merging into both ASoC's for-2.6.39 and Tegra's for-next branch, and contains a single commit to add file arch/arm/mach-tegra/include/mach/harmony_audio.h. It's branched off the latest kernel tag that Tegra's for-next pulled in (v2.6.37-rc4) and merges into both target branches without issues as far as I can tell. Note that since I wanted to avoid having to do more similar merges in the future, I added a few more fields to the platform data that I know will be needed to fully describe Harmony; the commit is now as shown below. Thanks! commit 1b1fab5161c555d25c3e3ee26b423c05b393e54c Author: Stephen Warren <swarren(a)nvidia.com> Date: Wed Jan 19 13:07:03 2011 -0700 ARM: tegra: Add Harmony sound platform data type Signed-off-by: Stephen Warren <swarren(a)nvidia.com> diff --git a/arch/arm/mach-tegra/include/mach/harmony_audio.h b/arch/arm/mach-tegra/include/mach/harmony_audio.h new file mode 100644 index 0000000..aa136a7 --- /dev/null +++ b/arch/arm/mach-tegra/include/mach/harmony_audio.h @@ -0,0 +1,23 @@ +/* + * arch/arm/mach-tegra/include/mach/harmony_audio.h + * + * Copyright 2011 NVIDIA, Inc. + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * 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. + * + */ + +struct harmony_audio_platform_data { + int gpio_spkr_en; + int gpio_hp_det; + int gpio_int_mic_en; + int gpio_ext_mic_en; + int gpio_cdc_irq; +};

2 2

[alsa-devel] [PATCH] ALSA: HDA: Fix dmesg output of HDMI supported bits
by David Henningsson 25 Jan '11

25 Jan '11

This typo caused the dmesg output of the supported bits of HDMI to be cut off early. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] alsa: fix invalid hardware.h include in ac97c for AVR32 architecture
by Hans-Christian Egtvedt 25 Jan '11

25 Jan '11

From: Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com> This patch fixes the non-compiling AC97C driver for AVR32 architecture by include mach/hardware.h only for AT91 architecture. The AVR32 architecture does not supply the hardware.h include file. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com> CC: stable(a)vger.kernel.org --- This patch also applies to the stable kernels, as the ac97c.c file has not been compilable for AVR32 since commit 7177395fdd919e561544a7d1c0ac196098a2ae2d sound/atmel/ac97c.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 10c3a87..b310702 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -33,9 +33,12 @@ #include <linux/dw_dmac.h> #include <mach/cpu.h> -#include <mach/hardware.h> #include <mach/gpio.h> +#ifdef CONFIG_ARCH_AT91 +#include <mach/hardware.h> +#endif + #include "ac97c.h" enum { -- 1.7.1

3 3

[PATCH] ALSA: HDA: Fix dmesg output of HDMI supported bits
by David Henningsson 25 Jan '11

25 Jan '11

This typo caused the dmesg output of the supported bits of HDMI to be cut off early. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_eld.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 4a66347..74b0560 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) snd_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) - snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); + snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); else if (a->max_bitrate) snprintf(buf2, sizeof(buf2), ", max bitrate = %d", a->max_bitrate); -- 1.7.1 --------------090609020005060300040600--

1 0

[alsa-devel] [PATCH] davinci: da8xx/omap-l1xx: match codec_name with i2c ids
by Rajashekhara, Sudhakar 25 Jan '11

25 Jan '11

The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c is not matching with the i2c ids in the board file. Without this fix the soundcard does not get detected on da850/omap-l138/am18x evm. Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> Tested-by: Dan Sharon <dansharon(a)nanometrics.ca> --- This patch applies to Linus's kernel tree. sound/soc/davinci/davinci-evm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 0c2d6ba..b36f0b3 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, -- 1.7.1

4 6

[alsa-devel] Fwd: aseqview patch for building with new autoconf
by Bas Wijnen 25 Jan '11

25 Jan '11

Hello, When trying to build aseqview from source (autoreconf -fis; ./configure; make), I found that it produced errors in configure. The web told me that those were caused by underquoted constructs in configure.in, which newer autoconf version have stricter requirements for than older versions. I've attached a patch which makes it build. Thanks, Bas Wijnen

1 0

[alsa-devel] [PATCH] ASoC: cs4271.c: improve error handling
by Alexander Sverdlin 25 Jan '11

25 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> CS4271 CODEC driver adapted to recently introduced error handling in snd_soc_update_bits(). Added snd_soc_cache_sync() error handling. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 59 +++++++++++++++++++++++++++++++++----------- 1 files changed, 44 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 25fed1b..5357ec5 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -209,6 +209,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); unsigned int val = 0; + int ret; switch (format & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -226,22 +227,27 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + if (ret < 0) + return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + if (ret < 0) + return ret; break; default: dev_err(codec->dev, "Invalid DAI format\n"); return -EINVAL; } - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); - + if (ret < 0) + return ret; return 0; } @@ -250,7 +256,7 @@ static int cs4271_deemph[] = {0, 44100, 48000, 32000}; static int cs4271_set_deemph(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int i; + int i, ret; int val = CS4271_DACCTL_DEM_DIS; if (cs4271->deemph) { @@ -263,8 +269,11 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - return snd_soc_update_bits(codec, CS4271_DACCTL, + ret = snd_soc_update_bits(codec, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); + if (ret < 0) + return ret; + return 0; } static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, @@ -294,7 +303,8 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - unsigned int i, ratio, val; + int i, ret; + unsigned int ratio, val; cs4271->rate = params_rate(params); ratio = cs4271->mclk / cs4271->rate; @@ -315,8 +325,10 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, else val |= cs4271_clk_tab[i].mclk_slave; - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + if (ret < 0) + return ret; return cs4271_set_deemph(codec); } @@ -324,6 +336,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + int ret; int val_a = 0; int val_b = 0; @@ -332,8 +345,12 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); - snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + if (ret < 0) + return ret; return 0; } @@ -392,17 +409,25 @@ struct snd_soc_dai_driver cs4271_dai = { #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { + int ret; /* Set power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + if (ret < 0) + return ret; return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { + int ret; /* Restore codec state */ - snd_soc_cache_sync(codec); + ret = snd_soc_cache_sync(codec); + if (ret < 0) + return ret; /* then disable the power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; return 0; } #else @@ -467,9 +492,13 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_update_bits(codec, CS4271_MODE2, 0, + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; /* Power-up sequence requires 85 uS */ udelay(85);

3 2

[alsa-devel] [PATCH] ASoC: correct link specifications for corgi, poodle and spitz
by Dmitry Eremin-Solenikov 25 Jan '11

25 Jan '11

ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms contained incorrect names for cpu_dai and codec, which effectievly disabled sound on theese platforms. Fix that errors. Cc: stable(a)vger.kernel.org Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index f451acd..135d903 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -305,10 +305,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai_name = "pxa-is2-dai", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001a", + .codec_name = "wm8731-codec-0.001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 84edd03..0eebe11 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -274,7 +274,7 @@ static struct snd_soc_dai_link poodle_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .codec_name = "wm8731-codec.0-001b", .init = poodle_wm8731_init, .ops = &poodle_ops, }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 0b30d7d..7604471 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -313,10 +313,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link spitz_dai = { .name = "wm8750", .stream_name = "WM8750", - .cpu_dai_name = "pxa-is2", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750-codec.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection
by Kuninori Morimoto 25 Jan '11

25 Jan '11

This patch add platform_device_id which can control PortA/PortB for FSI2-HDMI from platform data. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Paul Can you please check it ? because it change ap4/mackerel board arch/arm/mach-shmobile/board-ap4evb.c | 5 ++ arch/arm/mach-shmobile/board-mackerel.c | 5 ++ sound/soc/sh/fsi-hdmi.c | 66 ++++++++++++++++++++++++++++-- 3 files changed, 71 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index d503a74..920ed81 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -781,6 +781,10 @@ static struct platform_device hdmi_device = { }, }; +static struct platform_device fsi_hdmi_device = { + .name = "sh_fsi2_b_hdmi", +}; + static long ap4evb_clk_optimize(unsigned long target, unsigned long *best_freq, unsigned long *parent_freq) { @@ -934,6 +938,7 @@ static struct platform_device *ap4evb_devices[] __initdata = { &usb1_host_device, &fsi_device, &fsi_ak4643_device, + &fsi_hdmi_device, &sh_mmcif_device, &lcdc1_device, &lcdc_device, diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 425962d..aa4bcc3 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -400,6 +400,10 @@ static struct platform_device hdmi_device = { }, }; +static struct platform_device fsi_hdmi_device = { + .name = "sh_fsi2_b_hdmi", +}; + static int __init hdmi_init_pm_clock(void) { struct clk *hdmi_ick = clk_get(&hdmi_device.dev, "ick"); @@ -920,6 +924,7 @@ static struct platform_device *mackerel_devices[] __initdata = { &leds_device, &fsi_device, &fsi_ak4643_device, + &fsi_hdmi_device, &sdhi0_device, #if !defined(CONFIG_MMC_SH_MMCIF) &sdhi1_device, diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c index 96d8ce3..9719985 100644 --- a/sound/soc/sh/fsi-hdmi.c +++ b/sound/soc/sh/fsi-hdmi.c @@ -12,6 +12,12 @@ #include <linux/platform_device.h> #include <sound/sh_fsi.h> +struct fsi_hdmi_data { + const char *cpu_dai; + const char *card; + int id; +}; + static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu = rtd->cpu_dai; @@ -25,7 +31,6 @@ static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link fsi_dai_link = { .name = "HDMI", .stream_name = "HDMI", - .cpu_dai_name = "fsib-dai", /* fsi B */ .codec_dai_name = "sh_mobile_hdmi-hifi", .platform_name = "sh_fsi2", .codec_name = "sh-mobile-hdmi", @@ -33,21 +38,33 @@ static struct snd_soc_dai_link fsi_dai_link = { }; static struct snd_soc_card fsi_soc_card = { - .name = "FSI (SH MOBILE HDMI)", .dai_link = &fsi_dai_link, .num_links = 1, }; static struct platform_device *fsi_snd_device; -static int __init fsi_hdmi_init(void) +static int fsi_hdmi_probe(struct platform_device *pdev) { int ret = -ENOMEM; + const struct platform_device_id *id_entry; + struct fsi_hdmi_data *pdata; - fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); + id_entry = pdev->id_entry; + if (!id_entry) { + dev_err(&pdev->dev, "unknown fsi hdmi\n"); + return -ENODEV; + } + + pdata = (struct fsi_hdmi_data *)id_entry->driver_data; + + fsi_snd_device = platform_device_alloc("soc-audio", pdata->id); if (!fsi_snd_device) goto out; + fsi_dai_link.cpu_dai_name = pdata->cpu_dai; + fsi_soc_card.name = pdata->card; + platform_set_drvdata(fsi_snd_device, &fsi_soc_card); ret = platform_device_add(fsi_snd_device); @@ -58,9 +75,48 @@ out: return ret; } -static void __exit fsi_hdmi_exit(void) +static int fsi_hdmi_remove(struct platform_device *pdev) { platform_device_unregister(fsi_snd_device); + return 0; +} + +static struct fsi_hdmi_data fsi2_a_hdmi = { + .cpu_dai = "fsia-dai", + .card = "FSI2A (SH MOBILE HDMI)", + .id = FSI_PORT_A, +}; + +static struct fsi_hdmi_data fsi2_b_hdmi = { + .cpu_dai = "fsib-dai", + .card = "FSI2B (SH MOBILE HDMI)", + .id = FSI_PORT_B, +}; + +static struct platform_device_id fsi_id_table[] = { + /* FSI 2 */ + { "sh_fsi2_a_hdmi", (kernel_ulong_t)&fsi2_a_hdmi }, + { "sh_fsi2_b_hdmi", (kernel_ulong_t)&fsi2_b_hdmi }, + {}, +}; + +static struct platform_driver fsi_hdmi = { + .driver = { + .name = "fsi-hdmi-audio", + }, + .probe = fsi_hdmi_probe, + .remove = fsi_hdmi_remove, + .id_table = fsi_id_table, +}; + +static int __init fsi_hdmi_init(void) +{ + return platform_driver_register(&fsi_hdmi); +} + +static void __exit fsi_hdmi_exit(void) +{ + platform_driver_unregister(&fsi_hdmi); } module_init(fsi_hdmi_init); -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: Remove controls from sequenced PGA arguments
by Mark Brown 25 Jan '11

25 Jan '11

We have zero users for PGA controls and the core support for them was removed a while ago so no point in cut'n'pasting them into new macros, even if it's too much hassle to update the existing ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6c9ae23..6a25e69 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -158,11 +158,11 @@ .event = wevent, .event_flags = wflags} /* additional sequencing control within an event type */ -#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, wcontrols, \ - wncontrols, wevent, wflags) \ +#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \ + wevent, wflags) \ { .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ - .event = wevent, .event_flags = wflags, .subseq = wsubseq} + .invert = winvert, .event = wevent, .event_flags = wflags, \ + .subseq = wsubseq} #define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \ wflags) \ { .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ -- 1.7.2.3

2 1

Re: [alsa-devel] Questions about Multi-streaming playback with HDA codec ad1988
by Raymond Yau 25 Jan '11

25 Jan '11

it is strange that only this adi HDA codec does not has "stereo mix" as input source while the other adi codecs support stereo mix ? The "stereo mix" and "HD Audio Front Headphone" are available in SoundMax Control Panel it is also strange that I need to unmute "Analog Mix" in addition to those "Mic Playback" and "Line Playback Volume"

1 0

[alsa-devel] [PATCH 1/2 v2] ASoC: Fix codec device id format used by some dai_links
by Lars-Peter Clausen 25 Jan '11

25 Jan '11

The id part of an I2C device name is created with the "%d-%04x" format string. So for example for an I2C device which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This problem has been noted with the neo1993_gta02 sound board driver and has only been runtime tested for the neo1973 and neo1973_gta02 sound board drivers. But in theory the problem should be the same for all of the drivers changed by this patch. Changes since v1: * Fix only those drivers where it could be verified(By looking at the mach board file) that the codec is connected through an I2C bus. --- sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- 6 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e..5e4d499 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24..ad28663 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 3eec610..9e05e10 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -401,7 +401,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a2451..cf69e14 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index bb4292e..287a971 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -94,7 +94,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index fbba4e3..d2b14ba 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -85,7 +85,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", -- 1.7.2.3

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[alsa-devel] [PATCH] EDB93xx: Add support for CS4271 CODEC on EDB93xx boards
by Alexander Sverdlin 25 Jan '11

25 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Add support for CS4271 SPI-connected CODEC on EDB93xx boards. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 54 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 54 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 4b04316..9f9193d 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,12 +30,21 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> +#include <mach/ep93xx_spi.h> #include <mach/hardware.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) static void __init edb93xx_register_flash(void) { @@ -93,6 +102,49 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data cs4271_edb93xx_gpio = { + .gpio_nreset = EP93XX_GPIO_LINE_EGPIO1, + .gpio_disable = EP93XX_GPIO_LINE_EGPIO6, +}; + +static struct spi_board_info edb93xx_spi_board_info[] = { + { + .modalias = "cs4271", + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + .platform_data = &cs4271_edb93xx_gpio, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_spi(&edb93xx_spi_info, + edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); + } +} + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -117,6 +169,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); }

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[alsa-devel] Should "Master" control the "Line In" volume (loopback)
by Colin Guthrie 25 Jan '11

25 Jan '11

Hi, I have a question-by-proxy from a user who is confused because his "Master" kcontrol is affecting the volume via Line In (which he uses to loopback an analog signal to his speakers). I've not got much hardware here for comparison, but I'm pretty sure that Master usually doesn't affect Line In. So couple of questions: 1. Is it typical that Master affects Line In? 2. If not, can it be fixed for this h/w? ALSA info here: https://bzattachment.mandriva.com/attachment.cgi?id=18170 (for reference the full bug is here: https://qa.mandriva.com/show_bug.cgi?id=62218) Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] [PATCH 1/1] SOUND: azt3328, fix broken AZF_FMT_XLATE macro
by Jiri Slaby 25 Jan '11

25 Jan '11

Despite the AZF_FMT_XLATE macro looks totally weird and useless, it's heavily broken (apart it also broke my parser). The 'break' in the macro belongs to the do-while loop and not to the switch. So it always falls through the all cases till the end. Remove the do-while from the macro. Also people should terminate statements with semicolons. So force people to do so by removing the last one in the macro. I vote for removing that crap completely because it makes the code weird anyway -- you have to specify manually both of freq and bits which is I would expect to be avoided exactly by such a macro. Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> Cc: Andreas Mohr <andi(a)lisas.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/azt3328.c | 32 +++++++++++++++----------------- 1 files changed, 15 insertions(+), 17 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 6117595..134f55f 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -980,29 +980,27 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, snd_azf3328_dbgcallenter(); switch (bitrate) { #define AZF_FMT_XLATE(in_freq, out_bits) \ - do { \ case AZF_FREQ_ ## in_freq: \ freq = SOUNDFORMAT_FREQ_ ## out_bits; \ - break; \ - } while (0); - AZF_FMT_XLATE(4000, SUSPECTED_4000) - AZF_FMT_XLATE(4800, SUSPECTED_4800) + break + AZF_FMT_XLATE(4000, SUSPECTED_4000); + AZF_FMT_XLATE(4800, SUSPECTED_4800); /* the AZF3328 names it "5510" for some strange reason: */ - AZF_FMT_XLATE(5512, 5510) - AZF_FMT_XLATE(6620, 6620) - AZF_FMT_XLATE(8000, 8000) - AZF_FMT_XLATE(9600, 9600) - AZF_FMT_XLATE(11025, 11025) - AZF_FMT_XLATE(13240, SUSPECTED_13240) - AZF_FMT_XLATE(16000, 16000) - AZF_FMT_XLATE(22050, 22050) - AZF_FMT_XLATE(32000, 32000) + AZF_FMT_XLATE(5512, 5510); + AZF_FMT_XLATE(6620, 6620); + AZF_FMT_XLATE(8000, 8000); + AZF_FMT_XLATE(9600, 9600); + AZF_FMT_XLATE(11025, 11025); + AZF_FMT_XLATE(13240, SUSPECTED_13240); + AZF_FMT_XLATE(16000, 16000); + AZF_FMT_XLATE(22050, 22050); + AZF_FMT_XLATE(32000, 32000); default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); /* fall-through */ - AZF_FMT_XLATE(44100, 44100) - AZF_FMT_XLATE(48000, 48000) - AZF_FMT_XLATE(66200, SUSPECTED_66200) + AZF_FMT_XLATE(44100, 44100); + AZF_FMT_XLATE(48000, 48000); + AZF_FMT_XLATE(66200, SUSPECTED_66200); #undef AZF_FMT_XLATE } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ -- 1.7.3.2

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[alsa-devel] HDMI with more than 2ch ?
by Xavier Bestel 25 Jan '11

25 Jan '11

Hi, I'm using HDMI output on an HD 2600, using KMS from kernel 2.6.37: http://www.alsa-project.org/db/?f=b3f67a0890f9878b7b3f345578e24eda0aaa59ac The problem is that only 2ch are supported (not 1, not 3 or more). I tested with speaker-test, it only works with -c 2. So how can I access the other PCM channels ? There are 8 of them available in the HDMI standard, I thought the card should be able to drive all of them. Thanks, Xav

2 3

[alsa-devel] Bug 0002542
by George Morgan 24 Jan '11

24 Jan '11

Looking at the notes for Bug 0002542 (as a guest) it looks like there is an issue in the file: alsa-oss-1.0.17/alsa/aoss.in I think line 10 should be: LD_PRELOAD=@libdir@/libaoss.so${LD_PRELOAD:+:LD_PRELOAD} exec "$@" instead of: LD_PRELOAD=${exec_prefix}/\$LIB/libaoss.so${LD_PRELOAD:+:LD_PRELOAD} exec "$@" Let me know if I need to create an account in bugtrack and submit a patch. Thanks. George Morgan

2 3

[alsa-devel] [PATCH] ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct SPDIF output
by Andy Robinson 24 Jan '11

24 Jan '11

The attached patch adds a quirk to use the newly-added asus model for the Asus U50F. It also selects the correct digital outputs based on the pin config instead of always using node 0x21, since the Asus U50F uses node 0x12 for digital out.

2 3

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 24 Jan '11

24 Jan '11

The following changes since commit f758b53a79ce6076542739563985d5ff0521dc5a: Merge branch 'topic/hda' (2011-01-22 17:29:50 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Usual small, driver-specific updates. Dimitris Papastamos (1): ASoC: WM8995: Fix incorrect use of snd_soc_update_bits() Mark Brown (1): ASoC: Handle low measured DC offsets for wm_hubs devices Qiao Zhou (1): ASoC: WM8994: fix wrong value in tristate function Rajashekhara, Sudhakar (1): ASoC: da8xx/omap-l1xx: match codec_name with i2c ids sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm_hubs.c | 15 +++++++-------- sound/soc/davinci/davinci-evm.c | 2 +- 4 files changed, 10 insertions(+), 11 deletions(-)

2 1

[alsa-devel] Support of different channels for ADC driver
by subashchandra.bose＠wipro.com 24 Jan '11

24 Jan '11

Hi All, I am developing General ADC driver with input data for voice channel, High Speed channel and low speed channel. I went into alsa-project website and found alsa pcm interface can be used to configure voice channel. How can I use to configure high speed and low speed channels as well in same driver. Thanks in Advance, -Subash. Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[alsa-devel] [PATCH] ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662
by Anisse Astier 24 Jan '11

24 Jan '11

--- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7262980..a72c8bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18292,6 +18292,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), {} }; -- 1.7.3.2

3 2

Re: [alsa-devel] Bug in setting channels, related to recent ELD/EDID changes?
by Anssi Hannula 24 Jan '11

24 Jan '11

(restored CC I removed, sorry about that) On 11.01.2011 09:15, Takashi Iwai wrote: > At Mon, 10 Jan 2011 21:36:10 +0100, > Takashi Iwai wrote: >> >> At Mon, 10 Jan 2011 21:58:26 +0200, >> Anssi Hannula wrote: >>> >>> >>> Takashi Iwai kirjoitti: >>>> At Sun, 02 Jan 2011 12:16:44 +0200, >>>> Anssi Hannula wrote: >>>>> >>>>> Hi! >>>>> >>>>> Takashi, it seems the "allow all when not plugged in" behaviour you >>>>> implemented in bbbe3390 is not enough to allow applications to open the >>>>> device before the output device is ready. Switched-off A/V receivers may >>>>> provide the information of the plugged-in television which usually >>>>> restricts the max_channels to 2, causing applications to fail opening >>>>> the device with more channels when a the A/V receiver is switched off. >>>>> >>>>> Also, even if the application can gracefully fallback to stereo, AFAIK >>>>> there is no method for an application to get informed when the >>>>> limitation is lifted, so it can't automatically resume multichannel >>>>> output when the A/V receiver is switched on again. >>>>> >>>>> Based on this info, I guess the restrictions based on ELD should be >>>>> removed :/ >>>>> Unless you have some other ideas to fix the issue, of course. >>>> >>>> What about the patch below? If it's unplugged, the valid flag should >>>> be cleared, so the next open fall backs to the default state (i.e. >>>> allows all). >>> >>> But it is not unplugged. The multichannel-capable receiver forwards EDID >>> of TV which doesn't allow more than 2 channels, when the receiver is >>> swtiched off. >> >> Well, in that case, this is the correct behavior, IMO. At this >> moment, setting channels to 2 is the right thing. Then, how can you >> know that more channels will be available in future at all? >> >> What you want to have a static configuration way by ignoring the ELD >> information. But, the interface isn't pretty easy if it's implemented >> in control API, I suppose. > > ... or, maybe an easier option is to use the module parameter like > the patch below. It can be changed dynamically via sysfs as well. Better than nothing I guess. Note that the condition is backwards in the below patch :) > --- > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index d1b1b57..4115a39 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -31,10 +31,14 @@ > #include <linux/init.h> > #include <linux/delay.h> > #include <linux/slab.h> > +#include <linux/moduleparam.h> > #include <sound/core.h> > #include "hda_codec.h" > #include "hda_local.h" > > +static bool static_hdmi_pcm; > +module_param(static_hdmi_pcm, bool, 0644); > + > /* > * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device > * could support two independent pipes, each of them can be connected to one or > @@ -827,7 +831,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, > *codec_pars = *hinfo; > > eld = &spec->sink_eld[idx]; > - if (eld->sad_count > 0) { > + if (static_hdmi_pcm && eld->sad_count > 0) { > hdmi_eld_update_pcm_info(eld, hinfo, codec_pars); > if (hinfo->channels_min > hinfo->channels_max || > !hinfo->rates || !hinfo->formats) -- Anssi Hannula

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[alsa-devel] [PATCH] ASoC: Fix type for snd_soc_volatile_register()
by Mark Brown 24 Jan '11

24 Jan '11

We generally refer to registers as unsigned ints (including in the underlying CODEC driver operation). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 22 +++++++++++++++++++++- 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c184f84..1355ef0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -267,7 +267,8 @@ 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 snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b0e7689..ba81892 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2029,7 +2029,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) * * Boolean function indiciating if a CODEC register is volatile. */ -int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { if (codec->volatile_register) return codec->volatile_register(codec, reg); @@ -2039,6 +2040,25 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); /** + * snd_soc_codec_register_name: Return a plain text register name + * + * @codec: CODEC to query. + * @reg: Register to query for. + * + * Returns a string with a plain text name for the register, or NULL + * if none is available. + */ +const char *snd_soc_codec_register_name(struct snd_soc_codec *codec, + unsigned int reg) +{ + if (codec->volatile_register) + return codec->register_name(codec, reg); + else + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_register_name); + +/** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec * @ops: AC97 bus operations -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Fix codec device id format used by some dai_links
by Lars-Peter Clausen 24 Jan '11

24 Jan '11

The id part of a I2C device name is created with the "%d-%04x" format string. So for example for a I2C device, which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org (for 2.6.37) --- This problem has been noted with the neo1993_gta02 sound board driver and has only been runtime tested for the neo1973 and neo1973_gta02 sound board drivers, but in theory the problem should be the same for all of the drivers changed by this patch. --- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/imx/wm1133-ev1.c | 2 +- sound/soc/samsung/goni_wm8994.c | 4 ++-- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- 11 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 1aac2f4..e2b6595 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -356,7 +356,7 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = { .stream_name = "WM8510 PCM", .cpu_dai_name= "atmel-ssc-dai.0", .platform_name = "atmel-pcm-audio", - .codec_name = "wm8510-codec.0-0x1a", + .codec_name = "wm8510-codec.0-001a", .codec_dai_name = "wm8510-hifi", .init = playpaq_wm8510_init, .ops = &playpaq_wm8510_ops, diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e..5e4d499 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24..ad28663 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 75b4c72..f507e57 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -246,7 +246,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { .cpu_dai_name = "imx-ssi.0", .codec_dai_name = "wm8350-hifi", .platform_name = "imx-fiq-pcm-audio.0", - .codec_name = "wm8350-codec.0-0x1a", + .codec_name = "wm8350-codec.0-001a", .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, .symmetric_rates = 1, diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index f6b3a3c..3a7861c 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -238,7 +238,7 @@ static struct snd_soc_dai_link goni_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8994-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8994-codec.0-0x1a", + .codec_name = "wm8994-codec.0-001a", .init = goni_wm8994_init, .ops = &goni_hifi_ops, }, { @@ -247,7 +247,7 @@ static struct snd_soc_dai_link goni_dai[] = { .cpu_dai_name = "goni-voice-dai", .codec_dai_name = "wm8994-voice", .platform_name = "samsung-audio", - .codec_name = "wm8994-codec.0-0x1a", + .codec_name = "wm8994-codec.0-001a", .ops = &goni_voice_ops, }, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 3b53ad5..14eb6ea 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 69e08fd..5e77cd1 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -391,7 +391,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -400,7 +400,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a2451..cf69e14 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index d7b3e6e..2fa6d1b 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -84,7 +84,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index ff6168f..d219223 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -73,7 +73,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 0a2c4f2..d14f939 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -207,7 +207,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, }, -- 1.7.2.3

4 8

[alsa-devel] commit 109fef9edcc100952eec980acbc2e1295627fbab ALSA: timer: automatically load the high-resolution timer causing MIDI playback tempo problems
by Arthur Marsh 23 Jan '11

23 Jan '11

Hi, on an old machine I have with Intel 440BX chipset and PII-266 CPU, I found recent kernels causing MIDI sound output from a Soundblaster Audigy 2 ZS to be varying widely in tempo. dmesg reports with recent kernels (both those that allow correct playback of MIDI files and those that give wide variation in playback tempo): Clocksource tsc unstable (delta = 132040172 ns) Switching to clocksource pit After git-bisection I found: git bisect bad 109fef9edcc100952eec980acbc2e1295627fbab is the first bad commit commit 109fef9edcc100952eec980acbc2e1295627fbab Author: Clemens Ladisch <clemens(a)ladisch.de> Date: Thu Nov 18 09:53:54 2010 +0100 ALSA: timer: automatically load the high-resolution timer Increase the default timer limit so that snd-hrtimer.ko can be automatically loaded when needed, e.g., when used as the default sequencer timer. This replaces the check for the obsolete CONFIG_SND_HPET. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 c7b3b046ea99e563228f36ea4a00f9b763bf5813 741f137a201b31bdf5eaaa58c3031b4b2a68c6d6 M sound diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..b3aaa60 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -34,8 +34,8 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HPET) || defined(CONFIG_SND_HPET_MODULE) -#define DEFAULT_TIMER_LIMIT 3 +#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#define DEFAULT_TIMER_LIMIT 4 #elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) #define DEFAULT_TIMER_LIMIT 2 #else I am currently rebuilding kernel 2.6.38-rc2 with this patch reverted to verify that this commit triggers the problem with this machine. Arthur.

1 1

[alsa-devel] [PATCH] pcm: fix typo in snd_pcm_recover() error message
by cosoleto＠gmail.com 23 Jan '11

23 Jan '11

From: Francesco Cosoleto <cosoleto(a)gmail.com> Signed-off-by: Francesco Cosoleto <cosoleto(a)gmail.com> --- src/pcm/pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 906b502..02dea0d 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7313,7 +7313,7 @@ int snd_pcm_recover(snd_pcm_t *pcm, int err, int silent) else s = "overrun"; if (!silent) - SNDERR("%s occured", s); + SNDERR("%s occurred", s); err = snd_pcm_prepare(pcm); if (err < 0) { SNDERR("cannot recovery from %s, prepare failed: %s", s, snd_strerror(err)); -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: EDB93xx machine sound driver with CS4271
by Alexander Sverdlin 23 Jan '11

23 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for EDB93xx sound with CS4271 CODEC. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Limitations: - No DAPM support PS. The same code as submitted on 8 Dec 2010, but now all dependencies are applied. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 54 +++++++++++++++ sound/soc/ep93xx/Kconfig | 9 +++ sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 207 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 4b04316..f453e5a 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,12 +30,21 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> +#include <mach/ep93xx_spi.h> #include <mach/hardware.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) static void __init edb93xx_register_flash(void) { @@ -93,6 +102,49 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data cs4271_edb93xx_gpio = { + .gpio_nreset = EP93XX_GPIO_LINE_EGPIO1, + .gpio_disable = EP93XX_GPIO_LINE_EGPIO6, +}; + +static struct spi_board_info edb93xx_spi_board_info[] = { + { + .modalias = "cs4271-codec", + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + .platform_data = &cs4271_edb93xx_gpio, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_spi(&edb93xx_spi_info, + edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); + } +} + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -117,6 +169,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); } diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 5742904..91a28de 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE help Say Y or M here if you want to add support for AC97 audio on the Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile index 8e7977f..5514146 100644 --- a/sound/soc/ep93xx/Makefile +++ b/sound/soc/ep93xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support snd-soc-snappercl15-objs := snappercl15.o snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c new file mode 100644 index 0000000..b270085 --- /dev/null +++ b/sound/soc/ep93xx/edb93xx.c @@ -0,0 +1,142 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include "ep93xx-pcm.h" + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + unsigned int rate = params_rate(params); + /* + * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, + * because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + * MCLK = SCLK * 4 is the best recommended value, + * but we have to fall back to ratio 2 for higher + * sample rates. + */ + unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static struct platform_device *edb93xx_snd_device; + +static int __init edb93xx_init(void) +{ + int ret; + + if (!edb93xx_has_audio()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + edb93xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!edb93xx_snd_device) { + ret = -ENOMEM; + goto free_i2s; + } + + platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); + ret = platform_device_add(edb93xx_snd_device); + if (ret) + goto device_put; + + return 0; + +device_put: + platform_device_put(edb93xx_snd_device); +free_i2s: + ep93xx_i2s_release(); + return ret; +} +module_init(edb93xx_init); + +static void __exit edb93xx_exit(void) +{ + platform_device_unregister(edb93xx_snd_device); + ep93xx_i2s_release(); +} +module_exit(edb93xx_exit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL");

2 1

Re: [alsa-devel] ALSA Multichannel through HDMI
by John Ettedgui 23 Jan '11

23 Jan '11

On Thu, Nov 25, 2010 at 6:21 PM, John Ettedgui <john.ettedgui(a)gmail.com> wrote: > On Thu, Nov 25, 2010 at 6:16 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >> On 26.11.2010 04:02, John Ettedgui wrote: >>> On Thu, Nov 25, 2010 at 5:08 PM, John Ettedgui <john.ettedgui(a)gmail.com> wrote: >>>> On Thu, Nov 25, 2010 at 4:59 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>> On 26.11.2010 02:53, John Ettedgui wrote: >>>>>> On Thu, Nov 25, 2010 at 3:34 PM, John Ettedgui <john.ettedgui(a)gmail.com> wrote: >>>>>>> On Thu, Nov 25, 2010 at 3:15 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>> On 26.11.2010 01:11, John Ettedgui wrote: >>>>>>>>> On Thu, Nov 25, 2010 at 2:58 PM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>> On 26.11.2010 00:42, John Ettedgui wrote: >>>>>>>>>>> On Thu, Nov 25, 2010 at 11:19 AM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>>>> On 25.11.2010 14:21, John Ettedgui wrote: >>>>>>>>>>>>> On Thu, Nov 25, 2010 at 3:40 AM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>>>>>> On 25.11.2010 13:09, John Ettedgui wrote: >>>>>>>>>>>>>>> On Thu, Nov 25, 2010 at 2:49 AM, Anssi Hannula <anssi.hannula(a)iki.fi> wrote: >>>>>>>>>>>>>>>> On 25.11.2010 06:12, John Ettedgui wrote: >>>>>>>>>>>>>>>>> I'm actually trying to get a 5.1 setup working without passthrough, if >>>>>>>>>>>>>>>>> I use mplayer with passthrough my receiver behaves just fine, but I am >>>>>>>>>>>>>>>>> trying to send the 6 channels from the computer. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I was actually told that the current ALSA driver for radeon's hdmi >>>>>>>>>>>>>>>>> does not support more than 2 channels, and that would explain why I'm >>>>>>>>>>>>>>>>> getting this behavior. Is that true though? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Yes. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> However, strangely your log output seems to show the intel's hdmi parser >>>>>>>>>>>>>>>> being used instead: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> ALSA sound/pci/hda/patch_intelhdmi.c:842: hdmi_setup_stream: NID=0x2, >>>>>>>>>>>>>>>>> stream=0x1, new-format=0x11 >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Were you just trying some patch at the time? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> yes Wu gave me a patch to try against the kernel, I'm guessing this is >>>>>>>>>>>>>>> the reason you see something unexpected. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Did 6 channels work with that patch? (possibly with a wrong channel order) >>>>>>>>>>>>> Not any better as far as I remember. >>>>>>>>>>>> >>>>>>>>>>>> OK. Did 2 channel audio continue to work with it? >>>>>>>>>>>> >>>>>>>>>>>> If you are not sure, here's the patch doing the same thing against >>>>>>>>>>>> current alsa driver: >>>>>>>>>>>> http://stuff.onse.fi/0001-ALSA-hda-Use-generic-HDMI-code-for-ATI-HDMI-codec… >>>>>>>>>>>> >>>>>>>>>>>> Confirming that would help (it would allow to switch the ATI chip to use >>>>>>>>>>>> the generic parser instead of the ATI-specific one), though not with the >>>>>>>>>>>> multichannel issue (see below). >>>>>>>>>>>> >>>>>>>>>>>> [...] >>>>>>>>>>>> >>>>>>>>>>>>>>> Since I just moved to jack/rca cables I'm in no hurry anymore, but is >>>>>>>>>>>>>>> there any reason why the driver cannot work with more than 2 channels? >>>>>>>>>>>>>> >>>>>>>>>>>>>> I don't really know (it is simply not implemented). >>>>>>>>>>>>> Alright. >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Anything I could do to help with that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Possibly. But see above, so that we won't try the same things you >>>>>>>>>>>>>> already tried :) >>>>>>>>>>>>>> >>>>>>>>>>>>> Sure :) >>>>>>>>>>>> >>>>>>>>>>>> Looking again at the the alsa info output you provided, I don't think I >>>>>>>>>>>> have any great ideas. It looks like the chip simply reports that it >>>>>>>>>>>> supports a maximum of 2 channels. >>>>>>>>>>>> >>>>>>>>>>>> One option could be hardcoding "chans = 8;" in generic_hdmi_build_pcms() >>>>>>>>>>>> in sound/pci/hda/patch_hdmi.c (assuming you have the above patch and 2 >>>>>>>>>>>> channel audio works with it). However, I find it rather unlikely that it >>>>>>>>>>>> would make any difference. >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Anssi Hannula >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Anssi, >>>>>>>>>>> the patch fails on my kernel, I cannot find the hda_codec_preset >>>>>>>>>>> manually in there either. >>>>>>>>>>> Did you want me to try that against a 2.6.37 rc instead of a 2.6.36? >>>>>>>>>> >>>>>>>>>> For 2.6.36 I think you can just use your earlier patch. >>>>>>>>>> >>>>>>>>>> Note that on that kernel generic_hdmi_build_pcms() is >>>>>>>>>> intel_hdmi_build_pcms() in patch_intelhdmi.c. >>>>>>>>>> >>>>>>>>> ok >>>>>>>>>> BTW, I've just been informed by another ATI user that his card was >>>>>>>>>> actually stereo-only, while he thought earlier it was multichannel >>>>>>>>>> capable. So, are you sure your card is a multichannel one? >>>>>>>>>> (if not, it would explain why the card says it is not) >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anssi Hannula >>>>>>>>>> >>>>>>>>> Yes, it works fine in Window so I believe the hardware part is good. >>>>>>>> >>>>>>>> And you are sure it is multichannel PCM, and not e.g. multichannel DTS >>>>>>>> or multichannel AC-3 with on-the-fly compression, which are possible >>>>>>>> with stereo hdmi as well? >>>>>>>> >>>>>>>> Just making sure :) >>>>>>>> >>>>>>>> -- >>>>>>>> Anssi Hannula >>>>>>>> >>>>>>> >>>>>>> Well, I have not super tested it, but the reason I believe it is is >>>>>>> simple, when I use passthrough my receiver says Dolby digital or >>>>>>> something like that, but in Windows just going around or playing games >>>>>>> it says Multichannel. >>>>> >>>>> Sounds ok then. >>>>> >>>>>>> I'll get the patch applied in a few minutes, I was running out of >>>>>>> memory and my system was getting really slow... :) >>>>>>> >>>>>> >>>>>> I am not sure if I did it right but now I don't even get hdmi sound >>>>>> working anymore. >>>>>> It is not listed in aplay-l, and dmesg gives me this: >>>>>> [drm:radeon_dvi_detect] *ERROR* HDMI Type A-1: probed a monitor but >>>>>> no|invalid EDID >>>>>> >>>>>> I've attached the patch I used, since I could not used the previous >>>>>> one I had to hack it, maybe I broke something? >>>>>> (I have not set channels to 8 yet, just changed the parser) >>>>> >>>>> The patch looks ok. The above error message is weird, as it comes from >>>>> the display driver which should not be affected by the patch at all. >>>>> >>>>> I'll look at the code later (tomorrow maybe). Until then I'd advise you >>>>> to retry without the patch and with the patch again, just to make sure >>>>> this is not some unrelated issue... >>>>> Also, post the whole dmesg when it works (2 channel, without patch), and >>>>> when it doesn't (with patch). >>>>> >>>>> -- >>>>> Anssi Hannula >>>>> >>>> Following your message I restarted my computer and that error was >>>> gone, yet there is still nothing display in aplay -l / aplay -L about >>>> hdmi anymore. >>>> I'll go back to my non patched kernel and see what it does. >>>> >>> With the non-patched kernel I get back my hdmi in aplay -l, so it >>> looks like the patch messed up something. >> >> Indeed. >> >>> I really think that when I tried it before it did not do that, but it >>> was so long ago that I could be wrong... >> >> Well, the driver has changed since then, it could be more strict now :) >> >>> Anyway here are the 2 dmesgs (not sure if they'll help though). >> >> Indeed, there doesn't seem to be any difference. Maybe try with >> CONFIG_SND_DEBUG_VERBOSE? >> >> -- >> Anssi Hannula >> > > Sure I'll get that. > There you go.

2 6

Re: [alsa-devel] [PATCH][RFC] Sound, Xonar, CS43xx: Don't overrun static array
by Takashi Iwai 22 Jan '11

22 Jan '11

At Thu, 20 Jan 2011 22:37:43 +0100 (CET), Jesper Juhl wrote: > > > 'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of > 8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers() > > for (i = 2; i <= 8; ++i) > snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); > > will overrun the array when 'i == 8'. > > I guess that what's needed to fix it is the trivial patch below, but I > must admit that I have no idea about this code, so I may very well be > wrong. Additionally, I have no way to actually test this, so all I know is > that the below compiles. Someone who actually knows this code should take > a look before anything is comitted - consider the below (not much more > than) a bug report. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> Applied now. Thanks. Takashi > --- > xonar_cs43xx.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c > index 9f72d42..2527191 100644 > --- a/sound/pci/oxygen/xonar_cs43xx.c > +++ b/sound/pci/oxygen/xonar_cs43xx.c > @@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip, > unsigned int i; > > snd_iprintf(buffer, "\nCS4398: 7?"); > - for (i = 2; i <= 8; ++i) > + for (i = 2; i < 8; ++i) > snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); > snd_iprintf(buffer, "\n"); > dump_cs4362a_registers(data, buffer); > > -- > Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ > Don't top-post http://www.catb.org/~esr/jargon/html/T/top-post.html > Plain text mails only, please. >

1 0

Re: [alsa-devel] [PATCH][RFC] Sound, Xonar, CS43xx: Don't overrun static array
by Clemens Ladisch 22 Jan '11

22 Jan '11

Jesper Juhl wrote: > 'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of > 8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers() > > for (i = 2; i <= 8; ++i) > snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); > > will overrun the array when 'i == 8'. > > I guess that what's needed to fix it is the trivial patch below, but I > must admit that I have no idea about this code, so I may very well be > wrong. Additionally, I have no way to actually test this, so all I know is > that the below compiles. Someone who actually knows this code should take > a look before anything is comitted - consider the below (not much more > than) a bug report. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> Acked-by: Clemens Ladisch <clemens(a)ladisch.de> > --- > xonar_cs43xx.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/xonar_cs43xx.c > b/sound/pci/oxygen/xonar_cs43xx.c > index 9f72d42..2527191 100644 > --- a/sound/pci/oxygen/xonar_cs43xx.c > +++ b/sound/pci/oxygen/xonar_cs43xx.c > @@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip, > unsigned int i; > > snd_iprintf(buffer, "\nCS4398: 7?"); > - for (i = 2; i <= 8; ++i) > + for (i = 2; i < 8; ++i) > snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); > snd_iprintf(buffer, "\n"); > dump_cs4362a_registers(data, buffer);

1 0

[alsa-devel] [PATCH v3 0/4] Tegra: Harmony: Add internal speaker support
by Stephen Warren 22 Jan '11

22 Jan '11

Harmony has headers on the board for an internal speaker and internal mic. This patch series adds various infra-structure, and enables usage of the internal speaker. The patch series includes both changes to code within the ASoC subsystem and Tegra subsystem, the latter provided only for context when reading the ASoC changes. Would you prefer I separated these so that the ASoC maintainers can grab a whole series and apply it, rather than just a subset of the patches? While patch 2 is physically located in the Tegra subsystem, I'd consider it part of the ASoC driver. Can such a change be checked into the ASoC tree? This new header file will be required to build to ASoC driver once the following patches are in. I was considering placing this in include/sound, following the example of various codecs, but it doesn't seem a good idea to pollute that directory with platform-specific files. The current location for the file follows the example of s3c24xx/simtec. Perhaps it should be in sound/soc/tegra? Changelog: * v3: * Reverted change to WM8903_GPIO_NO_CONFIG * Program WM8903_GP1_FN_MASK in direction_{input,output} * Simplify harmony_event_int_spk * Didn't remove GPIO entries from harmony_wm8903_pdata, since this also sets up all the other fields to 0 too. Stephen Warren (4): ASoC: WM8903: Expose GPIOs through gpiolib ARM: tegra: Add Harmony sound platform data type ARM: tegra: Platform data fixes for ASoC driver updates ASoC: tegra: Harmony: Support the internal speaker arch/arm/mach-tegra/board-harmony.c | 31 ++++++ arch/arm/mach-tegra/gpio-names.h | 2 + arch/arm/mach-tegra/include/mach/harmony_audio.h | 19 ++++ include/sound/wm8903.h | 20 ++++- sound/soc/codecs/wm8903.c | 126 +++++++++++++++++++++- sound/soc/tegra/harmony.c | 87 +++++++++++++-- 6 files changed, 272 insertions(+), 13 deletions(-) create mode 100644 arch/arm/mach-tegra/include/mach/harmony_audio.h

5 26

[alsa-devel] [PATCH 0/3] ASoC: fsi/ak4642/da7210: select sound format from fsi-codecs
by Kuninori Morimoto 21 Jan '11

21 Jan '11

Dear Mark, Liam, Paul These patches are for fsi-ak4642, fsi-da7210 to select sound format. Kuninori Morimoto (3): ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support ASoC: sh: fsi: Add fsi_get_priv_frm_dai function ASoC: sh: fsi: Add snd_soc_dai_set_fmt support In current fsi-codecs, fsi format is selected from platform, and codecs format is selected from fsi-codecs. These patches modify it, and select format from fsi-codecs only. ak4642 didn't have SND_SOC_DAIFMT_FORMAT, so 1st patch add it. (da7210 already have it) These patches are tested on Ecovec, AP4EVB, Mackerel board, and Paul's rmobile-latest + Mark's for-2.6.38 branch. Best regards -- Kuninori Morimoto

4 15

[alsa-devel] [PATCH v2] ASoC: soc-cache: Add trace event for snd_soc_cache_sync()
by Dimitris Papastamos 21 Jan '11

21 Jan '11

This patch makes it easy to see when the syncing process begins and ends. You can also enable the snd_soc_reg_write tracepoint to see which registers are being synced. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/trace/events/asoc.h | 25 +++++++++++++++++++++++++ sound/soc/soc-cache.c | 10 ++++++++++ 2 files changed, 35 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 186e84d..ae973d2 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -229,6 +229,31 @@ TRACE_EVENT(snd_soc_jack_notify, TP_printk("jack=%s %x", __get_str(name), (int)__entry->val) ); +TRACE_EVENT(snd_soc_cache_sync, + + TP_PROTO(struct snd_soc_codec *codec, const char *type, + const char *status), + + TP_ARGS(codec, type, status), + + TP_STRUCT__entry( + __string( name, codec->name ) + __string( status, status ) + __string( type, type ) + __field( int, id ) + ), + + TP_fast_assign( + __assign_str(name, codec->name); + __assign_str(status, status); + __assign_str(type, type); + __entry->id = codec->id; + ), + + TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), + (int)__entry->id, __get_str(type), __get_str(status)) +); + #endif /* _TRACE_ASOC_H */ /* This part must be outside protection */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f834839..db66dc4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include <linux/bitmap.h> #include <linux/rbtree.h> +#include <trace/events/asoc.h> + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -1601,18 +1603,26 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); int snd_soc_cache_sync(struct snd_soc_codec *codec) { int ret; + const char *name; if (!codec->cache_sync) { return 0; } + if (codec->cache_ops->name) + name = codec->cache_ops->name; + else + name = "unknown"; + if (codec->cache_ops && codec->cache_ops->sync) { if (codec->cache_ops->name) dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); ret = codec->cache_ops->sync(codec); if (!ret) codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); return ret; } -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: cs4271.c: adapt to error handling introduced in snd_soc_update_bits()
by Alexander 21 Jan '11

21 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> CS4271 CODEC driver adapted to recently introduced error handling in snd_soc_update_bits(). Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 58 +++++++++++++++++++++++++++++++++----------- 1 files changed, 43 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 25fed1b..9cfb796 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -209,6 +209,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); unsigned int val = 0; + int ret; switch (format & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -226,22 +227,27 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + if (ret < 0) + return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + if (ret < 0) + return ret; break; default: dev_err(codec->dev, "Invalid DAI format\n"); return -EINVAL; } - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); - + if (ret < 0) + return ret; return 0; } @@ -263,8 +269,10 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - return snd_soc_update_bits(codec, CS4271_DACCTL, - CS4271_DACCTL_DEM_MASK, val); + i = snd_soc_update_bits(codec, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); + if (i < 0) + return i; + return 0; } static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, @@ -294,7 +302,8 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - unsigned int i, ratio, val; + int i; + unsigned int ratio, val; cs4271->rate = params_rate(params); ratio = cs4271->mclk / cs4271->rate; @@ -315,8 +324,10 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, else val |= cs4271_clk_tab[i].mclk_slave; - snd_soc_update_bits(codec, CS4271_MODE1, + i = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + if (i < 0) + return i; return cs4271_set_deemph(codec); } @@ -324,6 +335,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + int ret; int val_a = 0; int val_b = 0; @@ -332,8 +344,12 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); - snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + if (ret < 0) + return ret; return 0; } @@ -392,17 +408,25 @@ struct snd_soc_dai_driver cs4271_dai = { #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { + int ret; /* Set power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + if (ret < 0) + return ret; return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { + int ret; /* Restore codec state */ - snd_soc_cache_sync(codec); + ret = snd_soc_cache_sync(codec); + if (ret < 0) + return ret; /* then disable the power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; return 0; } #else @@ -467,9 +491,13 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_update_bits(codec, CS4271_MODE2, 0, + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; /* Power-up sequence requires 85 uS */ udelay(85);

2 1

[alsa-devel] [PATCH] ASoC: CS4271 codec support
by Alexander 21 Jan '11

21 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for CS4271 codec to ASoC. Update: Modified based on latest reviews. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- include/sound/cs4271.h | 25 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4271.c | 630 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 661 insertions(+), 0 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h new file mode 100644 index 0000000..16f8d32 --- /dev/null +++ b/include/sound/cs4271.h @@ -0,0 +1,25 @@ +/* + * Definitions for CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + */ + +#ifndef __CS4271_H +#define __CS4271_H + +struct cs4271_platform_data { + int gpio_nreset; /* GPIO driving Reset pin, if any */ + int gpio_disable; /* GPIO that disable serial bus, if any */ +}; + +#endif /* __CS4271_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c48b23c..c3056bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 @@ -155,6 +156,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CS4271 + tristate + config SND_SOC_CX20442 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 579af9c..ece05e1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o @@ -91,6 +92,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c new file mode 100644 index 0000000..25fed1b --- /dev/null +++ b/sound/soc/codecs/cs4271.c @@ -0,0 +1,630 @@ +/* + * CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/cs4271.h> + +#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * CS4271 registers + * High byte represents SPI chip address (0x10) + write command (0) + * Low byte - codec register address + */ +#define CS4271_MODE1 0x2001 /* Mode Control 1 */ +#define CS4271_DACCTL 0x2002 /* DAC Control */ +#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x2006 /* ADC Control */ +#define CS4271_MODE2 0x2007 /* Mode Control 2 */ +#define CS4271_CHIPID 0x2008 /* Chip ID */ + +#define CS4271_FIRSTREG CS4271_MODE1 +#define CS4271_LASTREG CS4271_MODE2 +#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1) + +/* Bit masks for the CS4271 registers */ +#define CS4271_MODE1_MODE_MASK 0xC0 +#define CS4271_MODE1_MODE_1X 0x00 +#define CS4271_MODE1_MODE_2X 0x80 +#define CS4271_MODE1_MODE_4X 0xC0 + +#define CS4271_MODE1_DIV_MASK 0x30 +#define CS4271_MODE1_DIV_1 0x00 +#define CS4271_MODE1_DIV_15 0x10 +#define CS4271_MODE1_DIV_2 0x20 +#define CS4271_MODE1_DIV_3 0x30 + +#define CS4271_MODE1_MASTER 0x08 + +#define CS4271_MODE1_DAC_DIF_MASK 0x07 +#define CS4271_MODE1_DAC_DIF_LJ 0x00 +#define CS4271_MODE1_DAC_DIF_I2S 0x01 +#define CS4271_MODE1_DAC_DIF_RJ16 0x02 +#define CS4271_MODE1_DAC_DIF_RJ24 0x03 +#define CS4271_MODE1_DAC_DIF_RJ20 0x04 +#define CS4271_MODE1_DAC_DIF_RJ18 0x05 + +#define CS4271_DACCTL_AMUTE 0x80 +#define CS4271_DACCTL_IF_SLOW 0x40 + +#define CS4271_DACCTL_DEM_MASK 0x30 +#define CS4271_DACCTL_DEM_DIS 0x00 +#define CS4271_DACCTL_DEM_441 0x10 +#define CS4271_DACCTL_DEM_48 0x20 +#define CS4271_DACCTL_DEM_32 0x30 + +#define CS4271_DACCTL_SVRU 0x08 +#define CS4271_DACCTL_SRD 0x04 +#define CS4271_DACCTL_INVA 0x02 +#define CS4271_DACCTL_INVB 0x01 + +#define CS4271_DACVOL_BEQUA 0x40 +#define CS4271_DACVOL_SOFT 0x20 +#define CS4271_DACVOL_ZEROC 0x10 + +#define CS4271_DACVOL_ATAPI_MASK 0x0F +#define CS4271_DACVOL_ATAPI_M_M 0x00 +#define CS4271_DACVOL_ATAPI_M_BR 0x01 +#define CS4271_DACVOL_ATAPI_M_BL 0x02 +#define CS4271_DACVOL_ATAPI_M_BLR2 0x03 +#define CS4271_DACVOL_ATAPI_AR_M 0x04 +#define CS4271_DACVOL_ATAPI_AR_BR 0x05 +#define CS4271_DACVOL_ATAPI_AR_BL 0x06 +#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07 +#define CS4271_DACVOL_ATAPI_AL_M 0x08 +#define CS4271_DACVOL_ATAPI_AL_BR 0x09 +#define CS4271_DACVOL_ATAPI_AL_BL 0x0A +#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B +#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C +#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D +#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E +#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F + +#define CS4271_VOLA_MUTE 0x80 +#define CS4271_VOLA_VOL_MASK 0x7F +#define CS4271_VOLB_MUTE 0x80 +#define CS4271_VOLB_VOL_MASK 0x7F + +#define CS4271_ADCCTL_DITHER16 0x20 + +#define CS4271_ADCCTL_ADC_DIF_MASK 0x10 +#define CS4271_ADCCTL_ADC_DIF_LJ 0x00 +#define CS4271_ADCCTL_ADC_DIF_I2S 0x10 + +#define CS4271_ADCCTL_MUTEA 0x08 +#define CS4271_ADCCTL_MUTEB 0x04 +#define CS4271_ADCCTL_HPFDA 0x02 +#define CS4271_ADCCTL_HPFDB 0x01 + +#define CS4271_MODE2_LOOP 0x10 +#define CS4271_MODE2_MUTECAEQUB 0x08 +#define CS4271_MODE2_FREEZE 0x04 +#define CS4271_MODE2_CPEN 0x02 +#define CS4271_MODE2_PDN 0x01 + +#define CS4271_CHIPID_PART_MASK 0xF0 +#define CS4271_CHIPID_REV_MASK 0x0F + +/* + * Default CS4271 power-up configuration + * Array contains non-existing in hw register at address 0 + * Array do not include Chip ID, as codec driver does not use + * registers read operations at all + */ +static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { + 0, + 0, + CS4271_DACCTL_AMUTE, + CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, + 0, + 0, + 0, + 0, +}; + +struct cs4271_private { + /* SND_SOC_I2C or SND_SOC_SPI */ + enum snd_soc_control_type bus_type; + void *control_data; + unsigned int mclk; + bool master; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; + /* GPIO that disable serial bus, if any */ + int gpio_disable; +}; + +struct cs4271_clk_cfg { + unsigned int ratio; /* MCLK / sample rate */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + u8 mclk_master; /* ratio bit mask for Master mode */ + u8 mclk_slave; /* ratio bit mask for Slave mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, + {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, + {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + +/* + * @freq is the desired MCLK rate + * MCLK rate should (c) be the sample rate, multiplied by one of the + * ratios listed in cs4271_mclk_fs_ratios table + */ +static int cs4271_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mclk = freq; + return 0; +} + +static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4271->master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4271->master = 1; + val |= CS4271_MODE1_MASTER; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val |= CS4271_MODE1_DAC_DIF_LJ; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + break; + case SND_SOC_DAIFMT_I2S: + val |= CS4271_MODE1_DAC_DIF_I2S; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); + + return 0; +} + +static int cs4271_deemph[] = {0, 44100, 48000, 32000}; + +static int cs4271_set_deemph(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int i; + int val = CS4271_DACCTL_DEM_DIS; + + if (cs4271->deemph) { + /* Find closest de-emphasis freq */ + val = 1; + for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++) + if (abs(cs4271_deemph[i] - cs4271->rate) < + abs(cs4271_deemph[val] - cs4271->rate)) + val = i; + val <<= 4; + } + + return snd_soc_update_bits(codec, CS4271_DACCTL, + CS4271_DACCTL_DEM_MASK, val); +} + +static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = cs4271->deemph; + return 0; +} + +static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->deemph = ucontrol->value.enumerated.item[0]; + return cs4271_set_deemph(codec); +} + +static int cs4271_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 cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int i, ratio, val; + + cs4271->rate = params_rate(params); + ratio = cs4271->mclk / cs4271->rate; + for (i = 0; i < CS4171_NR_RATIOS; i++) + if (cs4271_clk_tab[i].ratio == ratio) + break; + + if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + /* Configure DAC */ + val = cs4271_clk_tab[i].speed_mode; + + if (cs4271->master) + val |= cs4271_clk_tab[i].mclk_master; + else + val |= cs4271_clk_tab[i].mclk_slave; + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + + return cs4271_set_deemph(codec); +} + +static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int val_a = 0; + int val_b = 0; + + if (mute) { + val_a = CS4271_VOLA_MUTE; + val_b = CS4271_VOLB_MUTE; + } + + snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + return 0; +} + +/* CS4271 controls */ +static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0); + +static const struct snd_kcontrol_new cs4271_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB, + 0, 0x7F, 1, cs4271_dac_tlv), + SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + cs4271_get_deemph, cs4271_put_deemph), + SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0), + SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0), + SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0), + SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0), + SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1), + SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0), + SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1), + SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB, + 7, 1, 1), +}; + +static struct snd_soc_dai_ops cs4271_dai_ops = { + .hw_params = cs4271_hw_params, + .set_sysclk = cs4271_set_dai_sysclk, + .set_fmt = cs4271_set_dai_fmt, + .digital_mute = cs4271_digital_mute, +}; + +struct snd_soc_dai_driver cs4271_dai = { + .name = "cs4271-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .ops = &cs4271_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + /* Set power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + return 0; +} + +static int cs4271_soc_resume(struct snd_soc_codec *codec) +{ + /* Restore codec state */ + snd_soc_cache_sync(codec); + /* then disable the power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + return 0; +} +#else +#define cs4271_soc_suspend NULL +#define cs4271_soc_resume NULL +#endif /* CONFIG_PM */ + +static int cs4271_probe(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; + int ret; + int gpio_nreset = -EINVAL; + int gpio_disable = -EINVAL; + + codec->control_data = cs4271->control_data; + + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + if (gpio_is_valid(cs4271plat->gpio_disable)) + gpio_disable = cs4271plat->gpio_disable; + } + + if (gpio_disable >= 0) + if (gpio_request(gpio_disable, "CS4271 Disable")) + gpio_disable = -EINVAL; + if (gpio_disable >= 0) + gpio_direction_output(gpio_disable, 0); + + if (gpio_nreset >= 0) + if (gpio_request(gpio_nreset, "CS4271 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + } + + cs4271->gpio_nreset = gpio_nreset; + cs4271->gpio_disable = gpio_disable; + + /* + * In case of I2C, chip address specified in board data. + * So cache IO operations use 8 bit codec register address. + * In case of SPI, chip address and register address + * passed together as 16 bit value. + * Anyway, register address is masked with 0xFF inside + * soc-cache code. + */ + if (cs4271->bus_type == SND_SOC_SPI) + ret = snd_soc_codec_set_cache_io(codec, 16, 8, + cs4271->bus_type); + else + ret = snd_soc_codec_set_cache_io(codec, 8, 8, + cs4271->bus_type); + if (ret) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + snd_soc_update_bits(codec, CS4271_MODE2, 0, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + /* Power-up sequence requires 85 uS */ + udelay(85); + + return snd_soc_add_controls(codec, cs4271_snd_controls, + ARRAY_SIZE(cs4271_snd_controls)); +} + +static int cs4271_remove(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int gpio_nreset, gpio_disable; + + gpio_nreset = cs4271->gpio_nreset; + gpio_disable = cs4271->gpio_disable; + + if (gpio_is_valid(gpio_nreset)) { + /* Set codec to the reset state */ + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + } + + if (gpio_is_valid(gpio_disable)) + gpio_free(gpio_disable); + + return 0; +}; + +struct snd_soc_codec_driver soc_codec_dev_cs4271 = { + .probe = cs4271_probe, + .remove = cs4271_remove, + .suspend = cs4271_soc_suspend, + .resume = cs4271_soc_resume, + .reg_cache_default = cs4271_dflt_reg, + .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), + .reg_word_size = sizeof(cs4271_dflt_reg[0]), + .compress_type = SND_SOC_FLAT_COMPRESSION, +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + spi_set_drvdata(spi, cs4271); + cs4271->control_data = spi; + cs4271->bus_type = SND_SOC_SPI; + + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .probe = cs4271_spi_probe, + .remove = __devexit_p(cs4271_spi_remove), +}; +#endif /* defined(CONFIG_SPI_MASTER) */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_device_id cs4271_i2c_id[] = { + {"cs4271", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static int __devinit cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + i2c_set_clientdata(client, cs4271); + cs4271->control_data = client; + cs4271->bus_type = SND_SOC_I2C; + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .id_table = cs4271_i2c_id, + .probe = cs4271_i2c_probe, + .remove = __devexit_p(cs4271_i2c_remove), +}; +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +/* + * We only register our serial bus driver here without + * assignment to particular chip. So if any of the below + * fails, there is some problem with I2C or SPI subsystem. + * In most cases this module will be compiled with support + * of only one serial bus. + */ +static int __init cs4271_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&cs4271_i2c_driver); + if (ret) { + pr_err("Failed to register CS4271 I2C driver: %d\n", ret); + return ret; + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4271_spi_driver); + if (ret) { + pr_err("Failed to register CS4271 SPI driver: %d\n", ret); + return ret; + } +#endif + + return 0; +} +module_init(cs4271_modinit); + +static void __exit cs4271_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4271_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&cs4271_i2c_driver); +#endif +} +module_exit(cs4271_modexit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL");

4 3

Re: [alsa-devel] [PATCH] davinci: da8xx/omap-l1xx: match codec_name with i2c ids
by Mark Brown 21 Jan '11

21 Jan '11

On Fri, Jan 21, 2011 at 08:10:01PM +0530, Rajashekhara, Sudhakar wrote: > The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c > is not matching with the i2c ids in the board file. Without this fix the > soundcard does not get detected on da850/omap-l138/am18x evm. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: Handle low measured DC offsets for wm_hubs devices
by Mark Brown 21 Jan '11

21 Jan '11

The DC servo codes are actually signed numbers so need to be treated as such. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 15 +++++++-------- 1 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c466982..613df5d 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + s8 offset; u16 reg, reg_l, reg_r, dcs_cfg; /* If we're using a digital only path and have a previously @@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes); /* HPOUT1L */ - if (reg_l + hubs->dcs_codes > 0 && - reg_l + hubs->dcs_codes < 0xff) - reg_l += hubs->dcs_codes; - dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + offset = reg_l; + offset += hubs->dcs_codes; + dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - if (reg_r + hubs->dcs_codes > 0 && - reg_r + hubs->dcs_codes < 0xff) - reg_r += hubs->dcs_codes; - dcs_cfg |= reg_r; + offset = reg_r; + offset += hubs->dcs_codes; + dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Staticise twl6040_hs_jack_report()
by Mark Brown 21 Jan '11

21 Jan '11

It's an internal function so shouldn't be exported (as sparse points out). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4bbf1b1..482fcdb 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -724,8 +724,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } -void twl6040_hs_jack_report(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, int report) +static void twl6040_hs_jack_report(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int status; -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: soc-cache: Add trace event for snd_soc_cache_sync()
by Dimitris Papastamos 21 Jan '11

21 Jan '11

This patch makes it easy to see when the syncing process begins and ends. You can also enable the snd_soc_reg_write tracepoint to see which registers are being synced. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/trace/events/asoc.h | 25 +++++++++++++++++++++++++ sound/soc/soc-cache.c | 10 ++++++++++ 2 files changed, 35 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 186e84d..ce993a3 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -229,6 +229,31 @@ TRACE_EVENT(snd_soc_jack_notify, TP_printk("jack=%s %x", __get_str(name), (int)__entry->val) ); +TRACE_EVENT(snd_soc_cache_sync, + + TP_PROTO(struct snd_soc_codec *codec, const char *type, + const char *status), + + TP_ARGS(codec, type, status), + + TP_STRUCT__entry( + __string( name, codec->name ) + __string( status, status ) + __string( type, type ) + __field( int, id ) + ), + + TP_fast_assign( + __assign_str(name, codec->name); + __assign_str(status, status); + __assign_str(type, type); + __entry->id = codec->id + ), + + TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), + (int)__entry->id, __get_str(type), __get_str(status)) +); + #endif /* _TRACE_ASOC_H */ /* This part must be outside protection */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f834839..db66dc4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include <linux/bitmap.h> #include <linux/rbtree.h> +#include <trace/events/asoc.h> + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -1601,18 +1603,26 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); int snd_soc_cache_sync(struct snd_soc_codec *codec) { int ret; + const char *name; if (!codec->cache_sync) { return 0; } + if (codec->cache_ops->name) + name = codec->cache_ops->name; + else + name = "unknown"; + if (codec->cache_ops && codec->cache_ops->sync) { if (codec->cache_ops->name) dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); ret = codec->cache_ops->sync(codec); if (!ret) codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); return ret; } -- 1.7.3.5

1 0

[alsa-devel] [PATCH] ASoC: soc-cache: Add trace event for snd_soc_cache_sync()
by Dimitris Papastamos 21 Jan '11

21 Jan '11

This patch makes it easier to see which of the register writes are part of the cache syncing functionality. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/trace/events/asoc.h | 9 +++++++++ sound/soc/soc-cache.c | 5 +++++ 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 186e84d..7efed81 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -59,6 +59,15 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read, ); +DEFINE_EVENT(snd_soc_reg, snd_soc_cache_sync, + + TP_PROTO(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val), + + TP_ARGS(codec, reg, val) + +); + DECLARE_EVENT_CLASS(snd_soc_card, TP_PROTO(struct snd_soc_card *card, int val), diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f834839..accd96e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include <linux/bitmap.h> #include <linux/rbtree.h> +#include <trace/events/asoc.h> + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -895,6 +897,7 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) codec->cache_bypass = 0; if (ret) return ret; + trace_snd_soc_cache_sync(codec, rbnode->reg, val); dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", rbnode->reg, val); } @@ -1155,6 +1158,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) codec->cache_bypass = 0; if (ret) return ret; + trace_snd_soc_cache_sync(codec, i, val); dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", i, val); } @@ -1415,6 +1419,7 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_write(codec, i, val); if (ret) return ret; + trace_snd_soc_cache_sync(codec, i, val); dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", i, val); } -- 1.7.3.5

2 2

Re: [alsa-devel] [QUERY] [AUDIO][OMAP3 EVM] arecord not working on OMAP3 EVM
by Peter Ujfalusi 21 Jan '11

21 Jan '11

On 01/20/11 17:26, ext Koyamangalath, Abhilash wrote: > Hi all > While testing audio capture (using arecord e,g.),I'm getting only noise in the (zeroed out) output buffers on the OMAP3 EVM. > I have enabled AUXL and AUXR mic inputs through alsamixer, which according to the schematics are connected to the "Audio In" > connector on the board, even though the TWL4030 reg list does not allow me any ways ( or so I think) to specify mic-bias for > these aux inputs. Well, it is auxiliary input (line in). You do not need bias for it. I do not have access to OMAP3 EVM documentation, but Beagle also have AUXL/R as input. I guess you need to use correct cable, and correct source for the input (stereo jack I suppose). > With AUXL and AUXR muted - I get only silence, which seems to make sense. > Is there any basic startup setting I'm missing, maybe specific to OMAP3 EVM? Beagle? > Has anyone seen anything like this before, and could provide pointers? My bet goes for the wrong cable/accessory pair. -- Péter -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH 1/2] ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx
by David Henningsson 21 Jan '11

21 Jan '11

Four very similar procedures - one for each model - now refactored into one. This isn't all duplicated code, but a step in the right direction. (And given the next patch, that four procedures would have been five if it weren't for this one...) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

[PATCH 2/2] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
by David Henningsson 21 Jan '11

21 Jan '11

BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 24 +++++++++++++++++++++++- 2 files changed, 24 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 16ae430..0caf77e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5066 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60 + asus Asus K52JU, Lenovo G560 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7cd59b9..19f0daf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -127,6 +127,7 @@ struct conexant_spec { unsigned int ideapad:1; unsigned int thinkpad:1; unsigned int hp_laptop:1; + unsigned int asus:1; unsigned int ext_mic_present; unsigned int recording; @@ -2312,6 +2313,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_asus_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 0); +} + + /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_hp_laptop_automic(struct hda_codec *codec) { @@ -2400,6 +2414,8 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); + else if (spec->asus) + cxt5066_asus_automic(codec); } /* unsolicited event for jack sensing */ @@ -3045,6 +3061,7 @@ enum { CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3056,6 +3073,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", }; @@ -3068,6 +3086,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3077,6 +3096,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3132,13 +3152,15 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_ASUS: case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; - spec->hp_laptop = 1; + spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; + spec->asus = board_config == CXT5066_ASUS; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ -- 1.7.1 --------------050308060108010003060101--

1 0

[PATCH 1/2] ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx
by David Henningsson 21 Jan '11

21 Jan '11

Four very similar procedures - one for each model - now refactored into one. This isn't all duplicated code, but a step in the right direction. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 81 +++++++++++---------------------------- 1 files changed, 23 insertions(+), 58 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9bb030a..7cd59b9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2387,79 +2387,53 @@ static void cxt5066_hp_automute(struct hda_codec *codec) cxt5066_update_speaker(codec); } -/* unsolicited event for jack sensing */ -static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) +/* Dispatch the right mic autoswitch function */ +static void cxt5066_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - /* ignore mic events in DC mode; we're always using the jack */ - if (!spec->dc_enable) - cxt5066_olpc_automic(codec); - break; - } -} -/* unsolicited event for jack sensing */ -static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + if (spec->dell_vostro) cxt5066_vostro_automic(codec); - break; - } -} - -/* unsolicited event for jack sensing */ -static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + else if (spec->ideapad) cxt5066_ideapad_automic(codec); - break; - } + else if (spec->thinkpad) + cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } /* unsolicited event for jack sensing */ -static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + struct conexant_spec *spec = codec->spec; + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_hp_laptop_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } /* unsolicited event for jack sensing */ -static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_thinkpad_automic(codec); + cxt5066_automic(codec); break; } } + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -3039,20 +3013,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: init\n"); conexant_init(codec); if (codec->patch_ops.unsol_event) { cxt5066_hp_automute(codec); - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); + cxt5066_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3169,7 +3134,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; @@ -3207,7 +3172,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_DELL_VOSTRO: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_vostro_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; @@ -3224,7 +3189,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_IDEAPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_ideapad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_ideapad; @@ -3240,7 +3205,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_THINKPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_thinkpad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; -- 1.7.1 --------------090909050907040908000909--

1 0

Re: [alsa-devel] [PATCH - [hdspm] 2/3] Enable RME RayDAT and AIO
by Takashi Iwai 21 Jan '11

21 Jan '11

At Tue, 18 Jan 2011 15:12:18 +0100, Adrian Knoth wrote: > > @@ -29,12 +42,13 @@ > #include <linux/moduleparam.h> > #include <linux/slab.h> > #include <linux/pci.h> > -#include <linux/math64.h> No math64 stuff is necessary? > +#include <linux/sysfs.h> It's not implemented yet, so don't include it. > -static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { > +static struct pci_device_id snd_hdspm_ids[] __devinitdata = { Why open-coded again? > @@ -796,6 +1009,8 @@ static void hdspm_silence_playback(struct hdspm *hdspm) > int n = hdspm->period_bytes; > void *buf = hdspm->playback_buffer; > > + snd_printk(KERN_INFO "hdspm_silence_playback\n"); Don't put such a verbose debug message in frequently called callbacks. > static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) ... > n = div_u64(n, rate); > /* n should be less than 2^32 for being written to FREQ register */ > - snd_BUG_ON(n >> 32); > + snd_BUG_ON((n >> 32) == 0); This check is consistent with the comment above... The rest is too long to review. Would be nice if you can split the whole patch in logical orders... thanks, Takashi

2 2

[alsa-devel] Question about ASOC codec drivers
by Neil Jones 20 Jan '11

20 Jan '11

Hi, Im trying to write an ASOC Codec driver for a new codec and im having problems setting up the widgets and interconnects. The input structure is similar to the wm8731 so i have based my codec on this (my widget code is included at the bottom) problem is I get this error on startup: Failed to add route Line Input->Input Mux Add then when I run alsa-mixer I get a memfault in the strcpy in snd_soc_info_enum_double() it appears to be indexing a string 1 past the end, ie uinfo->value.enumerated.item = 3 but e->texts is pointing to the tansen_input_select[] array of strings which only has 3 entries, so it blows up (e->max = 7??). also shouldn't the strcpy by a strncpy as we are copying to a fixed 64byte buffer ? thanks for anyhelp. Neil my widget code: static const struct snd_kcontrol_new tansen_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TANSEN_TM_OP_TOP7, TANSEN_TM_OP_TOP8, 0, 0xFF, 0 ), SOC_DOUBLE("Line/IPOD-In Gain", TANSEN_TM_IP_TOP5, 5, 2, 0x7, 0), SOC_DOUBLE("Mic-In Gain", TANSEN_TM_IP_TOP4, 0, 4, 0xF, 0), }; static const char *tansen_input_select[] = { "Mic In", "Line In", "IPOD In", }; static const struct soc_enum tansen_ip_enum = SOC_ENUM_SINGLE(TANSEN_TM_IP_TOP2, 4, 0x7, tansen_input_select); /* Input mux */ static const struct snd_kcontrol_new tansen_input_mux_controls = SOC_DAPM_ENUM("Input Select", tansen_ip_enum); static const struct snd_soc_dapm_widget tansen_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &tansen_input_mux_controls), SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", TANSEN_TM_MICBIAS_1, 4, 1), SND_SOC_DAPM_INPUT("MICIN"), SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("LLINEIN"), SND_SOC_DAPM_INPUT("RIPODIN"), SND_SOC_DAPM_INPUT("LIPODIN") }; static const struct snd_soc_dapm_route intercon[] = { /* outputs */ {"RHPOUT", NULL, "DAC"}, {"LHPOUT", NULL, "DAC"}, /* input mux */ {"Input Mux", "Line In", "Line Input"}, {"Input Mux", "Mic In", "Mic bias"}, {"Input Mux", "Ipod In", "Ipod Input"}, {"ADC", NULL, "PGA"}, {"PGA", NULL, "Input Mux"}, /* inputs */ {"Line Input", NULL, "LLINEIN"}, {"Line Input", NULL, "RLINEIN"}, {"Ipod Input", NULL, "LIPODIN"}, {"Ipod Input", NULL, "RIPODIN"}, {"Mic bias", NULL, "MICIN"}, };

2 3

[alsa-devel] Automatically disabling internal speakers when headphones are plugged in
by Stephen Warren 20 Jan '11

20 Jan '11

I'm working on jack plug detect for Tegra Harmony, and have a question about automatically disabling speakers when a headphone jack is plugged in. For test purposes, I implemented this as follows: static struct snd_soc_jack_pin harmony_hp_jack_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, }, { .pin = "Int Spk", .mask = SND_JACK_HEADPHONE, .invert = 1, }, }; ... following the example of: omap/ams-delta.c samsung/rx1950_uda1380.c samsung/h1940_uda1380.c However, I wonder if that's the correct approach; should the plug detect logic only control the headphone output (i.e. only include the first array entry in jack_pins[] above), and leave it up to user-space whether to disable the speaker in this case, using a custom Speaker Function control created by the machine driver, as I see many other drivers expose? Do the jack reports make it up to user-space so that could be implemented? Thanks. -- nvpublic

2 1

[alsa-devel] [alsa-utils PATCH 1/3] Add missing call to AM_PROG_CC_C_O.
by Diego Elio Pettenò 20 Jan '11

20 Jan '11

Resolves this warning when rebuilding autotools: alsactl/Makefile.am:10: compiling `alsactl.c' with per-target flags requires `AM_PROG_CC_C_O' in `configure.in' --- configure.in | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/configure.in b/configure.in index 561ca46..6892fbf 100644 --- a/configure.in +++ b/configure.in @@ -24,6 +24,7 @@ then fi AC_PROG_CC +AM_PROG_CC_C_O dnl AC_PROG_CXX AC_PROG_INSTALL AC_PROG_MKDIR_P -- 1.7.4.rc2

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[alsa-devel] [alsa-lib PATCH] Update attributes.m4 to avoid warnings with autoconf-2.68
by Diego Elio Pettenò 20 Jan '11

20 Jan '11

--- m4/attributes.m4 | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/m4/attributes.m4 b/m4/attributes.m4 index e86456a..82b4a3c 100644 --- a/m4/attributes.m4 +++ b/m4/attributes.m4 @@ -25,7 +25,7 @@ dnl License when using or distributing such scripts, even though portions dnl of the text of the Macro appear in them. The GNU General Public dnl License (GPL) does govern all other use of the material that dnl constitutes the Autoconf Macro. -dnl +dnl dnl This special exception to the GPL applies to versions of the dnl Autoconf Macro released by this project. When you make and dnl distribute a modified version of the Autoconf Macro, you may extend @@ -39,7 +39,7 @@ AC_DEFUN([CC_CHECK_CFLAGS_SILENT], [ AC_CACHE_VAL(AS_TR_SH([cc_cv_cflags_$1]), [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $1" - AC_COMPILE_IFELSE([int a;], + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 0; }])], [eval "AS_TR_SH([cc_cv_cflags_$1])='yes'"], [eval "AS_TR_SH([cc_cv_cflags_$1])='no'"]) CFLAGS="$ac_save_CFLAGS" @@ -89,7 +89,7 @@ AC_DEFUN([CC_CHECK_LDFLAGS], [ AS_TR_SH([cc_cv_ldflags_$1]), [ac_save_LDFLAGS="$LDFLAGS" LDFLAGS="$LDFLAGS $1" - AC_LINK_IFELSE([int main() { return 1; }], + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 1; }])], [eval "AS_TR_SH([cc_cv_ldflags_$1])='yes'"], [eval "AS_TR_SH([cc_cv_ldflags_$1])="]) LDFLAGS="$ac_save_LDFLAGS" @@ -147,7 +147,7 @@ AC_DEFUN([CC_CHECK_ATTRIBUTE], [ AS_TR_SH([cc_cv_attribute_$1]), [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $cc_cv_werror" - AC_COMPILE_IFELSE([$3], + AC_COMPILE_IFELSE([AC_LANG_SOURCE([$3])], [eval "AS_TR_SH([cc_cv_attribute_$1])='yes'"], [eval "AS_TR_SH([cc_cv_attribute_$1])='no'"]) CFLAGS="$ac_save_CFLAGS" @@ -165,7 +165,16 @@ AC_DEFUN([CC_CHECK_ATTRIBUTE], [ AC_DEFUN([CC_ATTRIBUTE_CONSTRUCTOR], [ CC_CHECK_ATTRIBUTE( [constructor],, - [void __attribute__((constructor)) ctor() { int a; }], + [extern void foo(); + void __attribute__((constructor)) ctor() { foo(); }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_DESTRUCTOR], [ + CC_CHECK_ATTRIBUTE( + [destructor],, + [extern void foo(); + void __attribute__((destructor)) dtor() { foo(); }], [$1], [$2]) ]) @@ -257,7 +266,7 @@ AC_DEFUN([CC_FLAG_VISIBILITY], [ cc_cv_flag_visibility='yes', cc_cv_flag_visibility='no') CFLAGS="$cc_flag_visibility_save_CFLAGS"]) - + AS_IF([test "x$cc_cv_flag_visibility" = "xyes"], [AC_DEFINE([SUPPORT_FLAG_VISIBILITY], 1, [Define this if the compiler supports the -fvisibility flag]) -- 1.7.4.rc2

1 0

[alsa-devel] [PATCH 1/4] Update attributes.m4 to avoid warnings with autoconf-2.68
by Diego Elio Pettenò 20 Jan '11

20 Jan '11

--- m4/attributes.m4 | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/m4/attributes.m4 b/m4/attributes.m4 index e86456a..82b4a3c 100644 --- a/m4/attributes.m4 +++ b/m4/attributes.m4 @@ -25,7 +25,7 @@ dnl License when using or distributing such scripts, even though portions dnl of the text of the Macro appear in them. The GNU General Public dnl License (GPL) does govern all other use of the material that dnl constitutes the Autoconf Macro. -dnl +dnl dnl This special exception to the GPL applies to versions of the dnl Autoconf Macro released by this project. When you make and dnl distribute a modified version of the Autoconf Macro, you may extend @@ -39,7 +39,7 @@ AC_DEFUN([CC_CHECK_CFLAGS_SILENT], [ AC_CACHE_VAL(AS_TR_SH([cc_cv_cflags_$1]), [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $1" - AC_COMPILE_IFELSE([int a;], + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 0; }])], [eval "AS_TR_SH([cc_cv_cflags_$1])='yes'"], [eval "AS_TR_SH([cc_cv_cflags_$1])='no'"]) CFLAGS="$ac_save_CFLAGS" @@ -89,7 +89,7 @@ AC_DEFUN([CC_CHECK_LDFLAGS], [ AS_TR_SH([cc_cv_ldflags_$1]), [ac_save_LDFLAGS="$LDFLAGS" LDFLAGS="$LDFLAGS $1" - AC_LINK_IFELSE([int main() { return 1; }], + AC_LINK_IFELSE([AC_LANG_SOURCE([int main() { return 1; }])], [eval "AS_TR_SH([cc_cv_ldflags_$1])='yes'"], [eval "AS_TR_SH([cc_cv_ldflags_$1])="]) LDFLAGS="$ac_save_LDFLAGS" @@ -147,7 +147,7 @@ AC_DEFUN([CC_CHECK_ATTRIBUTE], [ AS_TR_SH([cc_cv_attribute_$1]), [ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS $cc_cv_werror" - AC_COMPILE_IFELSE([$3], + AC_COMPILE_IFELSE([AC_LANG_SOURCE([$3])], [eval "AS_TR_SH([cc_cv_attribute_$1])='yes'"], [eval "AS_TR_SH([cc_cv_attribute_$1])='no'"]) CFLAGS="$ac_save_CFLAGS" @@ -165,7 +165,16 @@ AC_DEFUN([CC_CHECK_ATTRIBUTE], [ AC_DEFUN([CC_ATTRIBUTE_CONSTRUCTOR], [ CC_CHECK_ATTRIBUTE( [constructor],, - [void __attribute__((constructor)) ctor() { int a; }], + [extern void foo(); + void __attribute__((constructor)) ctor() { foo(); }], + [$1], [$2]) +]) + +AC_DEFUN([CC_ATTRIBUTE_DESTRUCTOR], [ + CC_CHECK_ATTRIBUTE( + [destructor],, + [extern void foo(); + void __attribute__((destructor)) dtor() { foo(); }], [$1], [$2]) ]) @@ -257,7 +266,7 @@ AC_DEFUN([CC_FLAG_VISIBILITY], [ cc_cv_flag_visibility='yes', cc_cv_flag_visibility='no') CFLAGS="$cc_flag_visibility_save_CFLAGS"]) - + AS_IF([test "x$cc_cv_flag_visibility" = "xyes"], [AC_DEFINE([SUPPORT_FLAG_VISIBILITY], 1, [Define this if the compiler supports the -fvisibility flag]) -- 1.7.4.rc2

1 4

[alsa-devel] [PATCH 1/2] ASoC: soc-cache: Introduce the cache_bypass option
by Dimitris Papastamos 20 Jan '11

20 Jan '11

This is primarily needed to avoid writing back to the cache whenever we are syncing the cache with the hardware. This gives a performance benefit especially for large register maps. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 36 ++++++++++++++++++++++++------------ 2 files changed, 25 insertions(+), 12 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9952254..984dfd1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -488,6 +488,7 @@ struct snd_soc_codec { /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; + unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int cache_only:1; /* Suppress writes to hardware */ unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ unsigned int suspended:1; /* Codec is in suspend PM state */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d97a59f..1ebff9f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -25,7 +25,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -49,7 +50,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -106,7 +108,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -130,7 +133,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -191,7 +195,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -216,7 +221,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -271,7 +277,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -295,7 +302,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -450,7 +458,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -476,7 +485,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -568,7 +578,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -595,7 +606,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[3] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; -- 1.7.3.5

3 3

[alsa-devel] [PATCH v5 0/3] input: Add support for ECI (multimedia) accessories
by tapio.vihuri＠nokia.com 20 Jan '11

20 Jan '11

From: Tapio Vihuri <tapio.vihuri(a)nokia.com> Thank you for comments. ECI input driver changes: - Use sparse_keymap_setup() properly, remove unneeded input set_bits ECI controller driver changes: - Added "button" debugfs entry for input driver buttons handling testing Created against linux-2.6.38-rc1 at: 12fcdba Merge branch 'x86-fixes-for-linus' of git://git.kernel.org/pub/scm/linux --- v4 message --- Thank you for comments. ECI input driver changes: - Use driver name in debugfs - reject accessory stating more than maximum 31 buttons - clean out couple comments ECI controller driver changes: - name change, from ecibus to eci_xxx_ctrl - location change, from drivers/ecibus to drivers/input/misc - major rework from platform driver to i2c-driver - removed mrst platform dependency Checkpatch warn about Kconfig, but I think it is false positive, and I will report it to the maintainer. Platform code changes: - medfield_ prefix changed to mrst_ - added ECI controller needed power control - remover ECI controller platform driver stuff - added i2c stuff for ECI controller driver Created against linux-2.6.37-rc8 at: a25cecc Merge branch 'next' of git://git.monstr.eu/linux-2.6-microblaze --- v3 message --- Thank you for comments. ECI input driver changes: - Makefile and Kconfig sorted alphabetically - using sparse keymap library, much cleaner solution - correct string aligment with 80 column - debugfs initialization fix - sysfs cable entry using just 0/1 - using cheaper __set_bit - insane comments fixed - remove unneeded misc device Created against linux-2.6.37-rc8 at: 989d873 Merge master.kernel.org:/home/rmk/linux-2.6-arm --- v2 message --- Thank you for comments. In series two there Kconfig dependency problem fixed, preventing compilation if X86_MRST (Intel Moorestown) or INPUT_MISC is not selected. Also created against linux-2.6.37-rc8 --- v1 message --- Hi all This patch set introduce Multimedia Headset Accessory support for Nokia phones. Technically those are known as ECI (Enhancement Control Interface) If headset has many buttons, like play, vol+, vol- etc. then it is propably ECI accessory. Among several buttons ECI accessories contains memory for storing several parameters. This ECI input driver provides the following features: - reading ECI configuration memory - ECI buttons as input events Drive is constructed as follows: - ECI accessory input driver deals with headset accessory - ECI bus control driver deals the HW transfering data to/from headset - platform data match used HW In the future accessory detection logic will be added using ALSA jack reporting. Created against linux-2.6.37-rc6 Please review. Tapio Vihuri (3): ECI: input: introduce ECI accessory input driver ECI: introducing ECI controller driver ECI: adding platform data for ECI driver arch/x86/platform/mrst/mrst.c | 49 ++ drivers/input/misc/Kconfig | 60 +++ drivers/input/misc/Makefile | 3 +- drivers/input/misc/eci.c | 931 ++++++++++++++++++++++++++++++++++++ drivers/input/misc/eci_at20_ctrl.c | 569 ++++++++++++++++++++++ include/linux/input/eci.h | 179 +++++++ 6 files changed, 1790 insertions(+), 1 deletions(-) create mode 100644 drivers/input/misc/eci.c create mode 100644 drivers/input/misc/eci_at20_ctrl.c create mode 100644 include/linux/input/eci.h

1 3

[alsa-devel] [PATCH v4 0/3] input: Add support for ECI (multimedia) accessories
by tapio.vihuri＠nokia.com 20 Jan '11

20 Jan '11

From: Tapio Vihuri <tapio.vihuri(a)nokia.com> Thank you for comments. ECI input driver changes: - Use driver name in debugfs - reject accessory stating more than maximum 31 buttons - clean out couple comments ECI controller driver changes: - name change, from ecibus to eci_xxx_ctrl - location change, from drivers/ecibus to drivers/input/misc - major rework from platform driver to i2c-driver - removed mrst platform dependency Checkpatch warn about Kconfig, but I think it is false positive, and I will report it to the maintainer. Platform code changes: - medfield_ prefix changed to mrst_ - added ECI controller needed power control - remover ECI controller platform driver stuff - added i2c stuff for ECI controller driver Created against linux-2.6.37-rc8 at: a25cecc Merge branch 'next' of git://git.monstr.eu/linux-2.6-microblaze --- v3 message --- Thank you for comments. ECI input driver changes: - Makefile and Kconfig sorted alphabetically - using sparse keymap library, much cleaner solution - correct string aligment with 80 column - debugfs initialization fix - sysfs cable entry using just 0/1 - using cheaper __set_bit - insane comments fixed - remove unneeded misc device Created against linux-2.6.37-rc8 at: 989d873 Merge master.kernel.org:/home/rmk/linux-2.6-arm --- v2 message --- Thank you for comments. In series two there Kconfig dependency problem fixed, preventing compilation if X86_MRST (Intel Moorestown) or INPUT_MISC is not selected. Also created against linux-2.6.37-rc8 --- v1 message --- Hi all This patch set introduce Multimedia Headset Accessory support for Nokia phones. Technically those are known as ECI (Enhancement Control Interface) If headset has many buttons, like play, vol+, vol- etc. then it is propably ECI accessory. Among several buttons ECI accessories contains memory for storing several parameters. This ECI input driver provides the following features: - reading ECI configuration memory - ECI buttons as input events Drive is constructed as follows: - ECI accessory input driver deals with headset accessory - ECI bus control driver deals the HW transfering data to/from headset - platform data match used HW In the future accessory detection logic will be added using ALSA jack reporting. Created against linux-2.6.37-rc6 Please review. Tapio Vihuri (3): ECI: input: introduce ECI accessory input driver ECI: introducing ECI controller driver ECI: adding platform data for ECI driver arch/x86/platform/mrst/mrst.c | 49 ++ drivers/input/misc/Kconfig | 60 +++ drivers/input/misc/Makefile | 3 +- drivers/input/misc/eci.c | 946 ++++++++++++++++++++++++++++++++++++ drivers/input/misc/eci_at20_ctrl.c | 533 ++++++++++++++++++++ include/linux/input/eci.h | 179 +++++++ 6 files changed, 1769 insertions(+), 1 deletions(-) create mode 100644 drivers/input/misc/eci.c create mode 100644 drivers/input/misc/eci_at20_ctrl.c create mode 100644 include/linux/input/eci.h

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[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 20 Jan '11

20 Jan '11

The following changes since commit c56eb8fb6dccb83d9fe62fd4dc00c834de9bc470: Linux 2.6.38-rc1 (2011-01-18 15:14:02 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Barry Song (2): ASoC: Blackfin TDM: use external frame syncs ASoC: Blackfin: fix DAI/SPORT config dependency issues Dimitris Papastamos (11): ASoC: soc-core: Add support for NULL default register caches ASoC: WM8995: Remember to flush the cache on resume ASoC: soc-cache: Clean up the cache manipulation code ASoC: soc-core: Simplify compress_type overriding functionality ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache ASoC: soc-cache: Add reg_size as a member to snd_soc_codec ASoC: soc-cache: Add support for default readable()/volatile() functions ASoC: Update users of readable_register()/volatile_register() ASoC: Automatically assign the default readable()/volatile() functions ASoC: WM8991: Add initial WM8991 driver ASoC: WM8995: Add regulator handling code Harsha Priya (2): ASoC: sst_platform created helper functions ASoC: sst_platform porting sst dsp driver interface as per latest in Greg's staging tree Kuninori Morimoto (1): ASoC: sh: fsi-ak4642: fixup platform device id Mark Brown (8): ASoC: Taint the kernel if debugfs is used to write directly to CODECs ASoC: Update name of debugfs root symbol to snd_soc_ ASoC: Fix indentation in soc_remove() Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Avoid direct register cache access when setting defaults Merge branch 'for-2.6.38' into for-2.6.39 Merge branch 'for-2.6.38' of git://git.kernel.org/.../broonie/sound-2.6 into for-2.6.38 Merge branch 'for-2.6.38' into for-2.6.39 Mike Frysinger (2): ASoC: Blackfin TDM: fix missed snd_soc_dai_get_drvdata update ASoC: Blackfin AC97: fix build error after multi-component update Peter Ujfalusi (1): ASoC: core: Remove dapm_sync call from soc_post_component_init Seungwhan Youn (6): ASoC: SAMSUNG: Remove AC97 header file ASoC: SAMSUNG: Clean-up DMA header file ASoC: SAMSUNG: Move PCM specific definitions into pcm.c ASoC: SAMSUNG: Clean-up header includes ASoC: documentation updates ASoC: documentation updates Stephen Warren (10): ASoC: Export debugfs root dentry ASoC: tegra: Add tegra-das driver ASoC: tegra: Add tegra-pcm driver ASoC: tegra: Add tegra-i2s driver ASoC: tegra: Machine utility code ASoC: tegra: Harmony machine support ASoC: tegra: Kconfig and Makefile ASoC: tegra: s/IS_ERR_OR_NULL/IS_ERR/ for clk_get_sys ASoC: tegra: Remove TEGRA_I2S_AUDIO from Kconfig ASoC: tegra: Add DAPM widgets/routes for Harmony Timur Tabi (1): ASoC: let snd_soc_update_bits() return an error code Vasily Khoruzhick (2): ASoC: PXA: Fix jack detection on Zipit Z2 ASoC: PXA: Fix codec address on Zipit Z2 Vinod Koul (12): ASoC: sst v2: Add sn95031 codec driver ASoC: sst v2: Add mid platform driver ASoC sst v2: Add medfield machine driver ASoC: sst v2: Add makefiles and kconfigs changes ASoC: sn95031 fix the code style and format inconsistencies ASoC: sst platform - fix the style inconsistency ASoC: mid-x86 - remove the flag in makefile ASoC: soc core move card cleanup from soc_remove() ASoC: soc core move the card debugfs initialization ASoC: soc core add inline to handle card list initialzation ASoC: soc core allow machine driver to register the card ASoC: Add dependency on INTEL_SCU_IPC for Intel MID drivers Documentation/sound/alsa/soc/codec.txt | 45 +- Documentation/sound/alsa/soc/machine.txt | 38 +- Documentation/sound/alsa/soc/platform.txt | 12 +- include/sound/soc.h | 45 +- sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/blackfin/Kconfig | 11 +- sound/soc/blackfin/bf5xx-ac97.c | 4 +- sound/soc/blackfin/bf5xx-tdm.c | 10 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs4270.c | 4 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/sn95031.c | 495 ++++++++ sound/soc/codecs/sn95031.h | 99 ++ sound/soc/codecs/wm8523.c | 8 +- sound/soc/codecs/wm8741.c | 13 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 43 +- sound/soc/codecs/wm8955.c | 27 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 34 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8991.c | 1427 ++++++++++++++++++++++++ sound/soc/codecs/wm8991.h | 833 ++++++++++++++ sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 10 +- sound/soc/codecs/wm8995.c | 103 ++- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 45 +- sound/soc/mid-x86/Kconfig | 14 + sound/soc/mid-x86/Makefile | 5 + sound/soc/mid-x86/mfld_machine.c | 296 +++++ sound/soc/mid-x86/sst_platform.c | 459 ++++++++ sound/soc/mid-x86/sst_platform.h | 63 ++ sound/soc/pxa/z2.c | 3 +- sound/soc/samsung/ac97.c | 8 +- sound/soc/samsung/ac97.h | 21 - sound/soc/samsung/dma.c | 11 +- sound/soc/samsung/dma.h | 8 - sound/soc/samsung/goni_wm8994.c | 10 +- sound/soc/samsung/h1940_uda1380.c | 9 - sound/soc/samsung/i2s.c | 3 +- sound/soc/samsung/jive_wm8750.c | 11 - sound/soc/samsung/ln2440sbc_alc650.c | 7 - sound/soc/samsung/neo1973_gta02_wm8753.c | 14 +- sound/soc/samsung/pcm.c | 118 ++- sound/soc/samsung/pcm.h | 107 -- sound/soc/samsung/rx1950_uda1380.c | 11 - sound/soc/samsung/s3c-i2s-v2.c | 3 +- sound/soc/samsung/s3c2412-i2s.c | 12 +- sound/soc/samsung/s3c24xx-i2s.c | 14 +- sound/soc/samsung/s3c24xx_simtec.c | 7 - sound/soc/samsung/s3c24xx_simtec_hermes.c | 10 - sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 12 - sound/soc/samsung/s3c24xx_uda134x.c | 9 +- sound/soc/samsung/smartq_wm8987.c | 6 - sound/soc/samsung/smdk2443_wm9710.c | 7 - sound/soc/samsung/smdk_spdif.c | 5 - sound/soc/samsung/smdk_wm8580.c | 7 +- sound/soc/samsung/smdk_wm9713.c | 5 - sound/soc/samsung/spdif.c | 3 +- sound/soc/sh/fsi-ak4642.c | 11 +- sound/soc/soc-cache.c | 317 +++--- sound/soc/soc-core.c | 165 ++-- sound/soc/tegra/Kconfig | 26 + sound/soc/tegra/Makefile | 14 + sound/soc/tegra/harmony.c | 210 ++++ sound/soc/tegra/tegra_asoc_utils.c | 152 +++ sound/soc/tegra/tegra_asoc_utils.h | 31 + sound/soc/tegra/tegra_das.c | 264 +++++ sound/soc/tegra/tegra_das.h | 135 +++ sound/soc/tegra/tegra_i2s.c | 502 +++++++++ sound/soc/tegra/tegra_i2s.h | 165 +++ sound/soc/tegra/tegra_pcm.c | 401 +++++++ sound/soc/tegra/tegra_pcm.h | 55 + 78 files changed, 6347 insertions(+), 717 deletions(-) create mode 100644 sound/soc/codecs/sn95031.c create mode 100644 sound/soc/codecs/sn95031.h create mode 100644 sound/soc/codecs/wm8991.c create mode 100644 sound/soc/codecs/wm8991.h create mode 100644 sound/soc/mid-x86/Kconfig create mode 100644 sound/soc/mid-x86/Makefile create mode 100644 sound/soc/mid-x86/mfld_machine.c create mode 100644 sound/soc/mid-x86/sst_platform.c create mode 100644 sound/soc/mid-x86/sst_platform.h delete mode 100644 sound/soc/samsung/ac97.h create mode 100644 sound/soc/tegra/Kconfig create mode 100644 sound/soc/tegra/Makefile create mode 100644 sound/soc/tegra/harmony.c create mode 100644 sound/soc/tegra/tegra_asoc_utils.c create mode 100644 sound/soc/tegra/tegra_asoc_utils.h create mode 100644 sound/soc/tegra/tegra_das.c create mode 100644 sound/soc/tegra/tegra_das.h create mode 100644 sound/soc/tegra/tegra_i2s.c create mode 100644 sound/soc/tegra/tegra_i2s.h create mode 100644 sound/soc/tegra/tegra_pcm.c create mode 100644 sound/soc/tegra/tegra_pcm.h

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[alsa-devel] [PATCH] ALSA: HDA: Add SKU ignore for another Thinkpad Edge 14
by David Henningsson 20 Jan '11

20 Jan '11

BugLink: http://bugs.launchpad.net/bugs/705323 Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

Re: [alsa-devel] ALSA issue on DA850/OMAP-L138/AM18x
by Rajashekhara, Sudhakar 20 Jan '11

20 Jan '11

Hi Dan, Thanks for testing this patch. I'll add your Tested-by tag when I submit this patch. On Wed, Jan 19, 2011 at 00:32:21, Dan Sharon wrote: > These patches apply cleanly to 'v2.6.37' > 3c0eee3fe6a3a1c745379547c7e7c904aa64f6d5 > > of > git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git. > > The kernel was built with the 'da8xx_omapl_defconfig' from the omap-l1tree, > and 'make menuconfig' to enable ASoC (the defconfig already has CONFIG_REGULATOR=y, > CONFIG_REGULATOR_DUMMY=y, and CONFIG_REGULATOR_TPS6507X=y). > > Testing was done on a da850evm by tftp'ing the kernel into ram, and using the > SPI-flash-based rootfs from the PSP that shipped with the board. > > Using 'arecord -r 48000 -c 2 -f S32_BE -t raw -v -d 1 > /tmp/arecord.cap' produced > a file of 384000 bytes (48000 x 2 x 4 bytes/sample). > 'arecord -r 48000 -c 2 -f S32_BE -t raw -v > /dev/null' ran flawlessly for several > hours. > > I was not able to reproduce Sudhakar's problems with 'arecord -f dat | aplay -f dat'. > Did you try the above, immediately after the da850evm boots? I am not facing the issue if I run arecod and aplay together after executing arecord or aplay individually. Thanks, Sudhakar

1 0

[alsa-devel] Driver for TerraTec DMX 6Fire USB
by Torsten Schenk 19 Jan '11

19 Jan '11

Hello, i recently got a TerraTec DMX 6Fire USB. Since it's not supported by alsa, I decided to look into communications between the windows driver and the card. Here is the result. What should be revised: - I was unsure about naming conventions, so please check the name of card/midi/pcm/control/module. - Also the device needs firmware to be able to run. I did not included the firmware (from the windows driver) in the attachment since the 60kB mail size limit were exceeded last time I tried to post this driver. - Firmware: I read about that .ihex-parsing in drivers is not really approved, but I cannot convert the files easily into a binary format because the offsets are not contiguous. Of course there could be alternate ways to be thought about, which would imply that original files from TT are not supported since they are in ihex. - I didn't manage to create a configuration in /usr/share/alsa/cards. What is working: Everything except SPDIF - Hardware Master volume - PCM 44-192kHz@24 bits, 6 channels out, 4 channels in (analog) - MIDI in/out - firmware loading after cold start - phono/line switching If anything else comes to your attention, just let me know. Greetings, Torsten

1 0

[alsa-devel] [PATCH 4/4] ASoC: mid-x86 add jack detect support
by Harsha Priya 19 Jan '11

19 Jan '11

This patch adds the jack detection in machine driver and adds the jack detect logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/sn95031.c | 82 +++++++++++++- sound/soc/codecs/sn95031.h | 20 ++++ sound/soc/codecs/sn95031_adc_control.h | 194 ++++++++++++++++++++++++++++++++ sound/soc/mid-x86/mfld_machine.c | 159 +++++++++++++++++++++----- 4 files changed, 424 insertions(+), 31 deletions(-) create mode 100644 sound/soc/codecs/sn95031_adc_control.h diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 5e8d091..c436311 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -27,6 +27,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/input.h> #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -35,13 +36,15 @@ #include <sound/initval.h> #include <sound/tlv.h> #include "sn95031.h" +#include "sn95031_adc_control.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +struct snd_soc_codec *sn95031_codec; + /* * todo: - * jack detection * PM functions */ @@ -113,6 +116,15 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, return 0; } +/* enables mic bias voltage */ +void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, + BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, + BIT(2), BIT(2)); +} + static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -310,6 +322,7 @@ static const struct snd_kcontrol_new sn95031_snd_controls[] = { static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { /* all end points mic, hs etc */ + SND_SOC_DAPM_HP("HeadPhones", NULL), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("EPOUT"), @@ -428,6 +441,8 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { static const struct snd_soc_dapm_route sn95031_audio_map[] = { /* headset and earpiece map */ + { "HeadPhones", NULL, "HPOUTR" }, + { "HeadPhones", NULL, "HPOUTL" }, { "HPOUTL", NULL, "Headset Left Playback" }, { "HPOUTR", NULL, "Headset Right Playback" }, { "EPOUT", NULL, "Earpiece Playback" }, @@ -658,6 +673,66 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +int sn95031_get_headset_state(void) +{ + unsigned int mic_bias = sn95031_get_mic_bias(sn95031_codec); + + if (mic_bias >= SN_HP_START && mic_bias < SN_HP_END) { + pr_debug("sst: Detected Headphone!!!\n"); + snd_soc_update_bits(sn95031_codec, AUDIO_GPIO_CTRL, + BIT(1), BIT(0)); + } else if (mic_bias >= SN_AM_HS_START && mic_bias < SN_AM_HS_END) { + pr_debug("sst: Detected American headset\n"); + snd_soc_update_bits(sn95031_codec, AUDIO_GPIO_CTRL, + BIT(1), BIT(1)); + } else if (mic_bias >= SN_HS_START && mic_bias < SN_HS_END) { + pr_debug("sst: Detected Headset!!!\n"); + snd_soc_update_bits(sn95031_codec, AUDIO_GPIO_CTRL, + BIT(1), BIT(0)); + enable_jack_btn(sn95031_codec); + return SND_JACK_HEADSET; + } else + pr_debug("sst: Detected Open Cable!!!\n"); + + return SND_JACK_HEADPHONE; +} + +void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg) +{ + unsigned int status, mask; + static int jack_state = SND_JACK_HEADSET; + + if (!sn95031_codec) { + pr_err("codec is null\n"); + return; + } + pr_debug("interrupt id read in sram = 0x%x\n", jack_msg->intr_id); + if (jack_msg->intr_id & 0x1) { + pr_debug("short_push detected\n"); + mask = status = SND_JACK_BTN_0; + } else if (jack_msg->intr_id & 0x2) { + pr_debug("long_push detected\n"); + mask = status = SND_JACK_BTN_1; + } else if (jack_msg->intr_id & 0x4) { + mask = status = sn95031_get_headset_state(); + jack_state = mask; + } else if (jack_msg->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + mask = jack_state; + disable_jack_btn(sn95031_codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(mfld_jack, status, mask); + /*button pressed and released */ + if (status == SND_JACK_BTN_0 || status == SND_JACK_BTN_1) + snd_soc_jack_report(mfld_jack, 0, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { @@ -667,6 +742,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.idle_bias_off = 1; + sn95031_codec = codec; /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots @@ -733,7 +809,7 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sn95031_codec = { +struct snd_soc_codec_driver sn95031_soc_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, .read = sn95031_read, @@ -744,7 +820,7 @@ struct snd_soc_codec_driver sn95031_codec = { static int __devinit sn95031_device_probe(struct platform_device *pdev) { pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); - return snd_soc_register_codec(&pdev->dev, &sn95031_codec, + return snd_soc_register_codec(&pdev->dev, &sn95031_soc_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d9..e79176d 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -25,6 +25,7 @@ */ #ifndef _SN95031_H #define _SN95031_H +#include <sound/jack.h> /*register map*/ #define SN95031_VAUD 0xDB @@ -96,4 +97,23 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +struct mfld_jack_msg_wq { + u8 intr_id; + struct work_struct wq; +}; + +static inline void disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +extern void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg); + #endif diff --git a/sound/soc/codecs/sn95031_adc_control.h b/sound/soc/codecs/sn95031_adc_control.h new file mode 100644 index 0000000..0971321 --- /dev/null +++ b/sound/soc/codecs/sn95031_adc_control.h @@ -0,0 +1,194 @@ +#ifndef __MFLD_MACHINE_ADC_CONTROL_H__ +#define __MFLD_MACHINE_ADC_CONTROL_H_ +/* + * sn95031_adc_control.h - Module for adc programming + * + * Copyright (C) 2008-10 Intel Corporation + * Authors: R Durgadadoss <r.durgadoss(a)intel.com> + * Dharageswari R <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 + +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 + +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 + +#define ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define ADC_LOOP_MAX (ADC_CHANLS_MAX - 1) + +/* ADC channel code values */ +#define AUDIO_DETECT_CODE 0x06 + +#define AUDIO_GPIO_CTRL 0x070 + +/* ADC base addresses */ +#define ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ + +/* multipier to convert to mV */ +#define ADC_ONE_LSB_MULTIPLIER 2346 + +enum MIC_BIAS_ZONES { + SN_HP_START = 0, + SN_HP_END = 400, + SN_AM_HS_START = 400, + SN_AM_HS_END = 650, + SN_HS_START = 650, + SN_HS_END = 2000, + SN_UNDEFINED, +}; + +void sn95031_enable_mic_bias(struct snd_soc_codec *codec); + +/** + * configure_adc - enables/disables the ADC for conversion + * @val: zero: disables the ADC non-zero:enables the ADC + * + * Enable/Disable the ADC depending on the argument + * + * Can sleep + */ +static inline void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + else + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + +/** + * find_free_channel - finds an empty channel for conversion + * + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * + */ +static inline int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/** + * sn95031_initialize_adc - initializing the ADC + * @dev: our device structure + * + * Initialize the ADC for reading thermistor values. Can sleep. + */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, AUDIO_DETECT_CODE | 0x10); + + chnl_addr = ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/** + * sn95031_get_mic_bias - get the mic bias in mV + * + * reads the ADC registers and gets the mic bias value in mV. + */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + /* convesion into mV */ + mic_bias = (ADC_ONE_LSB_MULTIPLIER * mic_bias) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +#endif + diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1cd49a2..7926151 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -27,6 +27,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -35,10 +36,32 @@ #define MID_MONO 1 #define MID_STEREO 2 #define MID_MAX_CAP 5 +#define IRQ_BASE 0xFFFF7FCD +#define IRQ_SIZE 1 static unsigned int hs_switch; static unsigned int lo_dac; +struct mfld_mc_private { + struct platform_device *socdev; + struct workqueue_struct *mfld_jack_wq; + void __iomem *int_base; +}; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "HeadPhones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -67,13 +90,11 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) { pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } snd_soc_dapm_sync(&codec->dapm); @@ -91,12 +112,10 @@ static void lo_enable_out_pins(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } } @@ -130,8 +149,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, case 1: pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); break; @@ -175,8 +193,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(dapm, "HeadPhones"); /* default is lineout NC, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); @@ -185,11 +202,31 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - /* we are using DMICs here so configure how the clock and out pin is set */ + /* we are using DMICs here so configure how the + * clock and out pin is set */ snd_soc_write(codec, SN95031_DMICBUF0123, 0x85); snd_soc_write(codec, SN95031_DMICBUF45, 0x02); snd_soc_update_bits(codec, SN95031_DMICMUX, 0x7, 0x07); - return snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + /* Headset jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + disable_jack_btn(codec); + return ret_val; + } struct snd_soc_dai_link mfld_msic_dailink[] = { @@ -238,37 +275,103 @@ static struct snd_soc_card snd_soc_card_mfld = { .num_links = ARRAY_SIZE(mfld_msic_dailink), }; +void snd_mfld_jack_detection(struct work_struct *work) +{ + struct mfld_jack_msg_wq *jack_msg = + container_of(work, struct mfld_jack_msg_wq, wq); + + sn95031_jack_detection(&mfld_jack, jack_msg); + kfree(jack_msg); +} + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_jack_msg_wq *jack_msg; + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + jack_msg = kzalloc(sizeof(*jack_msg), GFP_ATOMIC); + memcpy_fromio(&jack_msg->intr_id, + ((void *)(mc_private->int_base)), + sizeof(u8)); + pr_debug("intr_handler called value %x\n", jack_msg->intr_id); + INIT_WORK(&jack_msg->wq, snd_mfld_jack_detection); + queue_work(mc_private->mfld_jack_wq, &jack_msg->wq); + + return IRQ_HANDLED; +} + static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) { - struct platform_device *socdev; - int ret_val = 0; + int ret_val = 0, irq; + struct mfld_mc_private *mc_private_data; pr_debug("snd_mfld_mc_probe called\n"); - socdev = platform_device_alloc("soc-audio", -1); - if (!socdev) { - pr_err("soc-audio device allocation failed\n"); + mc_private_data = kzalloc(sizeof(*mc_private_data), GFP_ATOMIC); + if (!mc_private_data) { + pr_err("mem alloation of mc_private failed\n"); return -ENOMEM; } - platform_set_drvdata(socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(socdev); + + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_private_data->int_base = ioremap_nocache(IRQ_BASE, IRQ_SIZE); + if (!mc_private_data->int_base) { + pr_err("Mapping of cache failed\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* create botoom half handler for interrupt */ + mc_private_data->mfld_jack_wq = create_workqueue("mfld_jack_wq"); + if (!mc_private_data->mfld_jack_wq) { + pr_err("cannot create jack workqueue\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* register for interrupt */ + ret_val = request_irq(irq, snd_mfld_jack_intr_handler, + IRQF_SHARED, pdev->dev.driver->name, + mc_private_data); + if (ret_val) { + pr_err("cannot register IRQ\n"); + goto destroy_wq; + } + /* create soc device */ + mc_private_data->socdev = platform_device_alloc("soc-audio", -1); + if (!mc_private_data->socdev) { + pr_err("soc-audio device allocation failed\n"); + ret_val = -ENOMEM; + goto destroy_wq; + } + platform_set_drvdata(mc_private_data->socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(mc_private_data->socdev); if (ret_val) { pr_err("Unable to add soc-audio device, err %d\n", ret_val); - platform_device_put(socdev); + goto unregister; } - - platform_set_drvdata(pdev, socdev); - + platform_set_drvdata(pdev, mc_private_data); pr_debug("successfully exited probe\n"); return ret_val; + +unregister: + platform_device_put(mc_private_data->socdev); +destroy_wq: + destroy_workqueue(mc_private_data->mfld_jack_wq); +unalloc: + kfree(mc_private_data); + return ret_val; } static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) { - struct platform_device *socdev = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); + struct mfld_mc_private *mc_private_data = platform_get_drvdata(pdev); - platform_device_unregister(socdev); + pr_debug("snd_mfld_mc_remove called\n"); + free_irq(platform_get_irq(pdev, 0), mc_private_data); + destroy_workqueue(mc_private_data->mfld_jack_wq); + platform_device_unregister(mc_private_data->socdev); platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.2.3

3 3

[alsa-devel] [PATCH] ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()
by Dimitris Papastamos 19 Jan '11

19 Jan '11

In the wm8995_set_tristate() function when updating the register bits use the value and not the register index as the value argument. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8995.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 7d56341..67eaaec 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1265,7 +1265,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } /* The size in bits of the FLL divide multiplied by 10 -- 1.7.3.5

3 2

Re: [alsa-devel] [PATCH] Asoc: WM8994: fix wrong value in tristate function
by Mark Brown 19 Jan '11

19 Jan '11

On Wed, Jan 19, 2011 at 07:10:47PM +0800, Qiao Zhou wrote: > fix wrong value in wm8994_set_tristate func. when updating reg bits, > it should use "value", not "reg". > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> Applied, thanks.

1 0

[alsa-devel] [PATCH 1/4] ASoC: sn95031: add capture support
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch adds the support for capture path in sn95031 codec. This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs are connected through a MUX to ADC. The TX paths can be assigned to any of the ADCs or DMICs. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/codecs/sn95031.c | 292 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 291 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 593632c..5e8d091 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -33,6 +33,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -40,7 +41,6 @@ /* * todo: - * capture paths * jack detection * PM functions */ @@ -145,6 +145,167 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, return 0; } +static int amic1_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int value = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("AMIC1 SND_SOC_DAPM_EVENT_ON doing\n"); + value = BIT(2); + } + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(2), value); + return 0; +} + +static int amic2_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int value = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("AMIC2 SND_SOC_DAPM_EVENT_ON doing\n"); + value = BIT(3); + } + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(3), value); + return 0; +} + +static int dmic12_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk = 0, out = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("DMIC12 SND_SOC_DAPM_EVENT_ON doing\n"); + ldo = BIT(5)|BIT(4); + clk = BIT(0); + out = BIT(3); + } + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + msleep(1); + /* enable/disable DMIC clock and o/p */ + snd_soc_update_bits(w->codec, SN95031_DMICLK, BIT(0), clk); + snd_soc_update_bits(w->codec, SN95031_DMICMUX, BIT(3), out); + return 0; +} + +static int dmic34_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk = 0, out = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("DMIC34 SND_SOC_DAPM_EVENT_ON doing\n"); + ldo = BIT(5)|BIT(4); + clk = BIT(1); + out = BIT(4); + } + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + msleep(1); + /* enable/disable DMIC clock */ + snd_soc_update_bits(w->codec, SN95031_DMICLK, BIT(1), clk); + snd_soc_update_bits(w->codec, SN95031_DMICMUX, BIT(4), out); + return 0; +} + +static int dmic56_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk = 0, out = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("DMIC56 SND_SOC_DAPM_EVENT_ON doing\n"); + ldo = BIT(7)|BIT(6); + clk = BIT(2); + out = BIT(5); + } + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); + msleep(1); + /* enable/disable DMIC clock */ + snd_soc_update_bits(w->codec, SN95031_DMICLK, BIT(2), clk); + snd_soc_update_bits(w->codec, SN95031_DMICMUX, BIT(5)|BIT(2), out); + return 0; +} +/* mux controls */ +static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; + +static const struct soc_enum sn95031_micl_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micl_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micl_enum); + +static const struct soc_enum sn95031_micr_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micr_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micr_enum); + +static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", + "DMIC4", "DMIC5", "DMIC6", + "ADC Left", "ADC Right" }; + +static const struct soc_enum sn95031_input1_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input1_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input1_enum); + +static const struct soc_enum sn95031_input2_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input2_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input2_enum); + +static const struct soc_enum sn95031_input3_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input3_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input3_enum); + +static const struct soc_enum sn95031_input4_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input4_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input4_enum); + +/* capture path controls */ + +/* 0dB to 30dB in 10dB steps */ +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); + +static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; + +static const struct soc_enum sn95031_micmode1_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); +static const struct soc_enum sn95031_micmode2_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); + +static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; + +static const struct soc_enum sn95031_dmic12_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic34_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic56_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); + +static const struct snd_kcontrol_new sn95031_snd_controls[] = { + SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), + SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum), + SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum), + SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum), + SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum), + SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1, + 2, 4, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2, + 2, 4, 0, mic_tlv), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { @@ -159,6 +320,20 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIB1OUT"), SND_SOC_DAPM_OUTPUT("VIB2OUT"), + SND_SOC_DAPM_MIC("AMIC1", amic1_mic_bias), /* headset mic */ + SND_SOC_DAPM_MIC("AMIC2", amic2_mic_bias), + SND_SOC_DAPM_MIC("DMIC1", dmic12_mic_bias), + SND_SOC_DAPM_MIC("DMIC2", dmic12_mic_bias), + SND_SOC_DAPM_MIC("DMIC3", dmic34_mic_bias), + SND_SOC_DAPM_MIC("DMIC4", dmic34_mic_bias), + SND_SOC_DAPM_MIC("DMIC5", dmic56_mic_bias), + SND_SOC_DAPM_MIC("DMIC6", dmic56_mic_bias), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + + SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, sn95031_vhs_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -209,6 +384,46 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SN95031_VIB1C5, 1, 0), SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", SN95031_VIB2C5, 1, 0), + + /* capture widgets */ + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "LineIn Enable Left", + SN95031_MICAMP1, 7, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "LineIn Enable Right", + SN95031_MICAMP2, 7, 1, 1, 0), + + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "MIC1 Enable", SN95031_MICAMP1, + 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "MIC2 Enable", SN95031_MICAMP2, + 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX1 Enable", SN95031_AUDIOTXEN, + 2, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX2 Enable", SN95031_AUDIOTXEN, + 3, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX3 Enable", SN95031_AUDIOTXEN, + 4, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX4 Enable", SN95031_AUDIOTXEN, + 5, 1, 1, 0), + + /* ADC have null stream as they will be turned ON by TX path */ + SND_SOC_DAPM_ADC("ADC Left", NULL, + SN95031_ADCCONFIG, 0, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, + SN95031_ADCCONFIG, 2, 0), + + SND_SOC_DAPM_MUX("Mic_InputL Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control), + SND_SOC_DAPM_MUX("Mic_InputR Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control), + + SND_SOC_DAPM_MUX("Txpath1 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control), + SND_SOC_DAPM_MUX("Txpath2 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control), + SND_SOC_DAPM_MUX("Txpath3 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control), + SND_SOC_DAPM_MUX("Txpath4 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control), + }; static const struct snd_soc_dapm_route sn95031_audio_map[] = { @@ -250,6 +465,69 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Lineout Right Playback", NULL, "Headset Right Filter"}, { "Lineout Right Playback", NULL, "Speaker Right Filter"}, { "Lineout Right Playback", NULL, "Vibra2 DAC"}, + + /* Headset (AMIC1) mic */ + { "MIC1 Enable", NULL, "AMIC1"}, + { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"}, + + /* AMIC2 */ + { "MIC2 Enable", NULL, "AMIC2"}, + { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"}, + + + /* Linein */ + { "LineIn Enable Left", NULL, "LINEINL"}, + { "LineIn Enable Right", NULL, "LINEINR"}, + { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"}, + { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"}, + + /* ADC connection */ + { "ADC Left", NULL, "Mic_InputL Capture Route"}, + { "ADC Right", NULL, "Mic_InputR Capture Route"}, + + /*TX path inputs*/ + { "Txpath1 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath2 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath3 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath4 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath1 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath1 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath2 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath3 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath4 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath1 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath2 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath3 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath4 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath1 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath2 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath3 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath4 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath1 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath2 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath3 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath4 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath1 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath2 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath3 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath4 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath1 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath2 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath3 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath4 Capture Route", "DMIC6", "DMIC6"}, + + /* tx path */ + { "TX1 Enable", NULL, "Txpath1 Capture Route"}, + { "TX2 Enable", NULL, "Txpath2 Capture Route"}, + { "TX3 Enable", NULL, "Txpath3 Capture Route"}, + { "TX4 Enable", NULL, "Txpath4 Capture Route"}, + { "PCM_Out", NULL, "TX1 Enable"}, + { "PCM_Out", NULL, "TX2 Enable"}, + { "PCM_Out", NULL, "TX3 Enable"}, + { "PCM_Out", NULL, "TX4 Enable"}, }; /* speaker and headset mutes, for audio pops and clicks */ @@ -339,6 +617,13 @@ struct snd_soc_dai_driver sn95031_dais[] = { .rates = SN95031_RATES, .formats = SN95031_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, .ops = &sn95031_headset_dai_ops, }, { .name = "SN95031 Speaker", @@ -390,6 +675,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32); /* pcm port setting * This sets the pcm port to slave and clock at 19.2Mhz which * can support 6slots, sampling rate set per stream in hw-params @@ -423,6 +710,9 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); + snd_soc_add_controls(codec, sn95031_snd_controls, + ARRAY_SIZE(sn95031_snd_controls)); + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, ARRAY_SIZE(sn95031_dapm_widgets)); if (ret) -- 1.7.2.3

3 3

[alsa-devel] [PATCH 3/4] ASoC: mid-x86: Add support for capture in machine driver
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> This configures the capture unused pins and sets the DMIC cfg by default (also done by user) Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be..1cd49a2 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -182,6 +182,13 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + /* we are using DMICs here so configure how the clock and out pin is set */ + snd_soc_write(codec, SN95031_DMICBUF0123, 0x85); + snd_soc_write(codec, SN95031_DMICBUF45, 0x02); + snd_soc_update_bits(codec, SN95031_DMICMUX, 0x7, 0x07); return snd_soc_dapm_sync(dapm); } -- 1.7.2.3

3 3

[alsa-devel] [PATCH 0/4] ASoC: patch series to add support for capture and jack sense in mid-x86 drivers
by Harsha, Priya 19 Jan '11

19 Jan '11

Hi Mark, This patch set adds support for capture and jack sense in mid-x6 platform, machine and TI sn95031 codec drivers. [PATCH 1/4] ASoC: sn95031: add capture support This patch adds support for capture codec dais and widgets [PATCH 2/4] ASoC: sst_platform: add support for capture stream on headset dai This patch adds mid-x86 sst platform driver support for capture on headset dai [PATCH 3/4] ASoC: mid-x86: Add support for capture in machine driver This patch adds support for enabling capture in mfld machine driver [PATCH 4/4] ASoC: mid-x86 add jack detect support This patch adds support for jack detection and reporting support in sn95031 codec and mfld_machine driver -Harsha

1 0

[alsa-devel] [PATCH 3/4] ASoC: mid-x86: Add support for capture in machine driver
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> This configures the capture unused pins and sets the DMIC cfg by default (also done by user) Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be..1cd49a2 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -182,6 +182,13 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + /* we are using DMICs here so configure how the clock and out pin is set */ + snd_soc_write(codec, SN95031_DMICBUF0123, 0x85); + snd_soc_write(codec, SN95031_DMICBUF45, 0x02); + snd_soc_update_bits(codec, SN95031_DMICMUX, 0x7, 0x07); return snd_soc_dapm_sync(dapm); } -- 1.7.2.3

1 0

Re: [alsa-devel] [PATCH 0/3] Tegra: Add internal speaker support
by Liam Girdwood 19 Jan '11

19 Jan '11

On Tue, 2011-01-18 at 16:37 -0700, Stephen Warren wrote: > From: Stephen Warren <swarren(a)nvidia.com> > > Harmony has headers on the board for an internal speaker and internal mic. > I've started looking at enabling these, along with full jack detection etc. > This first set of patches is an RFC (Request For Comments) on whether it's > the correct direction, particularly with respect to: > > a) Is this the right way to expose GPIO APIs from a codec? The tlv320aic3 > seems to do it this way, but wm8962.c uses the generic GPIO API. I shied > away from the latter, since I wasn't sure how to name the WM8903's GPIOs > in the gpio_* calls. I suppose it'd need a bunch of GPIO driver platform > data to hook it all together and name GPIOs from a mach-tegra/include > header file, but if this is the way to go, I need to read up on that more. > > b) Is the WM8903 platform_data handling full fleshed out; simply by having > platform_data, WM8903_WSEQ_ENA is set, whereas without any pdata, it > isn't. Should this be conditional, or should I always have specified some > platform data for Tegra, and I'm just getting lucky that it works without > any? > > Thanks for any coments. > 2 & 3 look fine and Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

1 0

[alsa-devel] [PATCH 2/4] ASoC: sst_platform: add support for capture stream on headset dai
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/sst_platform.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index a4e3fa3..0d8c430 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -73,6 +73,12 @@ struct snd_soc_dai_driver sst_platform_dai[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, }, { .name = "Speaker-cpu-dai", -- 1.7.2.3

1 0

Re: [alsa-devel] [PATCH] Asoc: WM8994: fix wrong value in tristate function
by Liam Girdwood 19 Jan '11

19 Jan '11

On Wed, 2011-01-19 at 19:10 +0800, Qiao Zhou wrote: > fix wrong value in wm8994_set_tristate func. when updating reg bits, > it should use "value", not "reg". > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

1 0

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 19 Jan '11

19 Jan '11

The following changes since commit 18b022eb117e7f70c191267551ff865f278a9258: Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-01-12 12:26:24 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Alexander Sverdlin (1): ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code Barry Song (2): ASoC: Blackfin TDM: use external frame syncs ASoC: Blackfin: fix DAI/SPORT config dependency issues Dimitris Papastamos (1): ASoC: WM8990: msleep() takes milliseconds not jiffies Matti J. Aaltonen (1): ASoC: WL1273 FM radio: Fix breakage with MFD API changes Mike Frysinger (2): ASoC: Blackfin TDM: fix missed snd_soc_dai_get_drvdata update ASoC: Blackfin AC97: fix build error after multi-component update Seungwhan Youn (1): ASoC: documentation updates Vasily Khoruzhick (2): ASoC: PXA: Fix jack detection on Zipit Z2 ASoC: PXA: Fix codec address on Zipit Z2 Documentation/sound/alsa/soc/codec.txt | 45 +++++++++---------- Documentation/sound/alsa/soc/machine.txt | 38 ++++------------ Documentation/sound/alsa/soc/platform.txt | 12 ++++- sound/soc/blackfin/Kconfig | 11 +++-- sound/soc/blackfin/bf5xx-ac97.c | 4 +- sound/soc/blackfin/bf5xx-tdm.c | 10 ++-- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 29 +++++------- sound/soc/codecs/wl1273.h | 71 ----------------------------- sound/soc/codecs/wm8990.c | 10 ++-- sound/soc/ep93xx/ep93xx-i2s.c | 18 ++++---- sound/soc/pxa/z2.c | 3 +- 12 files changed, 82 insertions(+), 171 deletions(-)

2 2

Re: [alsa-devel] [PATCH 0/3] Tegra: Add internal speaker support
by Mark Brown 19 Jan '11

19 Jan '11

On Tue, Jan 18, 2011 at 04:43:33PM -0800, Colin Cross wrote: > On Tue, Jan 18, 2011 at 4:29 PM, Stephen Warren <swarren(a)nvidia.com> wrote: > > I'll go read up on gpiolib some more to see how those API-level numbers get > > mapped to a specific controller and GPIO number. > They usually come in through platform data from the board file. See > Olof's recent Tegra sdhci patches for an example. Tegra GPIO names > are defined in arch/arm/mach-tegra/gpio-names.h. Yup, and gpiolib also supports automatic assignment of numbers if you specify -1 as the base number though that's a lot less useful within the kernel.

1 0

[PATCH] ALSA: HDA: Add SKU ignore for another Thinkpad Edge 14
by David Henningsson 19 Jan '11

19 Jan '11

BugLink: http://bugs.launchpad.net/bugs/705323 Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f006ee..b974164 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14951,6 +14951,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.7.1 --------------070500050103000507060302--

1 0

[alsa-devel] [PATCH 1/3] ASoC: Explicitly say if we're powering up or down
by Mark Brown 19 Jan '11

19 Jan '11

Rather than passing the sequence to use for DAPM widgets around by reference explicitly say if we're powering up or down until the point where we need the sequence itself. This should make no practical difference in itself but supports future refactoring. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 37 +++++++++++++++++++++++++------------ 1 files changed, 25 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 499730a..57e1c9f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -726,8 +726,15 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, - int sort[]) + bool power_up) { + int *sort; + + if (power_up) + sort = dapm_up_seq; + else + sort = dapm_down_seq; + if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; if (a->reg != b->reg) @@ -741,12 +748,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, /* Insert a widget in order into a DAPM power sequence. */ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, struct list_head *list, - int sort[]) + bool power_up) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, list, power_list) - if (dapm_seq_compare(new_widget, w, sort) < 0) { + if (dapm_seq_compare(new_widget, w, power_up) < 0) { list_add_tail(&new_widget->power_list, &w->power_list); return; } @@ -857,7 +864,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, * handled. */ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, - struct list_head *list, int event, int sort[]) + struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); @@ -865,6 +872,12 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, int cur_reg = SND_SOC_NOPM; struct snd_soc_dapm_context *cur_dapm = NULL; int ret; + int *sort; + + if (power_up) + sort = dapm_up_seq; + else + sort = dapm_down_seq; list_for_each_entry_safe(w, n, list, power_list) { ret = 0; @@ -1002,10 +1015,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(w, &card->widgets, list) { switch (w->id) { case snd_soc_dapm_pre: - dapm_seq_insert(w, &down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, false); break; case snd_soc_dapm_post: - dapm_seq_insert(w, &up_list, dapm_up_seq); + dapm_seq_insert(w, &up_list, true); break; default: @@ -1025,9 +1038,9 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_widget_power(w, power); if (power) - dapm_seq_insert(w, &up_list, dapm_up_seq); + dapm_seq_insert(w, &up_list, true); else - dapm_seq_insert(w, &down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, false); w->power = power; break; @@ -1086,12 +1099,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(dapm, &down_list, event, dapm_down_seq); + dapm_seq_run(dapm, &down_list, event, false); dapm_widget_update(dapm); /* Now power up. */ - dapm_seq_run(dapm, &up_list, event, dapm_up_seq); + dapm_seq_run(dapm, &up_list, event, true); list_for_each_entry(d, &dapm->card->dapm_list, list) { /* If we just powered the last thing off drop to standby bias */ @@ -2372,7 +2385,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) if (w->dapm != dapm) continue; if (w->power) { - dapm_seq_insert(w, &down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, false); w->power = 0; powerdown = 1; } @@ -2383,7 +2396,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) */ if (powerdown) { snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE); - dapm_seq_run(dapm, &down_list, 0, dapm_down_seq); + dapm_seq_run(dapm, &down_list, 0, false); snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY); } } -- 1.7.2.3

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[alsa-devel] [PATCH - [hdspm] 0/3] Support for RME RayDAT and AIO
by Adrian Knoth 19 Jan '11

19 Jan '11

Hi! As announced a few days ago, I've integrated the offspring hdspm driver by Florian Faber. The new code now also supports the 36in/36out RME RayDAT cards and RME AIOs. I tried to follow the kernel's coding style as much as possible, however, there are still a few overlong lines left. For example, I didn't want to break "if read_reg(...) & 0x3f" at "& 0x3f", as it worsens readability. If it's required to comply, feel free to add the missing line breaks. Right now, RayDATs and AIOs can only operate on a single period size of 1024 samples, because these cards always use a fixed 16k buffer per channel and only vary interrupt frequency. I'm going to try to fix this after I've learned more about the PCM interface. ;) Any suggestions welcome. Cheers Adrian Knoth (3): Add headers for RayDAT and AIO support. Enable RME RayDAT and AIO Add RayDAT and AIO strings to Kconfig include/hdspm.h | 442 +++++- pci/Kconfig | 6 +- pci/rme9652/hdspm.c | 4135 +++++++++++++++++++++++++++++++++++++-------------- 3 files changed, 3442 insertions(+), 1141 deletions(-) -- 1.7.2.3

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Re: [alsa-devel] [PATCH 0/3] Tegra: Add internal speaker support
by Mark Brown 19 Jan '11

19 Jan '11

On Tue, Jan 18, 2011 at 04:37:36PM -0700, Stephen Warren wrote: > a) Is this the right way to expose GPIO APIs from a codec? The tlv320aic3 > seems to do it this way, but wm8962.c uses the generic GPIO API. I shied > away from the latter, since I wasn't sure how to name the WM8903's GPIOs > in the gpio_* calls. I suppose it'd need a bunch of GPIO driver platform > data to hook it all together and name GPIOs from a mach-tegra/include > header file, but if this is the way to go, I need to read up on that more. What naming is required for the GPIOs? GPIOs are referred to by number in Linux. > b) Is the WM8903 platform_data handling full fleshed out; simply by having > platform_data, WM8903_WSEQ_ENA is set, whereas without any pdata, it > isn't. Should this be conditional, or should I always have specified some > platform data for Tegra, and I'm just getting lucky that it works without > any? Ah, your comment about the sequencer is clearer now. This is not enabling the write sequencer, that'll be done transiently when required if it's not set here. It's enabling a clock which is used by several parts of the chip including the write sequencer all the time so that the mic detection can function in the absence of any other software managing it. This is why it's conditional on microphone detection being configured, other functions can have this clock disabled most of the time.

2 1

Re: [alsa-devel] [PATCH 1/3] ASoC: WM8903: Add wm8903_set_gpio
by Mark Brown 19 Jan '11

19 Jan '11

On Tue, Jan 18, 2011 at 04:37:37PM -0700, Stephen Warren wrote: > From: Stephen Warren <swarren(a)nvidia.com> This should be implemented using gpiolib - see WM8962 for an example of this. That way the control can be plugged into more other subsystems in the kernel (probably including whenever I or someone else gets around to it a widget like the one you open code in a later patch). I know some of the drivers do export the GPIOs using custom APIs but that's something that should be fixed. > Also, fix platform_data handling to have an explicit "don't touch this pin" > option. This should've been a separate patch since it's OK to go in as-is. > Temporarily hack out the mic detection platform_data handling for simplicity There's no patch restoring the functionality... > tesing the GPIO changes. Should the write-sequencer enabling be conditional? Yes. It's not actually enabling the write sequencer itself, it's enabling a low rate clock that's shared between a number of functions in the device including jack detection. Without the clock jack detection will not function.

1 0

[alsa-devel] [PATCH] ASoC: Add dependency on INTEL_SCU_IPC for Intel MID drivers
by Koul, Vinod 18 Jan '11

18 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index e0123a4..1ad7538 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,5 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" + depends on INTEL_SCU_IPC select SND_SOC_SN95031 select SND_SST_PLATFORM help -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: core: Remove dapm_sync call from soc_post_component_init
by Peter Ujfalusi 18 Jan '11

18 Jan '11

snd_soc_dapm_new_widgets will call dapm_power_widgets at the end, so there is no need to call snd_soc_dapm_sync after snd_soc_dapm_new_widgets. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/soc-core.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291..952f104 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1467,7 +1467,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* Make sure all DAPM widgets are instantiated */ snd_soc_dapm_new_widgets(&codec->dapm); - snd_soc_dapm_sync(&codec->dapm); /* register the rtd device */ rtd->codec = codec; -- 1.7.4.rc2

4 3

[alsa-devel] [PATCH 1/2] ASoC: PXA: Fix jack detection on Zipit Z2
by Vasily Khoruzhick 18 Jan '11

18 Jan '11

Fix jack detection on Zipit Z2, otherwise it disables headphones output when jack is connected Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/pxa/z2.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 4cc841b..0da889d 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -105,6 +105,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { .name = "hsdet-gpio", .report = SND_JACK_HEADSET, .debounce_time = 200, + .invert = 1, }, }; -- 1.7.4.rc1

4 5

[alsa-devel] [RFC PATCH] theory about CS8247 failed reads
by Brian Bloniarz 18 Jan '11

18 Jan '11

On my M-Audio Delta 66 (rev. E), I see this line: [ 23.248721] invalid CS8427 signature 0x0: let me try again... This is caused by the first request to the CS8427 failing, and this workaround: > commit 4eb4550ab37d351ab0973ccec921a5a2d8560ec7, > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Sep 20 17:34:57 2007 +0200 > > [ALSA] Workaround for invalid signature read of CS8427 > > Reading the signature of CS8427 over SPI/I2C fails on some devices > by mysterious reason. In most cases, however, it succeeds at the > sequential read. So, let's give a second chance to check the signature > again. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > Signed-off-by: Jaroslav Kysela <perex(a)suse.cz> I have a theory, please read the patch and patch description below. With the patch applied, the printk goes away. Does this sound like a reasonable theory? Does anyone recall, did this failed read occur on any cards besides M-Audio Delta? I'm wondering if this is the whole explanation, in which case the previous workaround could be removed. ------------------------ [PATCH] [ALSA] ice1712 delta - initialize SPI clock. The driver was using an initial value for the clock on the SPI bus which was read from ICE1712 EEPROM, ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02) It appears some cards have it default high, some cards have it default low. On my Delta 66 rev. E: $ cat /proc/asound/M66/ice1712 | grep 'GPIO state' GPIO state : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */ On my Audiophile 2496: $ cat /proc/asound/M2496/ice1712 | grep 'GPIO state' GPIO state : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */ It must be raised before the first SPI write happens, or the write will fail, leading to: [ 23.248721] invalid CS8427 signature 0x0: let me try again... I theorize that 4eb4550ab37d351ab0973ccec921a5a2d8560ec7 is no longer needed, it was a different way to workaround the problem. Signed-off-by: Brian Bloniarz <brian.bloniarz(a)gmail.com> --- sound/pci/ice1712/delta.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 7b62de0..84a6031 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -622,6 +622,13 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) break; } + /* initialize the SPI clock to high */ + unsigned char tmp; + tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); + tmp |= ICE1712_DELTA_AP_CCLK; + snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp); + udelay(5); + /* initialize spdif */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_AUDIOPHILE: -- 1.7.1

2 2

[alsa-devel] [PATCH 4/4] ASoC: soc core allow machine driver to register the card
by Koul, Vinod 18 Jan '11

18 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> The machine driver can't register the card directly and need to do this thru soc-audio device creation This patch allows the register and unregister card to be called by machine drivers Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <harsha.priya(a)intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 541ddfa..9952254 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -258,6 +258,8 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_register_card(struct snd_soc_card *card); +int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8305712..87348ab 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -58,8 +58,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int snd_soc_register_card(struct snd_soc_card *card); -static int snd_soc_unregister_card(struct snd_soc_card *card); static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* @@ -1870,6 +1868,10 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; + /* no card, so machine driver is registering card */ + if (!card) + return 0; + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; snd_soc_initialize_card_lists(card); @@ -1919,6 +1921,9 @@ static int soc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + if (!card) + return 0; + snd_soc_unregister_card(card); return 0; } @@ -3105,11 +3110,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); * * @card: Card to register * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_register_card(struct snd_soc_card *card) +int snd_soc_register_card(struct snd_soc_card *card) { int i; @@ -3141,17 +3143,15 @@ static int snd_soc_register_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core * * @card: Card to unregister * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_unregister_card(struct snd_soc_card *card) +int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); @@ -3162,6 +3162,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_unregister_card); /* * Simplify DAI link configuration by removing ".-1" from device names -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Add dependency on X86 for Intel MID drivers
by Mark Brown 18 Jan '11

18 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/mid-x86/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index e0123a4..8adbc1e 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,5 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" + depends on X86 select SND_SOC_SN95031 select SND_SST_PLATFORM help -- 1.7.2.3

3 6

Re: [alsa-devel] [PATCH] ASoC: dapm: Fix race condition in widgets power list creation
by Mark Brown 18 Jan '11

18 Jan '11

On Tue, Jan 18, 2011 at 01:35:37PM +0200, Peter Ujfalusi wrote: > Hmm, the patch looks quite similar to the patch I've sent back in November: > http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033362.h… My thought exactly. > On 01/18/11 12:39, ext Mark Brown wrote: > > This needs a bit more analysis - why are we getting multiple > > simultaneous calls to dapm_power_widgets() and is that itself safe? The > > ASoC locking model has always been to have a big lock around the entire > > card rather than to lock subcomponents, and for example this isn't going > > to make sure we're consistent with register map accesses from other > > parts of the code. > As far as I have traced back than, it was due to the use of dapm_sync in > machine drivers. For me it looked that all other paths were satisfactory > protected. > In my case the use of SOC_DAPM_PIN_SWITCH() instead of the custom call > fixed the race (since this will take a card mutex, AFAIK). I *suspect* that we're in a similar case and either the locking has been broken by the multi-component stuff or there's other cases that need protection, probably the jack detection code. > PS: I might received this mail unintentionally, since I was not on the > TO/CC, and alsa-devel neither. > Should alsa-devel be on the CC? Yes, I bounced on the second copy - Misael had sent two copies of the mail, one omitting the CC, and I'd replied to the first one before I saw the second.

3 7

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 18 Jan '11

18 Jan '11

The following changes since commit 18b022eb117e7f70c191267551ff865f278a9258: Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-01-12 12:26:24 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 A few small bug fixes, the biggest change is the cleanup after the wl1273 merge. Alexander Sverdlin (1): ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code Dimitris Papastamos (1): ASoC: WM8990: msleep() takes milliseconds not jiffies Matti J. Aaltonen (1): ASoC: WL1273 FM radio: Fix breakage with MFD API changes sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 29 +++++++---------- sound/soc/codecs/wl1273.h | 71 ----------------------------------------- sound/soc/codecs/wm8990.c | 10 +++--- sound/soc/ep93xx/ep93xx-i2s.c | 18 +++++----- 5 files changed, 27 insertions(+), 103 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Acer Aspire 4930G
by David Henningsson 18 Jan '11

18 Jan '11

BugLink: http://bugs.launchpad.net/bugs/647970 According to the reporter, model=auto works better than the current quirk for Acer Aspire 4930G. Cc: stable(a)kernel.org (2.6.35+) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

[PATCH] ALSA: HDA: Remove quirk for Acer Aspire 4930G
by David Henningsson 18 Jan '11

18 Jan '11

BugLink: http://bugs.launchpad.net/bugs/647970 According to the reporter, model=auto works better than the current quirk for Acer Aspire 4930G. Cc: stable(a)kernel.org (2.6.35+) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f006ee..9c123d0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9842,8 +9842,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", - ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", -- 1.7.1 --------------060800040608070500030208--

1 0

Re: [alsa-devel] [PATCH 0/5] davinci: da850: clean up pinmux arrays in da850.c
by Rajashekhara, Sudhakar 18 Jan '11

18 Jan '11

Hi Sergei, On Tue, Jan 18, 2011 at 17:11:13, Sergei Shtylyov wrote: > Hello. > > On 18-01-2011 7:40, Rajashekhara, Sudhakar wrote: > > > I was testing Audio with 2.6.37 on DA850 from Kevin Hilman Linux tree > > at [1] and found that audio is broken. Below patch fixes the issue. > > > --- > > From: Rajashekhara, Sudhakar<sudhakar.raj(a)ti.com> > > > davinci: fixes for audio on da850/omap-l138/am18x > > > > On DA850/OMAP-L138/AM18x, AIC3x codec is at 0x18 slave address. > > But in sound/soc/davinci/davinci-evm.c file, "struct snd_soc_dai_link" > > has the wrong AIC3x codec slave address. This patch fixes this issue. > > > Also, this patch registers the platform device for davinci-pcm-audio. > > Shouldn't this be 2 patches? I don't see a connection between the changes. > Yes, definitely this should be 2 patches. I just posted the fix as-is to unblock any folks who are facing the issue. As you can see from my e-mail there are still some issues which I am debugging currently. I'll post the proper patch later. Regards, Sudhakar

1 0

[alsa-devel] [PATCH v2] ASoC: Add FreeScale SGTL5000 codec suppor
by Zeng Zhaoming 18 Jan '11

18 Jan '11

v1 url: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-December/034615.h… Changes since v1: - Add DAPM widges as Mark and Arnaud point out. - cleanup regulator code - new implement dai_ops functions - sort configs in Kconfig and Makefile

3 3

Re: [alsa-devel] [PATCH] ASoC: dapm: Fix race condition in widgets power list creation
by Mark Brown 18 Jan '11

18 Jan '11

On Mon, Jan 17, 2011 at 08:35:27PM -0600, Misael Lopez Cruz wrote: > Multiple calls to dapm_power_widgets() can create a race condition > causing power list to be corrupted. Those scenarios can occur in > multistream usecases, stream start/stop along with simultaneous > update power calls (mixer, mux, dapm_sync, stream_event). > A new 'dapm_mutex' is added to soc_card, which is held while power > list is created. This needs a bit more analysis - why are we getting multiple simultaneous calls to dapm_power_widgets() and is that itself safe? The ASoC locking model has always been to have a big lock around the entire card rather than to lock subcomponents, and for example this isn't going to make sure we're consistent with register map accesses from other parts of the code.

1 0

Re: [alsa-devel] [PATCH 0/5] davinci: da850: clean up pinmux arrays in da850.c
by Rajashekhara, Sudhakar 18 Jan '11

18 Jan '11

Hi, I was testing Audio with 2.6.37 on DA850 from Kevin Hilman Linux tree at [1] and found that audio is broken. Below patch fixes the issue. --- From: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> davinci: fixes for audio on da850/omap-l138/am18x On DA850/OMAP-L138/AM18x, AIC3x codec is at 0x18 slave address. But in sound/soc/davinci/davinci-evm.c file, "struct snd_soc_dai_link" has the wrong AIC3x codec slave address. This patch fixes this issue. Also, this patch registers the platform device for davinci-pcm-audio. Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- arch/arm/mach-davinci/devices-da8xx.c | 12 ++++++++++++ sound/soc/davinci/davinci-evm.c | 2 +- 2 files changed, 13 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index 9eec630..17c0dbc 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -473,6 +473,11 @@ static struct resource da850_mcasp_resources[] = { }, }; +struct platform_device davinci_pcm_device = { + .name = "davinci-pcm-audio", + .id = -1, +}; + static struct platform_device da850_mcasp_device = { .name = "davinci-mcasp", .id = 0, @@ -480,8 +485,15 @@ static struct platform_device da850_mcasp_device = { .resource = da850_mcasp_resources, }; +static void davinci_init_pcm(void) +{ + platform_device_register(&davinci_pcm_device); +} + void __init da8xx_register_mcasp(int id, struct snd_platform_data *pdata) { + davinci_init_pcm(); + /* DA830/OMAP-L137 has 3 instances of McASP */ if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index bc9e6b0..07db881 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -224,7 +224,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, --- Also, I found that either CONFIG_REGULATOR should not be defined or if CONFIG_REGULATOR is defined then CONFIG_REGULATOR_DUMMY should also be defined. Without this menuconfig fix, Soundcard does not get detected. With the above fixes, arecord and aplay does not work for the first time. Couple of times I get the below error: root@da850-omapl138-evm:~# arecord -f dat | aplay -f dat arecord: main:608: audio open error: Invalid argument aplay: playback:2297: read error root@da850-omapl138-evm:~# arecord -f dat | aplay -f dat aplay: main:608: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Third time arecord and aplay work normally. Has anyone seen such issues on DA850 or any other platform? I am currently debugging this issue. I'll submit the above patch to community once the issue of fixed. [1] http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=summa… Regards, Sudhakar

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[alsa-devel] [PATCH] ASoC: dapm: Fix race condition in widgets power list creation
by Misael Lopez Cruz 18 Jan '11

18 Jan '11

Multiple calls to dapm_power_widgets() can create a race condition causing power list to be corrupted. Those scenarios can occur in multistream usecases, stream start/stop along with simultaneous update power calls (mixer, mux, dapm_sync, stream_event). A new 'dapm_mutex' is added to soc_card, which is held while power list is created. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 2 ++ 3 files changed, 4 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7e65b01..3dfc608 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -614,6 +614,7 @@ struct snd_soc_card { struct list_head list; struct mutex mutex; + struct mutex dapm_mutex; bool instantiated; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1dc4b11..583a821 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3124,6 +3124,7 @@ static int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->list); card->instantiated = 0; mutex_init(&card->mutex); + mutex_init(&card->dapm_mutex); mutex_lock(&client_mutex); list_add(&card->list, &card_list); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3d310af..faee456 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -940,6 +940,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ + mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &dapm->widgets, list) { switch (w->id) { case snd_soc_dapm_pre: @@ -974,6 +975,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; } } + mutex_unlock(&card->dapm_mutex); /* If there are no DAPM widgets then try to figure out power from the * event type. -- 1.6.3.3 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ALSA: Sound barely audible since 2.6.37-rc1+
by Avinash Kurup 17 Jan '11

17 Jan '11

Hi Takashi, The sound on my computer since 2.6.37-rc1+ is barely audible (extremely soft), The sound remains the same even on increasing it on the desktop or the application. On accessing alsa-mixer I get the error, cannot load mixer controls: Invalid argument. I bisected the problem to this commit. 03b7a1ab557efe34e8f79b78660e514bd7374248 is the first bad commit commit 03b7a1ab557efe34e8f79b78660e514bd7374248 Author: David Henningsson <david.henningsson(a)canonical.com> Date: Tue Nov 9 14:35:30 2010 +0100 ALSA: HDA: Create mixers on ALC887 BugLink: http://launchpad.net/bugs/669092 ALC887 does not have any volume control ability on the mixer NIDs, so put the volume controls on the dac NIDs instead. Without this patch, ALC887 users cannot use alsamixer at all. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 f1e34cf6301ebd2ee2108af5195e2786c135e348 1e5d012759d201af4102a822dd94328080aa3a0b M sound Other related info... ---------------------- avinash@neurolink:~/GIT_KERNEL_REPO/linux-2.6$ uname -a Linux neurolink 2.6.37-rc1+ #164 SMP Mon Jan 17 20:15:43 IST 2011 x86_64 GNU/Linux ----------------------- 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) Subsystem: ASUSTeK Computer Inc. Device 840b Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 41 Region 0: Memory at feaf8000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel ----------------------- [ 0.705574] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 0.705614] HDA Intel 0000:00:1b.0: irq 41 for MSI/MSI-X [ 0.705631] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 0.718522] hda_codec: ALC887: BIOS auto-probing. [ 0.723019] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 0.723065] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 0.723299] ALSA device list: [ 0.723300] #0: HDA Intel at 0xfeaf8000 irq 41 [ 11.871675] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 11.873023] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 11.876868] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 11.877050] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.194611] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.198401] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.202090] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.205695] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.209255] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.213087] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.216947] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.220568] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.224190] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.227777] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.231447] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.235007] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.238700] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.243109] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.246691] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.251026] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.254755] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.258345] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.262087] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.265796] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.269377] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.273008] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.276692] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.280294] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.285131] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.288754] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.292446] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.296148] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.299781] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.303378] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.306921] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.310713] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.314817] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.318398] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.322047] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.326940] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.330627] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.334215] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.337925] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.341670] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.345255] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.348877] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.352547] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.356359] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.356549] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.360214] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 12.360405] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 15.192801] EXT4-fs (sda9): re-mounted. Opts: errors=remount-ro [ 15.483245] EXT4-fs (sda8): mounted filesystem without journal. Opts: (null) [ 15.855082] EXT4-fs (sda10): mounted filesystem with ordered data mode. Opts: (null) [ 16.148277] EXT4-fs (sda11): mounted filesystem with ordered data mode. Opts: (null) [ 16.462206] EXT4-fs (sda12): mounted filesystem with ordered data mode. Opts: (null) [ 16.572061] EXT4-fs (sda13): mounted filesystem with ordered data mode. Opts: (null) [ 16.617379] EXT4-fs (sda14): mounted filesystem with ordered data mode. Opts: (null) [ 18.412393] ATL1E 0000:01:00.0: irq 42 for MSI/MSI-X [ 18.412895] ADDRCONF(NETDEV_UP): eth0: link is not ready [ 18.414087] ATL1E 0000:01:00.0: eth0: NIC Link is Up <100 Mbps Full Duplex> [ 18.414493] ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [ 20.964039] uhci_hcd 0000:00:1d.1: reserve dev 2 ep81-INT, period 8, phase 4, 93 us [ 22.398020] netconsole: network logging stopped, interface sit0 unregistered [ 22.409016] netconsole: network logging stopped, interface lo unregistered [ 25.975782] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 26.036097] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 29.410004] eth0: no IPv6 routers present [ 38.350974] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) [ 38.365301] hda_codec: num_steps = 0 for NID=0x2 (ctl = Front Playback Volume) Please let me know if u need any more info...Any help would be appreciated. Thanks, Avinash.

3 2

[alsa-devel] Application output to ALSA looks like capture device
by George Morgan 17 Jan '11

17 Jan '11

Is there a way to configure a system without actual sound hardware to either have a virtual sound card or to create an application whose output looks like a virtual sound capture device to another ALSA or OSS application? Thanks. George Morgan

1 0

[alsa-devel] [Patch] Kill off warning: ‘inline’ is not at beginning of declaration
by Jesper Juhl 17 Jan '11

17 Jan '11

Fix a bunch of warning: ‘inline’ is not at beginning of declaration messages when building a 'make allyesconfig' kernel with -Wextra. These warnings are trivial to kill, yet rather annoying when building with -Wextra. The more we can cut down on pointless crap like this the better (IMHO). A previous patch to do this for a 'allnoconfig' build has already been merged. This just takes the cleanup a little further. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- arch/x86/oprofile/op_model_p4.c | 2 +- drivers/bluetooth/btusb.c | 4 ++-- drivers/cpuidle/sysfs.c | 2 +- drivers/edac/i7300_edac.c | 2 +- fs/ocfs2/dir.c | 2 +- kernel/trace/ring_buffer.c | 2 +- net/ipv6/inet6_hashtables.c | 2 +- net/mac80211/tx.c | 2 +- sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_core.c | 4 ++-- 10 files changed, 13 insertions(+), 13 deletions(-) diff --git a/arch/x86/oprofile/op_model_p4.c b/arch/x86/oprofile/op_model_p4.c index 9fadec0..98ab130 100644 --- a/arch/x86/oprofile/op_model_p4.c +++ b/arch/x86/oprofile/op_model_p4.c @@ -50,7 +50,7 @@ static inline void setup_num_counters(void) #endif } -static int inline addr_increment(void) +static inline int addr_increment(void) { #ifdef CONFIG_SMP return smp_num_siblings == 2 ? 2 : 1; diff --git a/drivers/bluetooth/btusb.c b/drivers/bluetooth/btusb.c index 1da773f..92d29bf 100644 --- a/drivers/bluetooth/btusb.c +++ b/drivers/bluetooth/btusb.c @@ -424,7 +424,7 @@ static void btusb_isoc_complete(struct urb *urb) } } -static void inline __fill_isoc_descriptor(struct urb *urb, int len, int mtu) +static inline void __fill_isoc_descriptor(struct urb *urb, int len, int mtu) { int i, offset = 0; @@ -775,7 +775,7 @@ static void btusb_notify(struct hci_dev *hdev, unsigned int evt) } } -static int inline __set_isoc_interface(struct hci_dev *hdev, int altsetting) +static inline int __set_isoc_interface(struct hci_dev *hdev, int altsetting) { struct btusb_data *data = hdev->driver_data; struct usb_interface *intf = data->isoc; diff --git a/drivers/cpuidle/sysfs.c b/drivers/cpuidle/sysfs.c index 0310ffa..be7917ec 100644 --- a/drivers/cpuidle/sysfs.c +++ b/drivers/cpuidle/sysfs.c @@ -300,7 +300,7 @@ static struct kobj_type ktype_state_cpuidle = { .release = cpuidle_state_sysfs_release, }; -static void inline cpuidle_free_state_kobj(struct cpuidle_device *device, int i) +static inline void cpuidle_free_state_kobj(struct cpuidle_device *device, int i) { kobject_put(&device->kobjs[i]->kobj); wait_for_completion(&device->kobjs[i]->kobj_unregister); diff --git a/drivers/edac/i7300_edac.c b/drivers/edac/i7300_edac.c index 05523b5..76d1f57 100644 --- a/drivers/edac/i7300_edac.c +++ b/drivers/edac/i7300_edac.c @@ -162,7 +162,7 @@ static struct edac_pci_ctl_info *i7300_pci; #define AMBPRESENT_0 0x64 #define AMBPRESENT_1 0x66 -const static u16 mtr_regs[MAX_SLOTS] = { +static const u16 mtr_regs[MAX_SLOTS] = { 0x80, 0x84, 0x88, 0x8c, 0x82, 0x86, 0x8a, 0x8e }; diff --git a/fs/ocfs2/dir.c b/fs/ocfs2/dir.c index d417b3f..f97b6f1 100644 --- a/fs/ocfs2/dir.c +++ b/fs/ocfs2/dir.c @@ -354,7 +354,7 @@ static inline int ocfs2_match(int len, /* * Returns 0 if not found, -1 on failure, and 1 on success */ -static int inline ocfs2_search_dirblock(struct buffer_head *bh, +static inline int ocfs2_search_dirblock(struct buffer_head *bh, struct inode *dir, const char *name, int namelen, unsigned long offset, diff --git a/kernel/trace/ring_buffer.c b/kernel/trace/ring_buffer.c index bd1c35a..6ee56b4 100644 --- a/kernel/trace/ring_buffer.c +++ b/kernel/trace/ring_buffer.c @@ -669,7 +669,7 @@ static struct list_head *rb_list_head(struct list_head *list) * the reader page). But if the next page is a header page, * its flags will be non zero. */ -static int inline +static inline int rb_is_head_page(struct ring_buffer_per_cpu *cpu_buffer, struct buffer_page *page, struct list_head *list) { diff --git a/net/ipv6/inet6_hashtables.c b/net/ipv6/inet6_hashtables.c index 633a6c2..b531972 100644 --- a/net/ipv6/inet6_hashtables.c +++ b/net/ipv6/inet6_hashtables.c @@ -124,7 +124,7 @@ out: } EXPORT_SYMBOL(__inet6_lookup_established); -static int inline compute_score(struct sock *sk, struct net *net, +static inline int compute_score(struct sock *sk, struct net *net, const unsigned short hnum, const struct in6_addr *daddr, const int dif) diff --git a/net/mac80211/tx.c b/net/mac80211/tx.c index 5950e3a..a449dd5 100644 --- a/net/mac80211/tx.c +++ b/net/mac80211/tx.c @@ -173,7 +173,7 @@ static __le16 ieee80211_duration(struct ieee80211_tx_data *tx, int group_addr, return cpu_to_le16(dur); } -static int inline is_ieee80211_device(struct ieee80211_local *local, +static inline int is_ieee80211_device(struct ieee80211_local *local, struct net_device *dev) { return local == wdev_priv(dev->ieee80211_ptr); diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index cf46bba..ecb8f4d 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -211,7 +211,7 @@ static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma); //static void vortex_adbdma_stopfifo(vortex_t *vortex, int adbdma); static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma); static void vortex_adbdma_resumefifo(vortex_t * vortex, int adbdma); -static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma); +static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma); static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma); #ifndef CHIP_AU8810 @@ -219,7 +219,7 @@ static void vortex_wtdma_startfifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_stopfifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_pausefifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_resumefifo(vortex_t * vortex, int wtdma); -static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma); +static inline int vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma); #endif /* global stuff. */ diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 23f49f3..d43252a 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1249,7 +1249,7 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { } } -static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) +static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; int temp; @@ -1498,7 +1498,7 @@ static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) POS_SHIFT) & POS_MASK); } #endif -static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) +static inline int vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) { stream_t *dma = &vortex->dma_wt[wtdma]; int temp; -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/~esr/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [PATCH] ASoC: CS4271 codec support
by Alexander 17 Jan '11

17 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for CS4271 codec to ASoC. Update: De-emphasis is managed automatically with on-off control. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- include/sound/cs4271.h | 25 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4271.c | 641 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 672 insertions(+), 0 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h new file mode 100644 index 0000000..16f8d32 --- /dev/null +++ b/include/sound/cs4271.h @@ -0,0 +1,25 @@ +/* + * Definitions for CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + */ + +#ifndef __CS4271_H +#define __CS4271_H + +struct cs4271_platform_data { + int gpio_nreset; /* GPIO driving Reset pin, if any */ + int gpio_disable; /* GPIO that disable serial bus, if any */ +}; + +#endif /* __CS4271_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c48b23c..c3056bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 @@ -155,6 +156,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CS4271 + tristate + config SND_SOC_CX20442 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 579af9c..ece05e1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o @@ -91,6 +92,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c new file mode 100644 index 0000000..4d11e89 --- /dev/null +++ b/sound/soc/codecs/cs4271.c @@ -0,0 +1,641 @@ +/* + * CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/cs4271.h> + +#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * CS4271 registers + * High byte represents SPI chip address (0x10) + write command (0) + * Low byte - codec register address + */ +#define CS4271_MODE1 0x2001 /* Mode Control 1 */ +#define CS4271_DACCTL 0x2002 /* DAC Control */ +#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x2006 /* ADC Control */ +#define CS4271_MODE2 0x2007 /* Mode Control 2 */ +#define CS4271_CHIPID 0x2008 /* Chip ID */ + +#define CS4271_FIRSTREG CS4271_MODE1 +#define CS4271_LASTREG CS4271_MODE2 +#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1) + +/* Bit masks for the CS4271 registers */ +#define CS4271_MODE1_MODE_MASK 0xC0 +#define CS4271_MODE1_MODE_1X 0x00 +#define CS4271_MODE1_MODE_2X 0x80 +#define CS4271_MODE1_MODE_4X 0xC0 + +#define CS4271_MODE1_DIV_MASK 0x30 +#define CS4271_MODE1_DIV_1 0x00 +#define CS4271_MODE1_DIV_15 0x10 +#define CS4271_MODE1_DIV_2 0x20 +#define CS4271_MODE1_DIV_3 0x30 + +#define CS4271_MODE1_MASTER 0x08 + +#define CS4271_MODE1_DAC_DIF_MASK 0x07 +#define CS4271_MODE1_DAC_DIF_LJ 0x00 +#define CS4271_MODE1_DAC_DIF_I2S 0x01 +#define CS4271_MODE1_DAC_DIF_RJ16 0x02 +#define CS4271_MODE1_DAC_DIF_RJ24 0x03 +#define CS4271_MODE1_DAC_DIF_RJ20 0x04 +#define CS4271_MODE1_DAC_DIF_RJ18 0x05 + +#define CS4271_DACCTL_AMUTE 0x80 +#define CS4271_DACCTL_IF_SLOW 0x40 + +#define CS4271_DACCTL_DEM_MASK 0x30 +#define CS4271_DACCTL_DEM_DIS 0x00 +#define CS4271_DACCTL_DEM_441 0x10 +#define CS4271_DACCTL_DEM_48 0x20 +#define CS4271_DACCTL_DEM_32 0x30 + +#define CS4271_DACCTL_SVRU 0x08 +#define CS4271_DACCTL_SRD 0x04 +#define CS4271_DACCTL_INVA 0x02 +#define CS4271_DACCTL_INVB 0x01 + +#define CS4271_DACVOL_BEQUA 0x40 +#define CS4271_DACVOL_SOFT 0x20 +#define CS4271_DACVOL_ZEROC 0x10 + +#define CS4271_DACVOL_ATAPI_MASK 0x0F +#define CS4271_DACVOL_ATAPI_M_M 0x00 +#define CS4271_DACVOL_ATAPI_M_BR 0x01 +#define CS4271_DACVOL_ATAPI_M_BL 0x02 +#define CS4271_DACVOL_ATAPI_M_BLR2 0x03 +#define CS4271_DACVOL_ATAPI_AR_M 0x04 +#define CS4271_DACVOL_ATAPI_AR_BR 0x05 +#define CS4271_DACVOL_ATAPI_AR_BL 0x06 +#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07 +#define CS4271_DACVOL_ATAPI_AL_M 0x08 +#define CS4271_DACVOL_ATAPI_AL_BR 0x09 +#define CS4271_DACVOL_ATAPI_AL_BL 0x0A +#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B +#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C +#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D +#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E +#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F + +#define CS4271_VOLA_MUTE 0x80 +#define CS4271_VOLA_VOL_MASK 0x7F +#define CS4271_VOLB_MUTE 0x80 +#define CS4271_VOLB_VOL_MASK 0x7F + +#define CS4271_ADCCTL_DITHER16 0x20 + +#define CS4271_ADCCTL_ADC_DIF_MASK 0x10 +#define CS4271_ADCCTL_ADC_DIF_LJ 0x00 +#define CS4271_ADCCTL_ADC_DIF_I2S 0x10 + +#define CS4271_ADCCTL_MUTEA 0x08 +#define CS4271_ADCCTL_MUTEB 0x04 +#define CS4271_ADCCTL_HPFDA 0x02 +#define CS4271_ADCCTL_HPFDB 0x01 + +#define CS4271_MODE2_LOOP 0x10 +#define CS4271_MODE2_MUTECAEQUB 0x08 +#define CS4271_MODE2_FREEZE 0x04 +#define CS4271_MODE2_CPEN 0x02 +#define CS4271_MODE2_PDN 0x01 + +#define CS4271_CHIPID_PART_MASK 0xF0 +#define CS4271_CHIPID_REV_MASK 0x0F + +/* CS4271 default register settings, except auto-mute is off */ +static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { + 0, 0, 0, CS4271_DACVOL_ATAPI_AL_BR, 0, 0, 0, + CS4271_MODE2_CPEN | CS4271_MODE2_PDN, +}; + +struct cs4271_private { + /* SND_SOC_I2C or SND_SOC_SPI */ + enum snd_soc_control_type bus_type; + void *control_data; + unsigned int mclk; + /* SND_SOC_DAIFMT_I2S or SND_SOC_DAIFMT_LEFT_J */ + unsigned int mode; + unsigned int master; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; + /* GPIO that disable serial bus, if any */ + int gpio_disable; + bool deemph; + int rate; +}; + +struct cs4271_clk_cfg { + unsigned int ratio; /* MCLK / sample rate */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + u8 mclk_master; /* ratio bit mask for Master mode */ + u8 mclk_slave; /* ratio bit mask for Slave mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, + {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, + {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + +/* + * @freq is the desired MCLK rate + * MCLK rate should (c) be the sample rate, multiplied by one of the + * ratios listed in cs4271_mclk_fs_ratios table + */ +static int cs4271_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mclk = freq; + return 0; +} + +static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4271->master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4271->master = 1; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int cs4271_deemph[] = {0, 44100, 48000, 32000}; + +static int cs4271_set_deemph(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int i; + int val = CS4271_DACCTL_DEM_DIS; + + if (cs4271->deemph) { + /* Find closest de-emphasis freq */ + val = 1; + for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++) + if (abs(cs4271_deemph[i] - cs4271->rate) < + abs(cs4271_deemph[val] - cs4271->rate)) + val = i; + val <<= 4; + } + + return snd_soc_update_bits(codec, CS4271_DACCTL, + CS4271_DACCTL_DEM_MASK, val); +} + +static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = cs4271->deemph; + return 0; +} + +static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->deemph = ucontrol->value.enumerated.item[0]; + return cs4271_set_deemph(codec); +} + +static int cs4271_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 cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int i, ratio, val; + + cs4271->rate = params_rate(params); + ratio = cs4271->mclk / cs4271->rate; + for (i = 0; i < CS4171_NR_RATIOS; i++) + if (cs4271_clk_tab[i].ratio == ratio) + break; + + if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + /* Configure DAC */ + val = cs4271_clk_tab[i].speed_mode; + + if (cs4271->master) + val |= cs4271_clk_tab[i].mclk_master | CS4271_MODE1_MASTER; + else + val |= cs4271_clk_tab[i].mclk_slave; + + switch (cs4271->mode) { + case SND_SOC_DAIFMT_LEFT_J: + val |= CS4271_MODE1_DAC_DIF_LJ; + break; + case SND_SOC_DAIFMT_I2S: + val |= CS4271_MODE1_DAC_DIF_I2S; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK | + CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); + + /* Configure ADC */ + val = (cs4271->mode == SND_SOC_DAIFMT_I2S) ? + CS4271_ADCCTL_ADC_DIF_I2S : CS4271_ADCCTL_ADC_DIF_LJ; + snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, + val); + + return cs4271_set_deemph(codec); +} + +static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int val_a = 0; + int val_b = 0; + + if (mute) { + val_a = CS4271_VOLA_MUTE; + val_b = CS4271_VOLB_MUTE; + } + + snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + return 0; +} + +/* CS4271 controls */ +static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0); + +static const struct snd_kcontrol_new cs4271_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB, + 0, 0x7F, 1, cs4271_dac_tlv), + SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + cs4271_get_deemph, cs4271_put_deemph), + SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0), + SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0), + SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0), + SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0), + SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1), + SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0), + SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1), + SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB, + 7, 1, 1), +}; + +static struct snd_soc_dai_ops cs4271_dai_ops = { + .hw_params = cs4271_hw_params, + .set_sysclk = cs4271_set_dai_sysclk, + .set_fmt = cs4271_set_dai_fmt, + .digital_mute = cs4271_digital_mute, +}; + +struct snd_soc_dai_driver cs4271_dai = { + .name = "cs4271-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .ops = &cs4271_dai_ops, + .symmetric_rates = 1, +}; + +/* + * This function writes all writeble registers from cache to codec. + * It's used to setup initial config and restore after suspend. + * The registers write order is essential (MODE1 written last), + * so we cannot use standard sync implementation. + */ +static int cs4271_write_cache(struct snd_soc_codec *codec) +{ + int i, ret; + + for (i = CS4271_LASTREG; i >= CS4271_FIRSTREG; i--) { + ret = snd_soc_write(codec, i, snd_soc_read(codec, i)); + if (ret) { + dev_err(codec->dev, "Cache write failed\n"); + return ret; + } + } + + return 0; +} + +#ifdef CONFIG_PM +static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + /* Set power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + + return 0; +} +#else +#define cs4271_soc_suspend NULL +#endif /* CONFIG_PM */ + +/* This function used also in codec probe function and not only for PM */ +static int cs4271_soc_resume(struct snd_soc_codec *codec) +{ + int ret; + + /* Restore codec state */ + ret = cs4271_write_cache(codec); + if (ret < 0) + return ret; + + /* then disable the power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + + return 0; +} + +static int cs4271_probe(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; + int ret; + int gpio_nreset = -EINVAL; + int gpio_disable = -EINVAL; + + codec->control_data = cs4271->control_data; + + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + if (gpio_is_valid(cs4271plat->gpio_disable)) + gpio_disable = cs4271plat->gpio_disable; + } + + if (gpio_disable >= 0) + if (gpio_request(gpio_disable, "CS4271 Disable")) + gpio_disable = -EINVAL; + if (gpio_disable >= 0) + gpio_direction_output(gpio_disable, 0); + + if (gpio_nreset >= 0) + if (gpio_request(gpio_nreset, "CS4271 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + gpio_direction_output(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + } + + cs4271->gpio_nreset = gpio_nreset; + cs4271->gpio_disable = gpio_disable; + + /* + * In case of I2C, chip address specified in board data. + * So cache IO operations use 8 bit codec register address. + * In case of SPI, chip address and register address + * passed together as 16 bit value. + * Anyway, register address is masked with 0xFF inside + * soc-cache code. + */ + ret = (cs4271->bus_type == SND_SOC_SPI) ? 16 : 8; + ret = snd_soc_codec_set_cache_io(codec, ret, 8, cs4271->bus_type); + if (ret) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = cs4271_soc_resume(codec); + if (ret < 0) + return ret; + + return snd_soc_add_controls(codec, cs4271_snd_controls, + ARRAY_SIZE(cs4271_snd_controls)); +} + +static int cs4271_remove(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int gpio_nreset, gpio_disable; + + gpio_nreset = cs4271->gpio_nreset; + gpio_disable = cs4271->gpio_disable; + + if (gpio_is_valid(gpio_nreset)) { + /* Set codec to the reset state */ + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + } + + if (gpio_is_valid(gpio_disable)) + gpio_free(gpio_disable); + + return 0; +}; + +struct snd_soc_codec_driver soc_codec_dev_cs4271 = { + .probe = cs4271_probe, + .remove = cs4271_remove, + .suspend = cs4271_soc_suspend, + .resume = cs4271_soc_resume, + .reg_cache_default = &cs4271_dflt_reg, + .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), + .reg_word_size = sizeof(cs4271_dflt_reg[0]), + .compress_type = SND_SOC_FLAT_COMPRESSION, + .reg_cache_step = 1, +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + spi_set_drvdata(spi, cs4271); + cs4271->control_data = spi; + cs4271->bus_type = SND_SOC_SPI; + + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .probe = cs4271_spi_probe, + .remove = __devexit_p(cs4271_spi_remove), +}; +#endif /* defined(CONFIG_SPI_MASTER) */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_device_id cs4271_i2c_id[] = { + {"cs4271", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static int __devinit cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + i2c_set_clientdata(client, cs4271); + cs4271->control_data = client; + cs4271->bus_type = SND_SOC_I2C; + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .id_table = cs4271_i2c_id, + .probe = cs4271_i2c_probe, + .remove = __devexit_p(cs4271_i2c_remove), +}; +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +/* + * We only register our serial bus driver here without + * assignment to particular chip. So if any of the below + * fails, there is some problem with I2C or SPI subsystem. + * In most cases this module will be compiled with support + * of only one serial bus. + */ +static int __init cs4271_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&cs4271_i2c_driver); + if (ret) { + pr_err("Failed to register CS4271 I2C driver: %d\n", ret); + return ret; + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4271_spi_driver); + if (ret) { + pr_err("Failed to register CS4271 SPI driver: %d\n", ret); + return ret; + } +#endif + + return 0; +} +module_init(cs4271_modinit); + +static void __exit cs4271_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4271_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&cs4271_i2c_driver); +#endif +} +module_exit(cs4271_modexit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL");

3 3

[alsa-devel] [PATCH] ASoC: WL1273 FM radio: Access I2C IO functions through pointers.
by Matti J. Aaltonen 17 Jan '11

17 Jan '11

These changes are needed to keep up with the changes in the MFD core and V4L2 parts of the wl1273 FM radio driver. Use function pointers instead of exported functions for I2C IO. Also move all preprocessor constants from the wl1273.h to include/linux/mfd/wl1273-core.h. Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen(a)nokia.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 29 +++++++----------- sound/soc/codecs/wl1273.h | 71 --------------------------------------------- 3 files changed, 13 insertions(+), 89 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3b5690d..937517a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,7 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C - select SND_SOC_WL1273 if WL1273_CORE + select SND_SOC_WL1273 if RADIO_WL1273 select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 0c47c78..4c22ae9 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -43,7 +43,7 @@ struct wl1273_priv { static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, int rate, int width) { - struct device *dev = &core->i2c_dev->dev; + struct device *dev = &core->client->dev; int r = 0; u16 mode; @@ -124,13 +124,13 @@ static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, dev_dbg(dev, "mode: 0x%04x\n", mode); if (core->i2s_mode != mode) { - r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode); + r = core->write(core, WL1273_I2S_MODE_CONFIG_SET, mode); if (r) goto out; core->i2s_mode = mode; - r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE, - WL1273_AUDIO_ENABLE_I2S); + r = core->write(core, WL1273_AUDIO_ENABLE, + WL1273_AUDIO_ENABLE_I2S); if (r) goto out; } @@ -143,8 +143,7 @@ out: static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, int channel_number) { - struct i2c_client *client = core->i2c_dev; - struct device *dev = &client->dev; + struct device *dev = &core->client->dev; int r = 0; dev_dbg(dev, "%s\n", __func__); @@ -155,17 +154,13 @@ static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, goto out; if (channel_number == 1 && core->mode == WL1273_MODE_RX) - r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, - WL1273_RX_MONO); + r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_MONO); else if (channel_number == 1 && core->mode == WL1273_MODE_TX) - r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, - WL1273_TX_MONO); + r = core->write(core, WL1273_MONO_SET, WL1273_TX_MONO); else if (channel_number == 2 && core->mode == WL1273_MODE_RX) - r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, - WL1273_RX_STEREO); + r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_STEREO); else if (channel_number == 2 && core->mode == WL1273_MODE_TX) - r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, - WL1273_TX_STEREO); + r = core->write(core, WL1273_MONO_SET, WL1273_TX_STEREO); else r = -EINVAL; out: @@ -238,7 +233,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, if (wl1273->core->audio_mode == val) return 0; - r = wl1273_fm_set_audio(wl1273->core, val); + r = wl1273->core->set_audio(wl1273->core, val); if (r < 0) return r; @@ -273,8 +268,8 @@ static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol, dev_dbg(codec->dev, "%s: enter.\n", __func__); - r = wl1273_fm_set_volume(wl1273->core, - ucontrol->value.integer.value[0]); + r = wl1273->core->set_volume(wl1273->core, + ucontrol->value.integer.value[0]); if (r) return r; diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h index 14ed027..43ec7e6 100644 --- a/sound/soc/codecs/wl1273.h +++ b/sound/soc/codecs/wl1273.h @@ -25,77 +25,6 @@ #ifndef __WL1273_CODEC_H__ #define __WL1273_CODEC_H__ -/* I2S protocol, left channel first, data width 16 bits */ -#define WL1273_PCM_DEF_MODE 0x00 - -/* Rx */ -#define WL1273_AUDIO_ENABLE_I2S (1 << 0) -#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1) - -/* Tx */ -#define WL1273_AUDIO_IO_SET_ANALOG 0 -#define WL1273_AUDIO_IO_SET_I2S 1 - -#define WL1273_POWER_SET_OFF 0 -#define WL1273_POWER_SET_FM (1 << 0) -#define WL1273_POWER_SET_RDS (1 << 1) -#define WL1273_POWER_SET_RETENTION (1 << 4) - -#define WL1273_PUPD_SET_OFF 0x00 -#define WL1273_PUPD_SET_ON 0x01 -#define WL1273_PUPD_SET_RETENTION 0x10 - -/* I2S mode */ -#define WL1273_IS2_WIDTH_32 0x0 -#define WL1273_IS2_WIDTH_40 0x1 -#define WL1273_IS2_WIDTH_22_23 0x2 -#define WL1273_IS2_WIDTH_23_22 0x3 -#define WL1273_IS2_WIDTH_48 0x4 -#define WL1273_IS2_WIDTH_50 0x5 -#define WL1273_IS2_WIDTH_60 0x6 -#define WL1273_IS2_WIDTH_64 0x7 -#define WL1273_IS2_WIDTH_80 0x8 -#define WL1273_IS2_WIDTH_96 0x9 -#define WL1273_IS2_WIDTH_128 0xa -#define WL1273_IS2_WIDTH 0xf - -#define WL1273_IS2_FORMAT_STD (0x0 << 4) -#define WL1273_IS2_FORMAT_LEFT (0x1 << 4) -#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4) -#define WL1273_IS2_FORMAT_USER (0x3 << 4) - -#define WL1273_IS2_MASTER (0x0 << 6) -#define WL1273_IS2_SLAVEW (0x1 << 6) - -#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7) -#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7) - -#define WL1273_IS2_SDOWS_RR (0x0 << 8) -#define WL1273_IS2_SDOWS_RF (0x1 << 8) -#define WL1273_IS2_SDOWS_FR (0x2 << 8) -#define WL1273_IS2_SDOWS_FF (0x3 << 8) - -#define WL1273_IS2_TRI_OPT (0x0 << 10) -#define WL1273_IS2_TRI_ALWAYS (0x1 << 10) - -#define WL1273_IS2_RATE_48K (0x0 << 12) -#define WL1273_IS2_RATE_44_1K (0x1 << 12) -#define WL1273_IS2_RATE_32K (0x2 << 12) -#define WL1273_IS2_RATE_22_05K (0x4 << 12) -#define WL1273_IS2_RATE_16K (0x5 << 12) -#define WL1273_IS2_RATE_12K (0x8 << 12) -#define WL1273_IS2_RATE_11_025 (0x9 << 12) -#define WL1273_IS2_RATE_8K (0xa << 12) -#define WL1273_IS2_RATE (0xf << 12) - -#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \ - WL1273_IS2_FORMAT_STD | \ - WL1273_IS2_MASTER | \ - WL1273_IS2_TRI_AFTER_SENDING | \ - WL1273_IS2_SDOWS_RR | \ - WL1273_IS2_TRI_OPT | \ - WL1273_IS2_RATE_48K) - int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt); #endif /* End of __WL1273_CODEC_H__ */ -- 1.6.1.3

3 12

[alsa-devel] [HDA - Add support for VMware HDA Controller 1/1]
by Bankim Bhavsar 17 Jan '11

17 Jan '11

[ALSA]: HDA - Add support for VMware controller. Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio Controller. Signed-off-by: Bankim Bhavsar <bbhavsar(a)vmware.com> diff --git a/pci/hda/hda_intel.c b/pci/hda/hda_intel.c index d3d18be..d9394d8 100644 --- a/pci/hda/hda_intel.c +++ b/pci/hda/hda_intel.c @@ -459,6 +459,7 @@ enum { AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, AZX_DRIVER_CTX, + AZX_DRIVER_VMWARE, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -475,6 +476,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_NVIDIA] = "HDA NVidia", [AZX_DRIVER_TERA] = "HDA Teradici", [AZX_DRIVER_CTX] = "HDA Creative", + [AZX_DRIVER_VMWARE] = "HDA VMware", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -2793,6 +2795,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, + /* VMware HDAudio */ + { PCI_DEVICE(0x15ad, 0x1977), .driver_data = AZX_DRIVER_VMWARE }, /* Creative X-Fi (CA0110-IBG) */ #if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) /* the following entry conflicts with snd-ctxfi driver, -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
by Alexander 17 Jan '11

17 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Changes to I2S code: - MCLK is turned on on driver probe. This is done for several codecs that need this for registers access (like CS4271). - SCLK and LRCLK rates are corrected, assuming we always send 32 bits * 2 channels to codec. - Formats list shortened to just S32_LE, this makes all the DMA transactions right, while ALSA will do all sample format translation for us. Changes to both I2S and PCM code: - Rates list extended up to 96kHz, it's tested on EDB9302 and works for both capture and playback. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 37 +++++++++++++++++++++++-------------- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 25 insertions(+), 16 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 4f48733..e5e9b9a 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -98,7 +98,6 @@ static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { /* Enable clocks */ - clk_enable(info->mclk); clk_enable(info->sclk); clk_enable(info->lrclk); @@ -136,7 +135,6 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) /* Disable clocks */ clk_disable(info->lrclk); clk_disable(info->sclk); - clk_disable(info->mclk); } } @@ -267,14 +265,16 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); /* - * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values. - * If the lrclk is pulse length is larger than the word size, then the - * bit clock will be gated for the unused bits. + * EP93xx I2S module can be setup so SCLK / LRCLK value can be + * 32, 64, 128. MCLK / SCLK value can be 2 and 4. + * We set LRCLK equal to `rate' and minimum SCLK / LRCLK + * value is 64, because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. */ - div = (clk_get_rate(info->mclk) / params_rate(params)) * - params_channels(params); + div = clk_get_rate(info->mclk) / params_rate(params); for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1) + for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) if (sdiv * lrdiv == div) { found = 1; goto out; @@ -316,6 +316,7 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + clk_disable(info->mclk); } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -325,6 +326,7 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) if (!dai->active) return; + clk_enable(info->mclk); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); } @@ -341,9 +343,7 @@ static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .set_fmt = ep93xx_i2s_set_dai_fmt, }; -#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver ep93xx_i2s_dai = { .symmetric_rates= 1, @@ -352,13 +352,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_96000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_96000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, @@ -404,10 +404,16 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) goto fail_unmap_mem; } + /* Minimal MCLK freq to enable some codecs */ + err = clk_set_rate(info->mclk, 8000 * 64 * 4); + if (err) + goto fail_put_mclk; + clk_enable(info->mclk); + info->sclk = clk_get(&pdev->dev, "sclk"); if (IS_ERR(info->sclk)) { err = PTR_ERR(info->sclk); - goto fail_put_mclk; + goto fail_disable_mclk; } info->lrclk = clk_get(&pdev->dev, "lrclk"); @@ -426,6 +432,8 @@ fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); +fail_disable_mclk: + clk_disable(info->mclk); fail_put_mclk: clk_put(info->mclk); fail_unmap_mem: @@ -444,6 +452,7 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); clk_put(info->lrclk); clk_put(info->sclk); + clk_disable(info->mclk); clk_put(info->mclk); iounmap(info->regs); release_mem_region(info->mem->start, resource_size(info->mem)); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 2f121dd..0667077 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_96000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_48000, + .rate_max = SNDRV_PCM_RATE_96000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |

3 17

[alsa-devel] [PATCH] ASoC: WM8990: msleep() takes milliseconds not jiffies
by Dimitris Papastamos 17 Jan '11

17 Jan '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5c87a63..100aeee 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1183,7 +1183,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_VMIDTOG); /* Delay to allow output caps to discharge */ - msleep(msecs_to_jiffies(300)); + msleep(300); /* Disable VMIDTOG */ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | @@ -1195,17 +1195,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* Enable outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); - msleep(msecs_to_jiffies(50)); + msleep(50); /* Enable VMID at 2x50k */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); - msleep(msecs_to_jiffies(100)); + msleep(100); /* Enable VREF */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); - msleep(msecs_to_jiffies(600)); + msleep(600); /* Enable BUFIOEN */ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | @@ -1250,7 +1250,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); - msleep(msecs_to_jiffies(300)); + msleep(300); /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: WM8995: Add regulator handling code
by Dimitris Papastamos 17 Jan '11

17 Jan '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8995.c | 102 ++++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 97 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index f0f678d..7d56341 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -30,6 +31,18 @@ #include "wm8995.h" +#define WM8995_NUM_SUPPLIES 8 +static const char *wm8995_supply_names[WM8995_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD1", + "DBVDD2", + "DBVDD3", + "AVDD1", + "AVDD2", + "CPVDD", + "MICVDD" +}; + static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = { [0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b, [24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0, @@ -126,8 +139,37 @@ struct wm8995_priv { int mclk[2]; int aifclk[2]; struct fll_config fll[2], fll_suspend[2]; + struct regulator_bulk_data supplies[WM8995_NUM_SUPPLIES]; + struct notifier_block disable_nb[WM8995_NUM_SUPPLIES]; + struct snd_soc_codec *codec; }; +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define WM8995_REGULATOR_EVENT(n) \ +static int wm8995_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct wm8995_priv *wm8995 = container_of(nb, struct wm8995_priv, \ + disable_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + wm8995->codec->cache_sync = 1; \ + } \ + return 0; \ +} + +WM8995_REGULATOR_EVENT(0) +WM8995_REGULATOR_EVENT(1) +WM8995_REGULATOR_EVENT(2) +WM8995_REGULATOR_EVENT(3) +WM8995_REGULATOR_EVENT(4) +WM8995_REGULATOR_EVENT(5) +WM8995_REGULATOR_EVENT(6) +WM8995_REGULATOR_EVENT(7) + static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(in1lr_pga_tlv, -1650, 150, 0); static const DECLARE_TLV_DB_SCALE(in1l_boost_tlv, 0, 600, 0); @@ -1483,6 +1525,11 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) + return ret; + ret = snd_soc_cache_sync(codec); if (ret) { dev_err(codec->dev, @@ -1492,13 +1539,13 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, WM8995_BG_ENA); - } break; case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, 0); - codec->cache_sync = 1; + regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); break; } @@ -1537,10 +1584,12 @@ static int wm8995_remove(struct snd_soc_codec *codec) static int wm8995_probe(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; + int i; int ret; codec->dapm.idle_bias_off = 1; wm8995 = snd_soc_codec_get_drvdata(codec); + wm8995->codec = codec; ret = snd_soc_codec_set_cache_io(codec, 16, 16, wm8995->control_type); if (ret < 0) { @@ -1548,21 +1597,58 @@ static int wm8995_probe(struct snd_soc_codec *codec) return ret; } + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) + wm8995->supplies[i].supply = wm8995_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8995->disable_nb[0].notifier_call = wm8995_regulator_event_0; + wm8995->disable_nb[1].notifier_call = wm8995_regulator_event_1; + wm8995->disable_nb[2].notifier_call = wm8995_regulator_event_2; + wm8995->disable_nb[3].notifier_call = wm8995_regulator_event_3; + wm8995->disable_nb[4].notifier_call = wm8995_regulator_event_4; + wm8995->disable_nb[5].notifier_call = wm8995_regulator_event_5; + wm8995->disable_nb[6].notifier_call = wm8995_regulator_event_6; + wm8995->disable_nb[7].notifier_call = wm8995_regulator_event_7; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) { + ret = regulator_register_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + if (ret) { + dev_err(codec->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_get; + } + ret = snd_soc_read(codec, WM8995_SOFTWARE_RESET); if (ret < 0) { dev_err(codec->dev, "Failed to read device ID: %d\n", ret); - return ret; + goto err_reg_enable; } if (ret != 0x8995) { dev_err(codec->dev, "Invalid device ID: %#x\n", ret); - return -EINVAL; + goto err_reg_enable; } ret = snd_soc_write(codec, WM8995_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - return ret; + goto err_reg_enable; } wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1597,6 +1683,12 @@ static int wm8995_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8995_intercon)); return 0; + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); +err_reg_get: + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); + return ret; } #define WM8995_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: WM8991: Add initial WM8991 driver
by Dimitris Papastamos 17 Jan '11

17 Jan '11

The WM8991 is a highly integrated ultra-low power hi-fi CODEC designed for handsets rich in multimedia features such as GPS, mobile TV, digital audio playback and gaming. This driver was originally written by Graeme Gregory and has been maintained out of tree by Mark Brown and Dimitris Papastamos. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8991.c | 1427 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8991.h | 833 ++++++++++++++++++++++++++ 4 files changed, 2266 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8991.c create mode 100644 sound/soc/codecs/wm8991.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 61e36ef..a18cff4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -77,6 +77,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C + select SND_SOC_WM8991 if I2C select SND_SOC_WM8993 if I2C select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI @@ -308,6 +309,9 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate +config SND_SOC_WM8991 + tristate + config SND_SOC_WM8993 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 333910a..68e76af 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -62,6 +62,7 @@ snd-soc-wm8978-objs := wm8978.o snd-soc-wm8985-objs := wm8985.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8994-tables.o snd-soc-wm8995-objs := wm8995.o @@ -143,6 +144,7 @@ obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8985) += snd-soc-wm8985.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c new file mode 100644 index 0000000..28fdfd6 --- /dev/null +++ b/sound/soc/codecs/wm8991.c @@ -0,0 +1,1427 @@ +/* + * wm8991.c -- WM8991 ALSA Soc Audio driver + * + * Copyright 2007-2010 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/slab.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 <asm/div64.h> + +#include "wm8991.h" + +struct wm8991_priv { + enum snd_soc_control_type control_type; + unsigned int pcmclk; +}; + +static const u16 wm8991_reg_defs[] = { + 0x8991, /* R0 - Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking (1) */ + 0x0000, /* R7 - Clocking (2) */ + 0x0040, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0100, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x1000, /* R19 - GPIO1 & GPIO2 */ + 0x1010, /* R20 - GPIO3 & GPIO4 */ + 0x1010, /* R21 - GPIO5 & GPIO6 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x0000, /* R28 - Left Output Volume */ + 0x0000, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0022, /* R31 - Out3/4 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - Speaker Volume */ + 0x0003, /* R35 - ClassD1 */ + 0x0000, /* R36 */ + 0x0100, /* R37 - ClassD3 */ + 0x0000, /* R38 */ + 0x0000, /* R39 - Input Mixer1 */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0180, /* R51 - Out3/4 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0008, /* R60 - PLL1 */ + 0x0031, /* R61 - PLL2 */ + 0x0026, /* R62 - PLL3 */ +}; + +#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) + +static const unsigned int rec_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1500, 600), +}; + +static const unsigned int in_pga_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 0x1F, TLV_DB_LINEAR_ITEM(-1650, 3000), +}; + +static const unsigned int out_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(0, -2100), +}; + +static const unsigned int out_pga_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 127, TLV_DB_LINEAR_ITEM(-7300, 600), +}; + +static const unsigned int out_omix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-600, 0), +}; + +static const unsigned int out_dac_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 255, TLV_DB_LINEAR_ITEM(-7163, 0), +}; + +static const unsigned int in_adc_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 255, TLV_DB_LINEAR_ITEM(-7163, 1763), +}; + +static const unsigned int out_sidetone_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 31, TLV_DB_LINEAR_ITEM(-3600, 0), +}; + +static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); +} + +static const char *wm8991_digital_sidetone[] = +{"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8991_left_digital_sidetone_enum = + SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACL_SHIFT, + WM8991_ADC_TO_DACL_MASK, + wm8991_digital_sidetone); + +static const struct soc_enum wm8991_right_digital_sidetone_enum = + SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACR_SHIFT, + WM8991_ADC_TO_DACR_MASK, + wm8991_digital_sidetone); + +static const char *wm8991_adcmode[] = +{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8991_right_adcmode_enum = + SOC_ENUM_SINGLE(WM8991_ADC_CTRL, + WM8991_ADC_HPF_CUT_SHIFT, + WM8991_ADC_HPF_CUT_MASK, + wm8991_adcmode); + +static const struct snd_kcontrol_new wm8991_snd_controls[] = { + /* INMIXL */ + SOC_SINGLE("LIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L12MNBST_BIT, 1, 0), + SOC_SINGLE("LIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L34MNBST_BIT, 1, 0), + /* INMIXR */ + SOC_SINGLE("RIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R12MNBST_BIT, 1, 0), + SOC_SINGLE("RIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R34MNBST_BIT, 1, 0), + + /* LOMIX */ + SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LLI3LOVOL_SHIFT, WM8991_LLI3LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LR12LOVOL_SHIFT, WM8991_LR12LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LL12LOVOL_SHIFT, WM8991_LL12LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRI3LOVOL_SHIFT, WM8991_LRI3LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv), + + /* ROMIX */ + SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RRI3ROVOL_SHIFT, WM8991_RRI3ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RL12ROVOL_SHIFT, WM8991_RL12ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RR12ROVOL_SHIFT, WM8991_RR12ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RLI3ROVOL_SHIFT, WM8991_RLI3ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RLBROVOL_SHIFT, WM8991_RLBROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RRBROVOL_SHIFT, WM8991_RRBROVOL_MASK, 1, out_mix_tlv), + + /* LOUT */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8991_LEFT_OUTPUT_VOLUME, + WM8991_LOUTVOL_SHIFT, WM8991_LOUTVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("LOUT ZC", WM8991_LEFT_OUTPUT_VOLUME, WM8991_LOZC_BIT, 1, 0), + + /* ROUT */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8991_RIGHT_OUTPUT_VOLUME, + WM8991_ROUTVOL_SHIFT, WM8991_ROUTVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("ROUT ZC", WM8991_RIGHT_OUTPUT_VOLUME, WM8991_ROZC_BIT, 1, 0), + + /* LOPGA */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8991_LEFT_OPGA_VOLUME, + WM8991_LOPGAVOL_SHIFT, WM8991_LOPGAVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("LOPGA ZC Switch", WM8991_LEFT_OPGA_VOLUME, + WM8991_LOPGAZC_BIT, 1, 0), + + /* ROPGA */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8991_RIGHT_OPGA_VOLUME, + WM8991_ROPGAVOL_SHIFT, WM8991_ROPGAVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("ROPGA ZC Switch", WM8991_RIGHT_OPGA_VOLUME, + WM8991_ROPGAZC_BIT, 1, 0), + + SOC_SINGLE("LON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LONMUTE_BIT, 1, 0), + SOC_SINGLE("LOP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LOPMUTE_BIT, 1, 0), + SOC_SINGLE("LOP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LOATTN_BIT, 1, 0), + SOC_SINGLE("RON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_RONMUTE_BIT, 1, 0), + SOC_SINGLE("ROP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_ROPMUTE_BIT, 1, 0), + SOC_SINGLE("ROP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_ROATTN_BIT, 1, 0), + + SOC_SINGLE("OUT3 Mute Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT3MUTE_BIT, 1, 0), + SOC_SINGLE("OUT3 Attenuation Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT3ATTN_BIT, 1, 0), + + SOC_SINGLE("OUT4 Mute Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT4MUTE_BIT, 1, 0), + SOC_SINGLE("OUT4 Attenuation Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT4ATTN_BIT, 1, 0), + + SOC_SINGLE("Speaker Mode Switch", WM8991_CLASSD1, + WM8991_CDMODE_BIT, 1, 0), + + SOC_SINGLE("Speaker Output Attenuation Volume", WM8991_SPEAKER_VOLUME, + WM8991_SPKVOL_SHIFT, WM8991_SPKVOL_MASK, 0), + SOC_SINGLE("Speaker DC Boost Volume", WM8991_CLASSD3, + WM8991_DCGAIN_SHIFT, WM8991_DCGAIN_MASK, 0), + SOC_SINGLE("Speaker AC Boost Volume", WM8991_CLASSD3, + WM8991_ACGAIN_SHIFT, WM8991_ACGAIN_MASK, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8991_LEFT_DAC_DIGITAL_VOLUME, + WM8991_DACL_VOL_SHIFT, + WM8991_DACL_VOL_MASK, + 0, + out_dac_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8991_RIGHT_DAC_DIGITAL_VOLUME, + WM8991_DACR_VOL_SHIFT, + WM8991_DACR_VOL_MASK, + 0, + out_dac_tlv), + + SOC_ENUM("Left Digital Sidetone", wm8991_left_digital_sidetone_enum), + SOC_ENUM("Right Digital Sidetone", wm8991_right_digital_sidetone_enum), + + SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE, + WM8991_ADCL_DAC_SVOL_SHIFT, WM8991_ADCL_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE, + WM8991_ADCR_DAC_SVOL_SHIFT, WM8991_ADCR_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + + SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8991_ADC_CTRL, + WM8991_ADC_HPF_ENA_BIT, 1, 0), + + SOC_ENUM("ADC HPF Mode", wm8991_right_adcmode_enum), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8991_LEFT_ADC_DIGITAL_VOLUME, + WM8991_ADCL_VOL_SHIFT, + WM8991_ADCL_VOL_MASK, + 0, + in_adc_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8991_RIGHT_ADC_DIGITAL_VOLUME, + WM8991_ADCR_VOL_SHIFT, + WM8991_ADCR_VOL_MASK, + 0, + in_adc_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LIN12VOL_SHIFT, + WM8991_LIN12VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("LIN12 ZC Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LI12ZC_BIT, 1, 0), + + SOC_SINGLE("LIN12 Mute Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LI12MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LIN34VOL_SHIFT, + WM8991_LIN34VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("LIN34 ZC Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LI34ZC_BIT, 1, 0), + + SOC_SINGLE("LIN34 Mute Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LI34MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RIN12VOL_SHIFT, + WM8991_RIN12VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("RIN12 ZC Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RI12ZC_BIT, 1, 0), + + SOC_SINGLE("RIN12 Mute Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RI12MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RIN34VOL_SHIFT, + WM8991_RIN34VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("RIN34 ZC Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RI34ZC_BIT, 1, 0), + + SOC_SINGLE("RIN34 Mute Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RI34MUTE_BIT, 1, 0), +}; + +/* + * _DAPM_ Controls + */ +static int inmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS); + + if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) | + (1 << WM8991_AINLMUX_PWR_BIT))) + reg |= WM8991_AINL_ENA; + else + reg &= ~WM8991_AINL_ENA; + + if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) | + (1 << WM8991_AINRMUX_PWR_BIT))) + reg |= WM8991_AINR_ENA; + else + reg &= ~WM8991_AINL_ENA; + + snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); + return 0; +} + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u32 reg_shift = kcontrol->private_value & 0xfff; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER1); + if (reg & WM8991_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + + case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER2); + if (reg & WM8991_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + + case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + if (reg & WM8991_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + + case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + if (reg & WM8991_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8991_dapm_lin12_pga_controls[] = { + SOC_DAPM_SINGLE("LIN1 Switch", WM8991_INPUT_MIXER2, WM8991_LMN1_BIT, 1, 0), + SOC_DAPM_SINGLE("LIN2 Switch", WM8991_INPUT_MIXER2, WM8991_LMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8991_dapm_lin34_pga_controls[] = { + SOC_DAPM_SINGLE("LIN3 Switch", WM8991_INPUT_MIXER2, WM8991_LMN3_BIT, 1, 0), + SOC_DAPM_SINGLE("LIN4 Switch", WM8991_INPUT_MIXER2, WM8991_LMP4_BIT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8991_dapm_rin12_pga_controls[] = { + SOC_DAPM_SINGLE("RIN1 Switch", WM8991_INPUT_MIXER2, WM8991_RMN1_BIT, 1, 0), + SOC_DAPM_SINGLE("RIN2 Switch", WM8991_INPUT_MIXER2, WM8991_RMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8991_dapm_rin34_pga_controls[] = { + SOC_DAPM_SINGLE("RIN3 Switch", WM8991_INPUT_MIXER2, WM8991_RMN3_BIT, 1, 0), + SOC_DAPM_SINGLE("RIN4 Switch", WM8991_INPUT_MIXER2, WM8991_RMP4_BIT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8991_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8991_INPUT_MIXER3, + WM8991_LDBVOL_SHIFT, WM8991_LDBVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8991_INPUT_MIXER5, WM8991_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), + SOC_DAPM_SINGLE("LINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT, + 1, 0), + SOC_DAPM_SINGLE("LINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8991_INPUT_MIXER4, + WM8991_RDBVOL_SHIFT, WM8991_RDBVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8991_INPUT_MIXER6, WM8991_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), + SOC_DAPM_SINGLE("RINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT, + 1, 0), + SOC_DAPM_SINGLE("RINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8991_ainlmux[] = +{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8991_ainlmux_enum = + SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, + ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux); + +static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = + SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8991_ainrmux[] = +{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8991_ainrmux_enum = + SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, + ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux); + +static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls = + SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8991_dapm_rxvoice_controls[] = { + SOC_DAPM_SINGLE_TLV("LIN4RXN", WM8991_INPUT_MIXER5, WM8991_LR4BVOL_SHIFT, + WM8991_LR4BVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("RIN4RXP", WM8991_INPUT_MIXER6, WM8991_RL4BVOL_SHIFT, + WM8991_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lomix_controls[] = { + SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LRBLO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LLBLO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LRI3LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LLI3LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LR12LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LL12LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8991_OUTPUT_MIXER1, + WM8991_LDLO_BIT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8991_dapm_romix_controls[] = { + SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RLBRO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RRBRO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RLI3RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RRI3RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RL12RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RR12RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8991_OUTPUT_MIXER2, + WM8991_RDRO_BIT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lonmix_controls[] = { + SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LLOPGALON_BIT, 1, 0), + SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LROPGALON_BIT, 1, 0), + SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8991_LINE_MIXER1, + WM8991_LOPLON_BIT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lopmix_controls[] = { + SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER1, + WM8991_LR12LOP_BIT, 1, 0), + SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER1, + WM8991_LL12LOP_BIT, 1, 0), + SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LLOPGALOP_BIT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8991_dapm_ronmix_controls[] = { + SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RROPGARON_BIT, 1, 0), + SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RLOPGARON_BIT, 1, 0), + SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8991_LINE_MIXER2, + WM8991_ROPRON_BIT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8991_dapm_ropmix_controls[] = { + SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER2, + WM8991_RL12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER2, + WM8991_RR12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RROPGAROP_BIT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8991_dapm_out3mix_controls[] = { + SOC_DAPM_SINGLE("OUT3MIX LIN4RXN Bypass Switch", WM8991_OUT3_4_MIXER, + WM8991_LI4O3_BIT, 1, 0), + SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8991_OUT3_4_MIXER, + WM8991_LPGAO3_BIT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8991_dapm_out4mix_controls[] = { + SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8991_OUT3_4_MIXER, + WM8991_RPGAO4_BIT, 1, 0), + SOC_DAPM_SINGLE("OUT4MIX RIN4RXP Bypass Switch", WM8991_OUT3_4_MIXER, + WM8991_RI4O4_BIT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8991_dapm_spkmix_controls[] = { + SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_LI2SPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_LB2SPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8991_SPEAKER_MIXER, + WM8991_LOPGASPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8991_SPEAKER_MIXER, + WM8991_LDSPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8991_SPEAKER_MIXER, + WM8991_RDSPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8991_SPEAKER_MIXER, + WM8991_ROPGASPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_RL12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_RI2SPK_BIT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("LIN4RXN"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("RIN4RXP"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("Internal ADC Source"), + + /* DACs */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2, + WM8991_ADCL_ENA_BIT, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8991_POWER_MANAGEMENT_2, + WM8991_ADCR_ENA_BIT, 0), + + /* Input PGAs */ + SND_SOC_DAPM_MIXER("LIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN12_ENA_BIT, + 0, &wm8991_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_lin12_pga_controls)), + SND_SOC_DAPM_MIXER("LIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN34_ENA_BIT, + 0, &wm8991_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_lin34_pga_controls)), + SND_SOC_DAPM_MIXER("RIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN12_ENA_BIT, + 0, &wm8991_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_rin12_pga_controls)), + SND_SOC_DAPM_MIXER("RIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN34_ENA_BIT, + 0, &wm8991_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)), + + /* INMIXL */ + SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0, + &wm8991_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8991_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* AINLMUX */ + SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0, + &wm8991_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* INMIXR */ + SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0, + &wm8991_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8991_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* AINRMUX */ + SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0, + &wm8991_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8991_POWER_MANAGEMENT_3, + WM8991_DACL_ENA_BIT, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8991_POWER_MANAGEMENT_3, + WM8991_DACR_ENA_BIT, 0), + + /* LOMIX */ + SND_SOC_DAPM_MIXER_E("LOMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOMIX_ENA_BIT, + 0, &wm8991_dapm_lomix_controls[0], + ARRAY_SIZE(wm8991_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + + /* LONMIX */ + SND_SOC_DAPM_MIXER("LONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LON_ENA_BIT, 0, + &wm8991_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8991_dapm_lonmix_controls)), + + /* LOPMIX */ + SND_SOC_DAPM_MIXER("LOPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOP_ENA_BIT, 0, + &wm8991_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8991_dapm_lopmix_controls)), + + /* OUT3MIX */ + SND_SOC_DAPM_MIXER("OUT3MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT3_ENA_BIT, 0, + &wm8991_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8991_dapm_out3mix_controls)), + + /* SPKMIX */ + SND_SOC_DAPM_MIXER_E("SPKMIX", WM8991_POWER_MANAGEMENT_1, WM8991_SPK_ENA_BIT, 0, + &wm8991_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8991_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + + /* OUT4MIX */ + SND_SOC_DAPM_MIXER("OUT4MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT4_ENA_BIT, 0, + &wm8991_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8991_dapm_out4mix_controls)), + + /* ROPMIX */ + SND_SOC_DAPM_MIXER("ROPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROP_ENA_BIT, 0, + &wm8991_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8991_dapm_ropmix_controls)), + + /* RONMIX */ + SND_SOC_DAPM_MIXER("RONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_RON_ENA_BIT, 0, + &wm8991_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8991_dapm_ronmix_controls)), + + /* ROMIX */ + SND_SOC_DAPM_MIXER_E("ROMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROMIX_ENA_BIT, + 0, &wm8991_dapm_romix_controls[0], + ARRAY_SIZE(wm8991_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + + /* LOUT PGA */ + SND_SOC_DAPM_PGA("LOUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_LOUT_ENA_BIT, 0, + NULL, 0), + + /* ROUT PGA */ + SND_SOC_DAPM_PGA("ROUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_ROUT_ENA_BIT, 0, + NULL, 0), + + /* LOPGA */ + SND_SOC_DAPM_PGA("LOPGA", WM8991_POWER_MANAGEMENT_3, WM8991_LOPGA_ENA_BIT, 0, + NULL, 0), + + /* ROPGA */ + SND_SOC_DAPM_PGA("ROPGA", WM8991_POWER_MANAGEMENT_3, WM8991_ROPGA_ENA_BIT, 0, + NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_MICBIAS("MICBIAS", WM8991_POWER_MANAGEMENT_1, + WM8991_MICBIAS_ENA_BIT, 0), + + SND_SOC_DAPM_OUTPUT("LON"), + SND_SOC_DAPM_OUTPUT("LOP"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("OUT4"), + SND_SOC_DAPM_OUTPUT("ROP"), + SND_SOC_DAPM_OUTPUT("RON"), + SND_SOC_DAPM_OUTPUT("OUT"), + + SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4RXN"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4RXP"}, + /* ADC */ + {"Left ADC", NULL, "AINLMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4RXP"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4RXP"}, + /* ADC */ + {"Right ADC", NULL, "AINRMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4RXN Bypass Switch", "LIN4RXN"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4RXP Bypass Switch", "RIN4RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 div2; + u32 n; + u32 k; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 16) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->div2 = 1; + Ndiv = target / source; + } else + pll_div->div2 = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8991 N value outwith recommended range! N = %d\n", Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, int src, unsigned int freq_in, unsigned int freq_out) +{ + u16 reg; + struct snd_soc_codec *codec = codec_dai->codec; + struct _pll_div pll_div; + + if (freq_in && freq_out) { + pll_factors(&pll_div, freq_out * 4, freq_in); + + /* Turn on PLL */ + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + reg |= WM8991_PLL_ENA; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg); + + /* sysclk comes from PLL */ + reg = snd_soc_read(codec, WM8991_CLOCKING_2); + snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC); + + /* set up N , fractional mode and pre-divisor if neccessary */ + snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM | + (pll_div.div2 ? WM8991_PRESCALE : 0)); + snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8)); + snd_soc_write(codec, WM8991_PLL3, (u8)(pll_div.k & 0xFF)); + } else { + /* Turn on PLL */ + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + reg &= ~WM8991_PLL_ENA; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg); + } + return 0; +} + +/* + * Set's ADC and Voice DAC format. + */ +static int wm8991_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8991_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8991_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8991_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8991_AIF_TMF_I2S; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8991_AIF_TMF_RIGHTJ; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8991_AIF_TMF_LEFTJ; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8991_AIF_TMF_DSP; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8991_AIF_TMF_DSP | WM8991_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8991_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8991_MCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_MCLK_DIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_DACCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_DAC_CLKDIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_ADCCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_ADC_CLKDIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_BCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_1) & + ~WM8991_BCLK_DIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8991_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u16 audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1); + + audio1 &= ~WM8991_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8991_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8991_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8991_AIF_WL_32BITS; + break; + } + + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8991_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val; + + val = snd_soc_read(codec, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE; + if (mute) + snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE); + else + snd_soc_write(codec, WM8991_DAC_CTRL, val); + return 0; +} + +static int wm8991_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) & + ~WM8991_VMID_MODE_MASK; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Enable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | + WM8991_DIS_RLINE | WM8991_DIS_OUT3 | + WM8991_DIS_OUT4 | WM8991_DIS_LOUT | + WM8991_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_VMIDTOG); + + /* Delay to allow output caps to discharge */ + msleep(300); + + /* Disable VMIDTOG */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL); + + /* disable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, 0); + + /* Enable outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1b00); + + msleep(50); + + /* Enable VMID at 2x50k */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f02); + + msleep(100); + + /* Enable VREF */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03); + + msleep(600); + + /* Enable BUFIOEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_BUFIOEN); + + /* Disable outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x3); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_BUFIOEN); + } + + /* VMID=2*250k */ + val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) & + ~WM8991_VMID_MODE_MASK; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x4); + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_POBCTRL | WM8991_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_BUFIOEN); + + /* mute DAC */ + val = snd_soc_read(codec, WM8991_DAC_CTRL); + snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE); + + /* Enable any disabled outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03); + + /* Disable VMID */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f01); + + msleep(300); + + /* Enable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | + WM8991_DIS_RLINE | WM8991_DIS_OUT3 | + WM8991_DIS_OUT4 | WM8991_DIS_LOUT | + WM8991_DIS_ROUT); + + /* Disable VREF */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x0); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, 0x0); + codec->cache_sync = 1; + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8991_resume(struct snd_soc_codec *codec) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +/* power down chip */ +static int wm8991_remove(struct snd_soc_codec *codec) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8991_probe(struct snd_soc_codec *codec) +{ + struct wm8991_priv *wm8991; + int ret; + unsigned int reg; + + wm8991 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); + return ret; + } + + ret = wm8991_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4); + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1); + + reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) & + ~WM8991_GPIO1_SEL_MASK; + snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1); + + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA| + WM8991_VMID_MODE_MASK); + + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA); + + snd_soc_write(codec, WM8991_DAC_CTRL, 0); + snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + snd_soc_add_controls(codec, wm8991_snd_controls, + ARRAY_SIZE(wm8991_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, + ARRAY_SIZE(wm8991_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + return 0; +} + +#define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8991_ops = { + .hw_params = wm8991_hw_params, + .digital_mute = wm8991_mute, + .set_fmt = wm8991_set_dai_fmt, + .set_clkdiv = wm8991_set_dai_clkdiv, + .set_pll = wm8991_set_dai_pll +}; + +/* + * The WM8991 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +static struct snd_soc_dai_driver wm8991_dai = { + /* ADC/DAC on primary */ + .name = "wm8991", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8991_FORMATS + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8991_FORMATS + }, + .ops = &wm8991_ops +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { + .probe = wm8991_probe, + .remove = wm8991_remove, + .suspend = wm8991_suspend, + .resume = wm8991_resume, + .set_bias_level = wm8991_set_bias_level, + .reg_cache_size = WM8991_MAX_REGISTER + 1, + .reg_word_size = sizeof(u16), + .reg_cache_default = wm8991_reg_defs +}; + +static __devinit int wm8991_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8991_priv *wm8991; + int ret; + + wm8991 = kzalloc(sizeof *wm8991, GFP_KERNEL); + if (!wm8991) + return -ENOMEM; + + wm8991->control_type = SND_SOC_I2C; + i2c_set_clientdata(i2c, wm8991); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8991, &wm8991_dai, 1); + if (ret < 0) + kfree(wm8991); + return ret; +} + +static __devexit int wm8991_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id wm8991_i2c_id[] = { + { "wm8991", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); + +static struct i2c_driver wm8991_i2c_driver = { + .driver = { + .name = "wm8991", + .owner = THIS_MODULE, + }, + .probe = wm8991_i2c_probe, + .remove = __devexit_p(wm8991_i2c_remove), + .id_table = wm8991_i2c_id, +}; + +static int __init wm8991_modinit(void) +{ + int ret; + ret = i2c_add_driver(&wm8991_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n", + ret); + } + return 0; +} +module_init(wm8991_modinit); + +static void __exit wm8991_exit(void) +{ + i2c_del_driver(&wm8991_i2c_driver); +} +module_exit(wm8991_exit); + +MODULE_DESCRIPTION("ASoC WM8991 driver"); +MODULE_AUTHOR("Graeme Gregory"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h new file mode 100644 index 0000000..8a942ef --- /dev/null +++ b/sound/soc/codecs/wm8991.h @@ -0,0 +1,833 @@ +/* + * wm8991.h -- audio driver for WM8991 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _WM8991_H +#define _WM8991_H + +/* + * Register values. + */ +#define WM8991_RESET 0x00 +#define WM8991_POWER_MANAGEMENT_1 0x01 +#define WM8991_POWER_MANAGEMENT_2 0x02 +#define WM8991_POWER_MANAGEMENT_3 0x03 +#define WM8991_AUDIO_INTERFACE_1 0x04 +#define WM8991_AUDIO_INTERFACE_2 0x05 +#define WM8991_CLOCKING_1 0x06 +#define WM8991_CLOCKING_2 0x07 +#define WM8991_AUDIO_INTERFACE_3 0x08 +#define WM8991_AUDIO_INTERFACE_4 0x09 +#define WM8991_DAC_CTRL 0x0A +#define WM8991_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8991_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8991_DIGITAL_SIDE_TONE 0x0D +#define WM8991_ADC_CTRL 0x0E +#define WM8991_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8991_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8991_GPIO_CTRL_1 0x12 +#define WM8991_GPIO1_GPIO2 0x13 +#define WM8991_GPIO3_GPIO4 0x14 +#define WM8991_GPIO5_GPIO6 0x15 +#define WM8991_GPIOCTRL_2 0x16 +#define WM8991_GPIO_POL 0x17 +#define WM8991_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8991_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8991_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8991_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8991_LEFT_OUTPUT_VOLUME 0x1C +#define WM8991_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8991_LINE_OUTPUTS_VOLUME 0x1E +#define WM8991_OUT3_4_VOLUME 0x1F +#define WM8991_LEFT_OPGA_VOLUME 0x20 +#define WM8991_RIGHT_OPGA_VOLUME 0x21 +#define WM8991_SPEAKER_VOLUME 0x22 +#define WM8991_CLASSD1 0x23 +#define WM8991_CLASSD3 0x25 +#define WM8991_INPUT_MIXER1 0x27 +#define WM8991_INPUT_MIXER2 0x28 +#define WM8991_INPUT_MIXER3 0x29 +#define WM8991_INPUT_MIXER4 0x2A +#define WM8991_INPUT_MIXER5 0x2B +#define WM8991_INPUT_MIXER6 0x2C +#define WM8991_OUTPUT_MIXER1 0x2D +#define WM8991_OUTPUT_MIXER2 0x2E +#define WM8991_OUTPUT_MIXER3 0x2F +#define WM8991_OUTPUT_MIXER4 0x30 +#define WM8991_OUTPUT_MIXER5 0x31 +#define WM8991_OUTPUT_MIXER6 0x32 +#define WM8991_OUT3_4_MIXER 0x33 +#define WM8991_LINE_MIXER1 0x34 +#define WM8991_LINE_MIXER2 0x35 +#define WM8991_SPEAKER_MIXER 0x36 +#define WM8991_ADDITIONAL_CONTROL 0x37 +#define WM8991_ANTIPOP1 0x38 +#define WM8991_ANTIPOP2 0x39 +#define WM8991_MICBIAS 0x3A +#define WM8991_PLL1 0x3C +#define WM8991_PLL2 0x3D +#define WM8991_PLL3 0x3E +#define WM8991_INTDRIVBITS 0x3F + +#define WM8991_REGISTER_COUNT 60 +#define WM8991_MAX_REGISTER 0x3F + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Reset + */ +#define WM8991_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID - [15:0] */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8991_SPK_ENA 0x1000 /* SPK_ENA */ +#define WM8991_SPK_ENA_BIT 12 +#define WM8991_OUT3_ENA 0x0800 /* OUT3_ENA */ +#define WM8991_OUT3_ENA_BIT 11 +#define WM8991_OUT4_ENA 0x0400 /* OUT4_ENA */ +#define WM8991_OUT4_ENA_BIT 10 +#define WM8991_LOUT_ENA 0x0200 /* LOUT_ENA */ +#define WM8991_LOUT_ENA_BIT 9 +#define WM8991_ROUT_ENA 0x0100 /* ROUT_ENA */ +#define WM8991_ROUT_ENA_BIT 8 +#define WM8991_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */ +#define WM8991_MICBIAS_ENA_BIT 4 +#define WM8991_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */ +#define WM8991_VREF_ENA 0x0001 /* VREF_ENA */ +#define WM8991_VREF_ENA_BIT 0 + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8991_PLL_ENA 0x8000 /* PLL_ENA */ +#define WM8991_PLL_ENA_BIT 15 +#define WM8991_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8991_TSHUT_ENA_BIT 14 +#define WM8991_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8991_TSHUT_OPDIS_BIT 13 +#define WM8991_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8991_OPCLK_ENA_BIT 11 +#define WM8991_AINL_ENA 0x0200 /* AINL_ENA */ +#define WM8991_AINL_ENA_BIT 9 +#define WM8991_AINR_ENA 0x0100 /* AINR_ENA */ +#define WM8991_AINR_ENA_BIT 8 +#define WM8991_LIN34_ENA 0x0080 /* LIN34_ENA */ +#define WM8991_LIN34_ENA_BIT 7 +#define WM8991_LIN12_ENA 0x0040 /* LIN12_ENA */ +#define WM8991_LIN12_ENA_BIT 6 +#define WM8991_RIN34_ENA 0x0020 /* RIN34_ENA */ +#define WM8991_RIN34_ENA_BIT 5 +#define WM8991_RIN12_ENA 0x0010 /* RIN12_ENA */ +#define WM8991_RIN12_ENA_BIT 4 +#define WM8991_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8991_ADCL_ENA_BIT 1 +#define WM8991_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8991_ADCR_ENA_BIT 0 + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8991_LON_ENA 0x2000 /* LON_ENA */ +#define WM8991_LON_ENA_BIT 13 +#define WM8991_LOP_ENA 0x1000 /* LOP_ENA */ +#define WM8991_LOP_ENA_BIT 12 +#define WM8991_RON_ENA 0x0800 /* RON_ENA */ +#define WM8991_RON_ENA_BIT 11 +#define WM8991_ROP_ENA 0x0400 /* ROP_ENA */ +#define WM8991_ROP_ENA_BIT 10 +#define WM8991_LOPGA_ENA 0x0080 /* LOPGA_ENA */ +#define WM8991_LOPGA_ENA_BIT 7 +#define WM8991_ROPGA_ENA 0x0040 /* ROPGA_ENA */ +#define WM8991_ROPGA_ENA_BIT 6 +#define WM8991_LOMIX_ENA 0x0020 /* LOMIX_ENA */ +#define WM8991_LOMIX_ENA_BIT 5 +#define WM8991_ROMIX_ENA 0x0010 /* ROMIX_ENA */ +#define WM8991_ROMIX_ENA_BIT 4 +#define WM8991_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8991_DACL_ENA_BIT 1 +#define WM8991_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8991_DACR_ENA_BIT 0 + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8991_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8991_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8991_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8991_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8991_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8991_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8991_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8991_AIF_WL_16BITS (0 << 5) +#define WM8991_AIF_WL_20BITS (1 << 5) +#define WM8991_AIF_WL_24BITS (2 << 5) +#define WM8991_AIF_WL_32BITS (3 << 5) +#define WM8991_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8991_AIF_TMF_RIGHTJ (0 << 3) +#define WM8991_AIF_TMF_LEFTJ (1 << 3) +#define WM8991_AIF_TMF_I2S (2 << 3) +#define WM8991_AIF_TMF_DSP (3 << 3) + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8991_DACL_SRC 0x8000 /* DACL_SRC */ +#define WM8991_DACR_SRC 0x4000 /* DACR_SRC */ +#define WM8991_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8991_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8991_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST - [11:10] */ +#define WM8991_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8991_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8991_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8991_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8991_LOOPBACK 0x0001 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking (1) + */ +#define WM8991_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8991_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8991_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */ +#define WM8991_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8991_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8991_BCLK_DIV_1 (0x0 << 1) +#define WM8991_BCLK_DIV_1_5 (0x1 << 1) +#define WM8991_BCLK_DIV_2 (0x2 << 1) +#define WM8991_BCLK_DIV_3 (0x3 << 1) +#define WM8991_BCLK_DIV_4 (0x4 << 1) +#define WM8991_BCLK_DIV_5_5 (0x5 << 1) +#define WM8991_BCLK_DIV_6 (0x6 << 1) +#define WM8991_BCLK_DIV_8 (0x7 << 1) +#define WM8991_BCLK_DIV_11 (0x8 << 1) +#define WM8991_BCLK_DIV_12 (0x9 << 1) +#define WM8991_BCLK_DIV_16 (0xA << 1) +#define WM8991_BCLK_DIV_22 (0xB << 1) +#define WM8991_BCLK_DIV_24 (0xC << 1) +#define WM8991_BCLK_DIV_32 (0xD << 1) +#define WM8991_BCLK_DIV_44 (0xE << 1) +#define WM8991_BCLK_DIV_48 (0xF << 1) + +/* + * R7 (0x07) - Clocking (2) + */ +#define WM8991_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8991_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8991_CLK_FORCE 0x2000 /* CLK_FORCE */ +#define WM8991_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */ +#define WM8991_MCLK_DIV_1 (0 << 11) +#define WM8991_MCLK_DIV_2 ( 2 << 11) +#define WM8991_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8991_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV - [7:5] */ +#define WM8991_ADC_CLKDIV_1 (0 << 5) +#define WM8991_ADC_CLKDIV_1_5 (1 << 5) +#define WM8991_ADC_CLKDIV_2 (2 << 5) +#define WM8991_ADC_CLKDIV_3 (3 << 5) +#define WM8991_ADC_CLKDIV_4 (4 << 5) +#define WM8991_ADC_CLKDIV_5_5 (5 << 5) +#define WM8991_ADC_CLKDIV_6 (6 << 5) +#define WM8991_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */ +#define WM8991_DAC_CLKDIV_1 (0 << 2) +#define WM8991_DAC_CLKDIV_1_5 (1 << 2) +#define WM8991_DAC_CLKDIV_2 (2 << 2) +#define WM8991_DAC_CLKDIV_3 (3 << 2) +#define WM8991_DAC_CLKDIV_4 (4 << 2) +#define WM8991_DAC_CLKDIV_5_5 (5 << 2) +#define WM8991_DAC_CLKDIV_6 (6 << 2) + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8991_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8991_AIF_MSTR2 0x4000 /* AIF_MSTR2 */ +#define WM8991_AIF_SEL 0x2000 /* AIF_SEL */ +#define WM8991_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */ +#define WM8991_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE - [10:0] */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8991_ALRCGPIO1 0x8000 /* ALRCGPIO1 */ +#define WM8991_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */ +#define WM8991_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8991_DACLRC_DIR 0x0800 /* DACLRC_DIR */ +#define WM8991_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE - [10:0] */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8991_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */ +#define WM8991_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8991_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8991_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8991_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */ +#define WM8991_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */ +#define WM8991_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8991_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8991_DACR_DATINV 0x0001 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8991_DAC_VU 0x0100 /* DAC_VU */ +#define WM8991_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8991_DACL_VOL_SHIFT 0 +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8991_DAC_VU 0x0100 /* DAC_VU */ +#define WM8991_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8991_DACR_VOL_SHIFT 0 +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8991_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL - [12:9] */ +#define WM8991_ADCL_DAC_SVOL_SHIFT 9 +#define WM8991_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL - [8:5] */ +#define WM8991_ADCR_DAC_SVOL_SHIFT 5 +#define WM8991_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */ +#define WM8991_ADC_TO_DACL_SHIFT 2 +#define WM8991_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */ +#define WM8991_ADC_TO_DACR_SHIFT 0 + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8991_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */ +#define WM8991_ADC_HPF_ENA_BIT 8 +#define WM8991_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */ +#define WM8991_ADC_HPF_CUT_SHIFT 5 +#define WM8991_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8991_ADCL_DATINV_BIT 1 +#define WM8991_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8991_ADCR_DATINV_BIT 0 + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8991_ADC_VU 0x0100 /* ADC_VU */ +#define WM8991_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8991_ADCL_VOL_SHIFT 0 + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8991_ADC_VU 0x0100 /* ADC_VU */ +#define WM8991_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8991_ADCR_VOL_SHIFT 0 + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8991_IRQ 0x1000 /* IRQ */ +#define WM8991_TEMPOK 0x0800 /* TEMPOK */ +#define WM8991_MICSHRT 0x0400 /* MICSHRT */ +#define WM8991_MICDET 0x0200 /* MICDET */ +#define WM8991_PLL_LCK 0x0100 /* PLL_LCK */ +#define WM8991_GPI8_STATUS 0x0080 /* GPI8_STATUS */ +#define WM8991_GPI7_STATUS 0x0040 /* GPI7_STATUS */ +#define WM8991_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */ +#define WM8991_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */ +#define WM8991_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */ +#define WM8991_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */ +#define WM8991_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */ +#define WM8991_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */ + +/* + * R19 (0x13) - GPIO1 & GPIO2 + */ +#define WM8991_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */ +#define WM8991_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */ +#define WM8991_GPIO2_PU 0x2000 /* GPIO2_PU */ +#define WM8991_GPIO2_PD 0x1000 /* GPIO2_PD */ +#define WM8991_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */ +#define WM8991_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */ +#define WM8991_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */ +#define WM8991_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8991_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8991_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - GPIO3 & GPIO4 + */ +#define WM8991_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */ +#define WM8991_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */ +#define WM8991_GPIO4_PU 0x2000 /* GPIO4_PU */ +#define WM8991_GPIO4_PD 0x1000 /* GPIO4_PD */ +#define WM8991_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */ +#define WM8991_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */ +#define WM8991_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */ +#define WM8991_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8991_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8991_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ + +/* + * R21 (0x15) - GPIO5 & GPIO6 + */ +#define WM8991_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */ +#define WM8991_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */ +#define WM8991_GPIO6_PU 0x2000 /* GPIO6_PU */ +#define WM8991_GPIO6_PD 0x1000 /* GPIO6_PD */ +#define WM8991_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */ +#define WM8991_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */ +#define WM8991_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */ +#define WM8991_GPIO5_PU 0x0020 /* GPIO5_PU */ +#define WM8991_GPIO5_PD 0x0010 /* GPIO5_PD */ +#define WM8991_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8991_RD_3W_ENA 0x8000 /* RD_3W_ENA */ +#define WM8991_MODE_3W4W 0x4000 /* MODE_3W4W */ +#define WM8991_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */ +#define WM8991_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */ +#define WM8991_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */ +#define WM8991_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */ +#define WM8991_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */ +#define WM8991_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */ +#define WM8991_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8991_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */ +#define WM8991_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */ +#define WM8991_GPI7_ENA 0x0001 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8991_IRQ_INV 0x1000 /* IRQ_INV */ +#define WM8991_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8991_MICSHRT_POL 0x0400 /* MICSHRT_POL */ +#define WM8991_MICDET_POL 0x0200 /* MICDET_POL */ +#define WM8991_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */ +#define WM8991_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8991_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8991_GPIO6_POL 0x0020 /* GPIO6_POL */ +#define WM8991_GPIO5_POL 0x0010 /* GPIO5_POL */ +#define WM8991_GPIO4_POL 0x0008 /* GPIO4_POL */ +#define WM8991_GPIO3_POL 0x0004 /* GPIO3_POL */ +#define WM8991_GPIO2_POL 0x0002 /* GPIO2_POL */ +#define WM8991_GPIO1_POL 0x0001 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_LI12MUTE 0x0080 /* LI12MUTE */ +#define WM8991_LI12MUTE_BIT 7 +#define WM8991_LI12ZC 0x0040 /* LI12ZC */ +#define WM8991_LI12ZC_BIT 6 +#define WM8991_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */ +#define WM8991_LIN12VOL_SHIFT 0 +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_LI34MUTE 0x0080 /* LI34MUTE */ +#define WM8991_LI34MUTE_BIT 7 +#define WM8991_LI34ZC 0x0040 /* LI34ZC */ +#define WM8991_LI34ZC_BIT 6 +#define WM8991_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */ +#define WM8991_LIN34VOL_SHIFT 0 + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_RI12MUTE 0x0080 /* RI12MUTE */ +#define WM8991_RI12MUTE_BIT 7 +#define WM8991_RI12ZC 0x0040 /* RI12ZC */ +#define WM8991_RI12ZC_BIT 6 +#define WM8991_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */ +#define WM8991_RIN12VOL_SHIFT 0 + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_RI34MUTE 0x0080 /* RI34MUTE */ +#define WM8991_RI34MUTE_BIT 7 +#define WM8991_RI34ZC 0x0040 /* RI34ZC */ +#define WM8991_RI34ZC_BIT 6 +#define WM8991_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */ +#define WM8991_RIN34VOL_SHIFT 0 + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_LOZC 0x0080 /* LOZC */ +#define WM8991_LOZC_BIT 7 +#define WM8991_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8991_LOUTVOL_SHIFT 0 +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_ROZC 0x0080 /* ROZC */ +#define WM8991_ROZC_BIT 7 +#define WM8991_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8991_ROUTVOL_SHIFT 0 +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8991_LONMUTE 0x0040 /* LONMUTE */ +#define WM8991_LONMUTE_BIT 6 +#define WM8991_LOPMUTE 0x0020 /* LOPMUTE */ +#define WM8991_LOPMUTE_BIT 5 +#define WM8991_LOATTN 0x0010 /* LOATTN */ +#define WM8991_LOATTN_BIT 4 +#define WM8991_RONMUTE 0x0004 /* RONMUTE */ +#define WM8991_RONMUTE_BIT 2 +#define WM8991_ROPMUTE 0x0002 /* ROPMUTE */ +#define WM8991_ROPMUTE_BIT 1 +#define WM8991_ROATTN 0x0001 /* ROATTN */ +#define WM8991_ROATTN_BIT 0 + +/* + * R31 (0x1F) - Out3/4 Volume + */ +#define WM8991_OUT3MUTE 0x0020 /* OUT3MUTE */ +#define WM8991_OUT3MUTE_BIT 5 +#define WM8991_OUT3ATTN 0x0010 /* OUT3ATTN */ +#define WM8991_OUT3ATTN_BIT 4 +#define WM8991_OUT4MUTE 0x0002 /* OUT4MUTE */ +#define WM8991_OUT4MUTE_BIT 1 +#define WM8991_OUT4ATTN 0x0001 /* OUT4ATTN */ +#define WM8991_OUT4ATTN_BIT 0 + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_LOPGAZC 0x0080 /* LOPGAZC */ +#define WM8991_LOPGAZC_BIT 7 +#define WM8991_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */ +#define WM8991_LOPGAVOL_SHIFT 0 + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_ROPGAZC 0x0080 /* ROPGAZC */ +#define WM8991_ROPGAZC_BIT 7 +#define WM8991_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */ +#define WM8991_ROPGAVOL_SHIFT 0 +/* + * R34 (0x22) - Speaker Volume + */ +#define WM8991_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ +#define WM8991_SPKVOL_SHIFT 0 + +/* + * R35 (0x23) - ClassD1 + */ +#define WM8991_CDMODE 0x0100 /* CDMODE */ +#define WM8991_CDMODE_BIT 8 + +/* + * R37 (0x25) - ClassD3 + */ +#define WM8991_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */ +#define WM8991_DCGAIN_SHIFT 3 +#define WM8991_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ +#define WM8991_ACGAIN_SHIFT 0 +/* + * R39 (0x27) - Input Mixer1 + */ +#define WM8991_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */ +#define WM8991_AINLMODE_SHIFT 2 +#define WM8991_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */ +#define WM8991_AINRMODE_SHIFT 0 + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8991_LMP4 0x0080 /* LMP4 */ +#define WM8991_LMP4_BIT 7 /* LMP4 */ +#define WM8991_LMN3 0x0040 /* LMN3 */ +#define WM8991_LMN3_BIT 6 /* LMN3 */ +#define WM8991_LMP2 0x0020 /* LMP2 */ +#define WM8991_LMP2_BIT 5 /* LMP2 */ +#define WM8991_LMN1 0x0010 /* LMN1 */ +#define WM8991_LMN1_BIT 4 /* LMN1 */ +#define WM8991_RMP4 0x0008 /* RMP4 */ +#define WM8991_RMP4_BIT 3 /* RMP4 */ +#define WM8991_RMN3 0x0004 /* RMN3 */ +#define WM8991_RMN3_BIT 2 /* RMN3 */ +#define WM8991_RMP2 0x0002 /* RMP2 */ +#define WM8991_RMP2_BIT 1 /* RMP2 */ +#define WM8991_RMN1 0x0001 /* RMN1 */ +#define WM8991_RMN1_BIT 0 /* RMN1 */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8991_L34MNB 0x0100 /* L34MNB */ +#define WM8991_L34MNB_BIT 8 +#define WM8991_L34MNBST 0x0080 /* L34MNBST */ +#define WM8991_L34MNBST_BIT 7 +#define WM8991_L12MNB 0x0020 /* L12MNB */ +#define WM8991_L12MNB_BIT 5 +#define WM8991_L12MNBST 0x0010 /* L12MNBST */ +#define WM8991_L12MNBST_BIT 4 +#define WM8991_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */ +#define WM8991_LDBVOL_SHIFT 0 + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8991_R34MNB 0x0100 /* R34MNB */ +#define WM8991_R34MNB_BIT 8 +#define WM8991_R34MNBST 0x0080 /* R34MNBST */ +#define WM8991_R34MNBST_BIT 7 +#define WM8991_R12MNB 0x0020 /* R12MNB */ +#define WM8991_R12MNB_BIT 5 +#define WM8991_R12MNBST 0x0010 /* R12MNBST */ +#define WM8991_R12MNBST_BIT 4 +#define WM8991_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */ +#define WM8991_RDBVOL_SHIFT 0 + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8991_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */ +#define WM8991_LI2BVOL_SHIFT 6 +#define WM8991_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */ +#define WM8991_LR4BVOL_SHIFT 3 +#define WM8991_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */ +#define WM8991_LL4BVOL_SHIFT 0 + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8991_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */ +#define WM8991_RI2BVOL_SHIFT 6 +#define WM8991_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */ +#define WM8991_RL4BVOL_SHIFT 3 +#define WM8991_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */ +#define WM8991_RR4BVOL_SHIFT 0 + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8991_LRBLO 0x0080 /* LRBLO */ +#define WM8991_LRBLO_BIT 7 +#define WM8991_LLBLO 0x0040 /* LLBLO */ +#define WM8991_LLBLO_BIT 6 +#define WM8991_LRI3LO 0x0020 /* LRI3LO */ +#define WM8991_LRI3LO_BIT 5 +#define WM8991_LLI3LO 0x0010 /* LLI3LO */ +#define WM8991_LLI3LO_BIT 4 +#define WM8991_LR12LO 0x0008 /* LR12LO */ +#define WM8991_LR12LO_BIT 3 +#define WM8991_LL12LO 0x0004 /* LL12LO */ +#define WM8991_LL12LO_BIT 2 +#define WM8991_LDLO 0x0001 /* LDLO */ +#define WM8991_LDLO_BIT 0 + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8991_RLBRO 0x0080 /* RLBRO */ +#define WM8991_RLBRO_BIT 7 +#define WM8991_RRBRO 0x0040 /* RRBRO */ +#define WM8991_RRBRO_BIT 6 +#define WM8991_RLI3RO 0x0020 /* RLI3RO */ +#define WM8991_RLI3RO_BIT 5 +#define WM8991_RRI3RO 0x0010 /* RRI3RO */ +#define WM8991_RRI3RO_BIT 4 +#define WM8991_RL12RO 0x0008 /* RL12RO */ +#define WM8991_RL12RO_BIT 3 +#define WM8991_RR12RO 0x0004 /* RR12RO */ +#define WM8991_RR12RO_BIT 2 +#define WM8991_RDRO 0x0001 /* RDRO */ +#define WM8991_RDRO_BIT 0 + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8991_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */ +#define WM8991_LLI3LOVOL_SHIFT 6 +#define WM8991_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */ +#define WM8991_LR12LOVOL_SHIFT 3 +#define WM8991_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */ +#define WM8991_LL12LOVOL_SHIFT 0 + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8991_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */ +#define WM8991_RRI3ROVOL_SHIFT 6 +#define WM8991_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */ +#define WM8991_RL12ROVOL_SHIFT 3 +#define WM8991_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */ +#define WM8991_RR12ROVOL_SHIFT 0 + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8991_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */ +#define WM8991_LRI3LOVOL_SHIFT 6 +#define WM8991_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */ +#define WM8991_LRBLOVOL_SHIFT 3 +#define WM8991_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */ +#define WM8991_LLBLOVOL_SHIFT 0 + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8991_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */ +#define WM8991_RLI3ROVOL_SHIFT 6 +#define WM8991_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */ +#define WM8991_RLBROVOL_SHIFT 3 +#define WM8991_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */ +#define WM8991_RRBROVOL_SHIFT 0 + +/* + * R51 (0x33) - Out3/4 Mixer + */ +#define WM8991_VSEL_MASK 0x0180 /* VSEL - [8:7] */ +#define WM8991_LI4O3 0x0020 /* LI4O3 */ +#define WM8991_LI4O3_BIT 5 +#define WM8991_LPGAO3 0x0010 /* LPGAO3 */ +#define WM8991_LPGAO3_BIT 4 +#define WM8991_RI4O4 0x0002 /* RI4O4 */ +#define WM8991_RI4O4_BIT 1 +#define WM8991_RPGAO4 0x0001 /* RPGAO4 */ +#define WM8991_RPGAO4_BIT 0 +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8991_LLOPGALON 0x0040 /* LLOPGALON */ +#define WM8991_LLOPGALON_BIT 6 +#define WM8991_LROPGALON 0x0020 /* LROPGALON */ +#define WM8991_LROPGALON_BIT 5 +#define WM8991_LOPLON 0x0010 /* LOPLON */ +#define WM8991_LOPLON_BIT 4 +#define WM8991_LR12LOP 0x0004 /* LR12LOP */ +#define WM8991_LR12LOP_BIT 2 +#define WM8991_LL12LOP 0x0002 /* LL12LOP */ +#define WM8991_LL12LOP_BIT 1 +#define WM8991_LLOPGALOP 0x0001 /* LLOPGALOP */ +#define WM8991_LLOPGALOP_BIT 0 +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8991_RROPGARON 0x0040 /* RROPGARON */ +#define WM8991_RROPGARON_BIT 6 +#define WM8991_RLOPGARON 0x0020 /* RLOPGARON */ +#define WM8991_RLOPGARON_BIT 5 +#define WM8991_ROPRON 0x0010 /* ROPRON */ +#define WM8991_ROPRON_BIT 4 +#define WM8991_RL12ROP 0x0004 /* RL12ROP */ +#define WM8991_RL12ROP_BIT 2 +#define WM8991_RR12ROP 0x0002 /* RR12ROP */ +#define WM8991_RR12ROP_BIT 1 +#define WM8991_RROPGAROP 0x0001 /* RROPGAROP */ +#define WM8991_RROPGAROP_BIT 0 + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8991_LB2SPK 0x0080 /* LB2SPK */ +#define WM8991_LB2SPK_BIT 7 +#define WM8991_RB2SPK 0x0040 /* RB2SPK */ +#define WM8991_RB2SPK_BIT 6 +#define WM8991_LI2SPK 0x0020 /* LI2SPK */ +#define WM8991_LI2SPK_BIT 5 +#define WM8991_RI2SPK 0x0010 /* RI2SPK */ +#define WM8991_RI2SPK_BIT 4 +#define WM8991_LOPGASPK 0x0008 /* LOPGASPK */ +#define WM8991_LOPGASPK_BIT 3 +#define WM8991_ROPGASPK 0x0004 /* ROPGASPK */ +#define WM8991_ROPGASPK_BIT 2 +#define WM8991_LDSPK 0x0002 /* LDSPK */ +#define WM8991_LDSPK_BIT 1 +#define WM8991_RDSPK 0x0001 /* RDSPK */ +#define WM8991_RDSPK_BIT 0 + +/* + * R55 (0x37) - Additional Control + */ +#define WM8991_VROI 0x0001 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8991_DIS_LLINE 0x0020 /* DIS_LLINE */ +#define WM8991_DIS_RLINE 0x0010 /* DIS_RLINE */ +#define WM8991_DIS_OUT3 0x0008 /* DIS_OUT3 */ +#define WM8991_DIS_OUT4 0x0004 /* DIS_OUT4 */ +#define WM8991_DIS_LOUT 0x0002 /* DIS_LOUT */ +#define WM8991_DIS_ROUT 0x0001 /* DIS_ROUT */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8991_SOFTST 0x0040 /* SOFTST */ +#define WM8991_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8991_BUFDCOPEN 0x0004 /* BUFDCOPEN */ +#define WM8991_POBCTRL 0x0002 /* POBCTRL */ +#define WM8991_VMIDTOG 0x0001 /* VMIDTOG */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8991_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */ +#define WM8991_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */ +#define WM8991_MCD 0x0004 /* MCD */ +#define WM8991_MBSEL 0x0001 /* MBSEL */ + +/* + * R60 (0x3C) - PLL1 + */ +#define WM8991_SDM 0x0080 /* SDM */ +#define WM8991_PRESCALE 0x0040 /* PRESCALE */ +#define WM8991_PLLN_MASK 0x000F /* PLLN - [3:0] */ + +/* + * R61 (0x3D) - PLL2 + */ +#define WM8991_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */ + +/* + * R62 (0x3E) - PLL3 + */ +#define WM8991_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ + +/* + * R63 (0x3F) - Internal Driver Bits + */ +#define WM8991_INMIXL_PWR_BIT 0 +#define WM8991_AINLMUX_PWR_BIT 1 +#define WM8991_INMIXR_PWR_BIT 2 +#define WM8991_AINRMUX_PWR_BIT 3 + +#define WM8991_MCLK_DIV 0 +#define WM8991_DACCLK_DIV 1 +#define WM8991_ADCCLK_DIV 2 +#define WM8991_BCLK_DIV 3 + +#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ + tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +#endif /* _WM8991_H */ -- 1.7.3.5

3 2

[alsa-devel] [PATCH 4/4] ASoC: soc core allow machine driver to register the card
by Koul, Vinod 17 Jan '11

17 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> The machine driver can't register the card directly and need to do this thru soc-audio device creation This patch allows the register and unregister card to be directly called by machine drivers Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 541ddfa..9952254 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -258,6 +258,8 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_register_card(struct snd_soc_card *card); +int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8305712..69117b6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -58,8 +58,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int snd_soc_register_card(struct snd_soc_card *card); -static int snd_soc_unregister_card(struct snd_soc_card *card); static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* @@ -1870,6 +1868,13 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; + /* + * no card, so machine driver should be registering card + * we should not be here in that case so ret error + */ + if (!card) + return -EINVAL; + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; snd_soc_initialize_card_lists(card); @@ -3105,11 +3110,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); * * @card: Card to register * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_register_card(struct snd_soc_card *card) +int snd_soc_register_card(struct snd_soc_card *card) { int i; @@ -3141,17 +3143,15 @@ static int snd_soc_register_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core * * @card: Card to unregister * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_unregister_card(struct snd_soc_card *card) +int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); @@ -3162,6 +3162,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_unregister_card); /* * Simplify DAI link configuration by removing ".-1" from device names -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ASoC: Avoid direct register cache access when setting defaults
by Mark Brown 17 Jan '11

17 Jan '11

Directly accessing the register cache means that we can't use anything except a flat register cache so use snd_soc_update_bits(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 6 +++--- sound/soc/codecs/wm8741.c | 13 ++++++++----- sound/soc/codecs/wm8904.c | 41 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm8955.c | 27 +++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 30 ++++++++++++++++++++---------- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm9090.c | 41 ++++++++++++++++++++++++----------------- 7 files changed, 101 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5eb2f50..9b8603f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -414,7 +414,6 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret, i; codec->hw_write = (hw_write_t)i2c_master_send; @@ -471,8 +470,9 @@ static int wm8523_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; - reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + snd_soc_update_bits(codec, WM8523_DAC_GAINR, + WM8523_DACR_VU, WM8523_DACR_VU); + snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC); wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 494f2d3..25af901 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -421,7 +421,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); @@ -437,10 +436,14 @@ static int wm8741_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU */ - reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; - reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, ARRAY_SIZE(wm8741_snd_controls)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9de44a4..86c1cb8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2436,19 +2436,28 @@ static int wm8904_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; - reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; - reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | - WM8904_HPOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | - WM8904_HPOUTRZC; - reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTRZC; - reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); /* Apply configuration from the platform data. */ if (wm8904->pdata) { @@ -2469,10 +2478,12 @@ static int wm8904_probe(struct snd_soc_codec *codec) /* Set Class W by default - this will be managed by the Class * G widget at runtime where bypass paths are available. */ - reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + snd_soc_update_bits(codec, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); /* Use normal bias source */ - reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 7167dfc..5e0214d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -934,16 +934,27 @@ static int wm8955_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; - reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; - reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; - reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; - reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; - reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; - reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + snd_soc_update_bits(codec, WM8955_LEFT_DAC_VOLUME, + WM8955_LDVU, WM8955_LDVU); + snd_soc_update_bits(codec, WM8955_RIGHT_DAC_VOLUME, + WM8955_RDVU, WM8955_RDVU); + snd_soc_update_bits(codec, WM8955_LOUT1_VOLUME, + WM8955_LO1VU | WM8955_LO1ZC, + WM8955_LO1VU | WM8955_LO1ZC); + snd_soc_update_bits(codec, WM8955_ROUT1_VOLUME, + WM8955_RO1VU | WM8955_RO1ZC, + WM8955_RO1VU | WM8955_RO1ZC); + snd_soc_update_bits(codec, WM8955_LOUT2_VOLUME, + WM8955_LO2VU | WM8955_LO2ZC, + WM8955_LO2VU | WM8955_LO2ZC); + snd_soc_update_bits(codec, WM8955_ROUT2_VOLUME, + WM8955_RO2VU | WM8955_RO2ZC, + WM8955_RO2VU | WM8955_RO2ZC); + snd_soc_update_bits(codec, WM8955_MONOOUT_VOLUME, + WM8955_MOZC, WM8955_MOZC); /* Also enable adaptive bass boost by default */ - reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + snd_soc_update_bits(codec, WM8955_BASS_CONTROL, WM8955_BB, WM8955_BB); /* Set platform data values */ if (pdata) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b9cb1fc..c827057 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3822,16 +3822,26 @@ static int wm8962_probe(struct snd_soc_codec *codec) } /* Latch volume update bits */ - reg_cache[WM8962_LEFT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_RIGHT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_LEFT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_RIGHT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_LEFT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_RIGHT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_SPKOUTL_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_SPKOUTR_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_HPOUTL_VOLUME] |= WM8962_HPOUT_VU; - reg_cache[WM8962_HPOUTR_VOLUME] |= WM8962_HPOUT_VU; + snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); wm8962_add_widgets(codec); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc344..30fb48e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -965,7 +965,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) * written. */ for (i = 0; i < ARRAY_SIZE(update_reg); i++) - ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); /* Reset the codec */ ret = snd_soc_write(codec, WM8978_RESET, 0); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a788c42..3c5d325 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -551,7 +551,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret; codec->control_data = wm9090->control_data; @@ -576,22 +575,30 @@ static int wm9090_probe(struct snd_soc_codec *codec) /* Configure some defaults; they will be written out when we * bring the bias up. */ - reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1A_ZC; - reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1B_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2A_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2B_ZC; - reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |= - WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC; - reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC; - reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC; - - reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA; + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_A_VOLUME, + WM9090_IN1_VU | WM9090_IN1A_ZC, + WM9090_IN1_VU | WM9090_IN1A_ZC); + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_B_VOLUME, + WM9090_IN1_VU | WM9090_IN1B_ZC, + WM9090_IN1_VU | WM9090_IN1B_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_A_VOLUME, + WM9090_IN2_VU | WM9090_IN2A_ZC, + WM9090_IN2_VU | WM9090_IN2A_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_B_VOLUME, + WM9090_IN2_VU | WM9090_IN2B_ZC, + WM9090_IN2_VU | WM9090_IN2B_ZC); + snd_soc_update_bits(codec, WM9090_SPEAKER_VOLUME_LEFT, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC); + snd_soc_update_bits(codec, WM9090_LEFT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC); + snd_soc_update_bits(codec, WM9090_RIGHT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC); + + snd_soc_update_bits(codec, WM9090_CLOCKING_1, + WM9090_TOCLK_ENA, WM9090_TOCLK_ENA); wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ALSA : HDA - add multi-streaming playback of front panel headphone and rear panel audio jacks at same time for hda ad1988
by Raymond Yau 17 Jan '11

17 Jan '11

Attached a patch which add a new model to support multi-streaming playback for ad1988 playback another stereo stream through the front panel headphone on device 2 while playback through the speakers connected to rear panel on device 0 at the same time. Tested with ad1988a rev2 codec on asus P5B-V motherboard Signed-off-by : Raymond Yau < superquad.vortex2 <at> gmail.com >

2 1

[alsa-devel] [PATCH] ALSA : au88x0 - fix playback/capture channels of au8830
by Raymond Yau 17 Jan '11

17 Jan '11

patch to change supported playback channels of au8830 to 1,2,4 and capture channels to 1,2 This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to set 3 channels

2 1

[alsa-devel] Card driver: device or resource busy
by torsten.schenk 17 Jan '11

17 Jan '11

Hello, I'm currently developing a driver for a card. PCM playback works so far, I just encountered some playback issues when firefox (flash) and another alsa client wanted to play back at the same time. I figured out that the difference between my driver and existing ones is, that my driver only allows one client and others multiple. I get a "Device or resource busy" when I play back two times the same file (also different ones of course) with mplayer f.ex. I was able to track this down to the fact, that (only) the O_APPEND Flag is missing in file->f_flags (snd_pcm_attach_substream), but was completely unable to find out where this flag comes from in other drivers. Another point (perhaps not unrelated to this) is, that I get a bug (see below) that seems to tell me, that prepare cannot schedule. Contrary to the documentation that says that prepare is now non-atomic. It is useful to wait in the prepare function since I need to set the sample rate and wait for a reply (USB device). Before I rewrite it, i wanted to ask about atomicity in prepare on this ML. Greetings, Torsten [ 3662.318321] usb 1-2: unlink qh0-00ff/f69a5300 start 0 [2/0 us] [ 3662.318434] BUG: scheduling while atomic: aplay/17902/0x00000002 [ 3662.318435] Modules linked in: snd_usb_6fire snd_rawmidi snd_seq_device snd_pcm snd_timer snd soundcore snd_page_alloc nvidia(P) [last unloaded: soundcore] [ 3662.318443] Pid: 17902, comm: aplay Tainted: P 2.6.36-gentoo-r5 #3 [ 3662.318445] Call Trace: [ 3662.318448] [<c1025fa0>] __schedule_bug+0x48/0x4d [ 3662.318451] [<c136f438>] schedule+0x9d/0x6b6 [ 3662.318454] [<c10383f6>] ? __mod_timer+0xf5/0x100 [ 3662.318457] [<c136fc83>] schedule_timeout+0x73/0x90 [ 3662.318460] [<c1037ffe>] ? process_timeout+0x0/0xa [ 3662.318463] [<fa335622>] pcm_alsa_prepare+0x353/0x3a5 [snd_usb_6fire] [ 3662.318467] [<c1043121>] ? autoremove_wake_function+0x0/0x2f [ 3662.318472] [<fa2fb700>] snd_pcm_do_prepare+0xc/0x1c [snd_pcm] [ 3662.318476] [<fa2fb4d3>] snd_pcm_action_single+0x25/0x4b [snd_pcm] [ 3662.318481] [<fa2fc4cc>] snd_pcm_action_nonatomic+0x42/0x55 [snd_pcm] [ 3662.318486] [<fa2fe0c9>] snd_pcm_common_ioctl1+0x379/0xa19 [snd_pcm] [ 3662.318489] [<c1086437>] ? find_get_page+0x6b/0x74 [ 3662.318492] [<c10259f4>] ? get_parent_ip+0xb/0x31 [ 3662.318497] [<fa2fed24>] snd_pcm_playback_ioctl1+0x2d4/0x2eb [snd_pcm] [ 3662.318500] [<c1168866>] ? file_has_perm+0x7c/0x85 [ 3662.318505] [<fa2fed5b>] snd_pcm_playback_ioctl+0x20/0x2d [snd_pcm] [ 3662.318513] [<fa2fed3b>] ? snd_pcm_playback_ioctl+0x0/0x2d [snd_pcm] [ 3662.318515] [<c10b9e1a>] do_vfs_ioctl+0x43c/0x481 [ 3662.318519] [<c1168a26>] ? selinux_file_ioctl+0x3e/0x41 [ 3662.318522] [<c10b9ea0>] sys_ioctl+0x41/0x61 [ 3662.318525] [<c100278c>] sysenter_do_call+0x12/0x22

3 2

[PATCH] ASoC: Add FreeScale SGTL5000 codec support
by Zeng Zhaoming 17 Jan '11

17 Jan '11

add FreeScale SGTL5000 codec support Signed-off-by: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sgtl5000.c | 1016 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 403 +++++++++++++++++ 4 files changed, 1424 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sgtl5000.c create mode 100644 sound/soc/codecs/sgtl5000.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883a312..19bbb73 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SGTL5000 if I2C select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS @@ -176,6 +177,9 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +config SND_SOC_SGTL5000 + tristate + config SND_SOC_SPDIF tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 579af9c..78e36cc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 0000000..9fe474f --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1016 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * 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/slab.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <mach/hardware.h> + +#include "sgtl5000.h" + +/* default value of all sgtl5000 registers */ +static const u16 sgtl5000_regs[] = { + /* 0x0 0x2 0x4 0x6 0x8 0xa 0xc 0xe */ + 0xa011, 0x0000, 0x0000, 0x0000, 0x0008, 0x0000, 0x0010, 0x0000,/* 0x */ + 0x0010, 0x0000, 0x0010, 0x0000, 0x0000, 0x0000, 0x323c, 0x0000,/* 1x */ + 0x3c3c, 0x0000, 0x3c3c, 0x0000, 0x555f, 0x0000, 0x0000, 0x0000,/* 2x */ + 0x0000, 0x0000, 0x0000, 0x0000, 0x408c, 0x0000, 0x0008, 0x0000,/* 3x */ + 0x0000, 0x0000, 0x1818, 0x0000, 0x0111, 0x0000, 0x0000, 0x0000,/* 4x */ + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0404, 0x0000,/* 5x */ + 0x7060, 0x0000, 0x5000, 0x0000, 0x0000, 0x0000, 0x0017, 0x0000,/* 6x */ + 0x01c0, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,/* 7x */ +}; + +/* regulator supplies for sgtl5000, VDDD is an option external supply */ +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + + +static const char *supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +/* sgtl5000 private structure in codec */ +struct sgtl5000_priv { + int sysclk; /* sysclk rate */ + int master; /* i2s master or not? */ + int fmt; /* i2s data format */ + int lrclk; /* frame clock rate */ + int capture_channels; /* the num of channels for capture. */ + int small_pop; /* hw assistant to eliminate pop */ + struct regulator *supplies[SGTL5000_SUPPLY_NUM]; /* all regulators. */ +}; + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* change mic bias resistor to 4Kohm */ + snd_soc_write(w->codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_4k); + + return 0; +} + +static int dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP|SGTL5000_DAC_POWERUP, + SGTL5000_VAG_POWERUP|SGTL5000_DAC_POWERUP); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_DAC_POWERUP, 0); + break; + default: + break; + } + + return 0; +} + +/* + * if small_pop is enabled, hp_powerup or lineout_powerup + * should be set before vag_powerup be cleared, this introduce + * 200~400ms latency + */ +static int small_pop_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct sgtl5000_priv *priv = snd_soc_codec_get_drvdata(w->codec); + int reg; + + if (!priv->small_pop) + return 0; + + reg = snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER); + if (reg & SGTL5000_VAG_POWERUP) { + reg &= ~SGTL5000_VAG_POWERUP; + snd_soc_write(w->codec, SGTL5000_CHIP_ANA_POWER, reg); + msleep(400); + } + + return 0; +} + +static int adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_DAC_POWERUP, + SGTL5000_DAC_POWERUP); + + return 0; +} + +/* input sources for ADC */ +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +/* input sources for DAC */ +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + small_pop_event, SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, + small_pop_event, SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + /* aif for i2s input */ + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", + 0, SGTL5000_CHIP_DIG_POWER, + 0, 0), + + /* aif for i2s output */ + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", + 0, SGTL5000_CHIP_DIG_POWER, + 1, 0), + + SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_DIG_POWER, 6, 0, + adc_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_DIG_POWER, 5, 0, + dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +/* routes for sgtl5000 */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ + {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + + {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ + {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + + {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ + {"LO", NULL, "DAC"}, /* dac --> line_out */ + + {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ + {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ + + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +/* custom function to fetch info of PCM playback volume */ +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +/* custom function to get of PCM playback volume */ +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + l = l < 0x3c ? 0x3c : l; + l = l > 0xfc ? 0xfc : l; + r = r < 0x3c ? 0x3c : r; + r = r > 0xfc ? 0xfc : r; + + /* revert it */ + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +/* custom put function for PCM playback volume */ +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg, l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + l = l < 0 ? 0 : l; + l = l > 0xfc - 0x3c ? 0xfc - 0x3c : l; + r = r < 0 ? 0 : r; + r = r > 0xfc - 0x3c ? 0xfc - 0x3c : r; + l = 0xfc - l; + r = 0xfc - r; + + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg); + + return 0; +} + +/* tlv for mic gain, 0db 20db 30db 40db */ +static const unsigned int mic_gain_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), +}; + +/* tlv for hp volume, -51.5db to 12.0db, step .5db */ +static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_SINGLE("Capture Attenuate Switch (-6db)", + SGTL5000_CHIP_ANA_ADC_CTRL, + 8, 1, 0), + + /* we need SOC_DOUBLE_S8_TLV with invert */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = dac_info_volsw, + .get = dac_get_volsw, + .put = dac_put_volsw, + }, + + SOC_DOUBLE_TLV("Headphone Playback Volume", + SGTL5000_CHIP_ANA_HP_CTRL, + 0, 8, + 0x7f, 1, + headphone_volume), + + SOC_SINGLE_TLV("Mic Volume", + SGTL5000_CHIP_MIC_CTRL, + 0, 4, 0, mic_gain_tlv), +}; + +/* mutt the codec used by alsa core */ +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + if (mute) + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, adcdac_ctrl); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, 0); + + return 0; +} + +/* set codec format */ +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + /* + * i2s clock and frame master setting. + * ONLY support: + * - clock and frame slave, + * - clock and frame master + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + break; + default: + return -EINVAL; + } + + /* setting i2s data format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +/* set codec sysclk */ +static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_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 sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int clk_ctl = 0; + int pll_ctl = 0; + int i2s_ctl; + int div2 = 0; + int reg; + int sys_fs; + + /* sysclk should already set */ + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + /* power up adc, and set it accroding to stereo or not */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + sgtl5000->capture_channels = channels; + + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + reg |= SGTL5000_ADC_POWERUP; + + if (sgtl5000->capture_channels == 1) + reg &= ~SGTL5000_ADC_STEREO; + else + reg |= SGTL5000_ADC_STEREO; + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, reg); + } + + /* get frame clock rate */ + sgtl5000->lrclk = params_rate(params); + + /* + * frame clock is divided from sys fs, + * if frame clock lower than 44.1khz, sys fs should set to + * 32khz or 44.1khz. + */ + switch (sgtl5000->lrclk) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = sgtl5000->lrclk; + break; + } + + /* get devided factor of frame clock */ + switch (sys_fs / sgtl5000->lrclk) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + default: + break; + } + + /* set the sys_fs accroding to frame clock */ + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "sample rate %d not supported\n", + sgtl5000->lrclk); + return -EINVAL; + } + + /* if codec is slave, just set register to match master parameters */ + if (!sgtl5000->master) { + sys_fs = sgtl5000->lrclk; + clk_ctl = SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + if (sys_fs * 256 == sgtl5000->sysclk) + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << + SGTL5000_MCLK_FREQ_SHIFT; + else if (sys_fs * 384 == sgtl5000->sysclk && sys_fs != 96000) + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << + SGTL5000_MCLK_FREQ_SHIFT; + else if (sys_fs * 512 == sgtl5000->sysclk && sys_fs != 96000) + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << + SGTL5000_MCLK_FREQ_SHIFT; + else { + pr_err("%s: PLL not supported in slave mode\n", + __func__); + return -EINVAL; + } + } else + /* if codec is master, use PLL to generate sys_fs */ + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << SGTL5000_MCLK_FREQ_SHIFT; + + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + unsigned int in, int_div, frac_div; + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + } + + i2s_ctl = snd_soc_read(codec, SGTL5000_CHIP_I2S_CTRL); + + /* set i2s data format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "fs=%d,clk_ctl=%d,pll_ctl=%d,i2s_ctl=%d,div2=%d\n", + sgtl5000->lrclk, clk_ctl, pll_ctl, i2s_ctl, div2); + + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + if (div2) + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + SGTL5000_INPUT_FREQ_DIV2); + else + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + 0); + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + } + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl); + + return 0; +} + +/* + * set dac bias + * common state changes: + * startup: + * off --> standby --> prepare --> on + * standby --> prepare --> on + * + * stop: + * on --> prepare --> standby + */ +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + codec->bias_level = level; + return 0; +} + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + /* + * only support 8~48K + 96K, + * TODO modify hw_param to support more + */ + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + return 1; + } + + return 0; +} + +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sgtl5000_restore_reg(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + return snd_soc_write(codec, reg, cache[reg >> 1]); +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + int i; + + /* Restore refs first in same order as in sgtl5000_probe */ + sgtl5000_restore_reg(codec, SGTL5000_CHIP_LINREG_CTRL); + sgtl5000_restore_reg(codec, SGTL5000_CHIP_ANA_POWER); + msleep(10); + sgtl5000_restore_reg(codec, SGTL5000_CHIP_REF_CTRL); + sgtl5000_restore_reg(codec, SGTL5000_CHIP_LINE_OUT_CTRL); + + /* Restore everythine else */ + for (i = 0; i < ARRAY_SIZE(sgtl5000_regs); i++) + sgtl5000_restore_reg(codec, i); + + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* Bring the codec back up to standby first to minimise pop/clicks */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 reg, ana_pwr, lreg_ctrl; + int vag; + int ret; + int vddd, vdda, vddio; + int rev; + int i; + + /* setup i2c data ops */ + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* get and enable all regulators */ + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + struct regulator *reg; + reg = regulator_get(codec->dev, supply_names[i]); + + if (IS_ERR(reg)) + continue; + + regulator_enable(reg); + sgtl5000->supplies[i] = reg; + } + + /* vdda and vddio regulator must configed */ + if (!sgtl5000->supplies[VDDA] || !sgtl5000->supplies[VDDIO]) { + dev_err(codec->dev, + "Not set vdda or vddio regulator correctly\n"); + /* FIXME, no such platform regulator configed */ + return -ENODEV; + } + + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto err_out; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + + /* reset value */ + ana_pwr = SGTL5000_DAC_STEREO | + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP | + SGTL5000_ADC_STEREO | SGTL5000_REFTOP_POWERUP; + lreg_ctrl = 0; + + vdda = regulator_get_voltage(sgtl5000->supplies[VDDA]) / 1000; + vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO]) / 1000; + + if (regulator_get_voltage(sgtl5000->supplies[VDDD])) + vddd = regulator_get_voltage(sgtl5000->supplies[VDDD]) / 1000; + else + vddd = 0; + + /* workaround for rev 0x11: use vddd linear regulator */ + if (!vddd || (rev >= 0x11)) { + /* set VDDD to 1.2v */ + lreg_ctrl |= 0x8 << SGTL5000_LINREG_VDDD_SHIFT; + /* power internal linear regulator */ + ana_pwr |= SGTL5000_LINEREG_D_POWERUP; + } else { + /* turn of startup power */ + ana_pwr &= ~SGTL5000_STARTUP_POWERUP; + ana_pwr &= ~SGTL5000_LINREG_SIMPLE_POWERUP; + } + + if (vddio < 3100 && vdda < 3100) { + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } + + if (vddio >= 3100 && vdda >= 3100) { + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + /* If PLL is powered up (such as on power cycle) leave it on. */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr |= reg & (SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* set ADC/DAC ref voltage to vdda / 2 */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + /* set line out ref voltage to vddio / 2 */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + SGTL5000_LINE_OUT_GND_STP * + SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + msleep(10); + + /* For rev 0x11, if vddd linear reg has been enabled, we have + to disable simple reg to get proper VDDD voltage. */ + if ((ana_pwr & SGTL5000_LINEREG_D_POWERUP) && (rev >= 0x11)) { + ana_pwr &= ~SGTL5000_LINREG_SIMPLE_POWERUP; + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + msleep(10); + } + + /* enable small pop, introduce 200~400ms delay in turning on/off */ + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT | SGTL5000_SMALL_POP); + + sgtl5000->small_pop = 1; + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + /* enable short cut detect */ + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + + /* + * set sound switch + * TODO: add sound switch to control and dapm widge. + */ + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, 0); + + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + reg = snd_soc_read(codec, SGTL5000_CHIP_ANA_ADC_CTRL); + reg &= ~SGTL5000_ADC_VOL_M6DB; + reg &= ~(SGTL5000_ADC_VOL_LEFT_MASK | SGTL5000_ADC_VOL_RIGHT_MASK); + reg |= (0xf << SGTL5000_ADC_VOL_LEFT_SHIFT) + | (0xf << SGTL5000_ADC_VOL_RIGHT_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_ANA_ADC_CTRL, reg); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_CLK_TOP_CTRL, 0); + + /* + * disable DAP + * TODO: + * Enable DAP in control and dapm. + */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* leading to standby state */ + ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto err_out; + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(codec, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + + return 0; + +err_out: + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + if (!sgtl5000->supplies[i]) + continue; + + regulator_disable(sgtl5000->supplies[i]); + regulator_put(sgtl5000->supplies[i]); + } + + return ret; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int i; + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_dapm_free(codec); + + for (i = 0; i < SGTL5000_SUPPLY_NUM; i++) { + if (!sgtl5000->supplies[i]) + continue; + + regulator_disable(sgtl5000->supplies[i]); + regulator_put(sgtl5000->supplies[i]); + } + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + i2c_set_clientdata(client, sgtl5000); + + msleep(1); + + ret = snd_soc_register_codec(&client->dev, + &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(sgtl5000); + return ret; + } + + return 0; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000-codec", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000-codec", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("ASoC sgtl5000 driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 0000000..7f24655 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,403 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2008-2009 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _SGTL5000_H +#define _SGTL5000_H + +#include <linux/i2c.h> + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- 1.7.0.4 --hQiwHBbRI9kgIhsi--

1 0

[alsa-devel] jack-sense where is it?
by Paulo Silva 16 Jan '11

16 Jan '11

Hi, I'm not sure if this is the right place to post this, so forgive me if I'm making a mistake. I'm trying to find where in the source of alsa kernel modules is the part that enumerates the jacks (outputs) of the sound card. This is because I have two headphone jacks and since I upgraded to 2.6.26, when I plug anything to the 2nd headphone the first gets muted. I think this is a jack sensing related bug, since the 1st jack is enumerated as "Front" and the 2nd as "Headphone" when in reality they are both "Headphone" 1 and 2. If someone can give me a hint to where the code related with the enumeration of the jack sensing is located, then I could make some tests myself. I just want to save some work searching for it. Regards, paulo

2 1

Re: [alsa-devel] [PATCH] sound: constify functions in ac97
by Takashi Iwai 14 Jan '11

14 Jan '11

At Wed, 12 Jan 2011 14:44:07 +0200, Alexey Dobriyan wrote: > > On Wed, Jan 12, 2011 at 1:03 PM, Hanno Böck <hanno(a)hboeck.de> wrote: > > struct snd_ac97_build_ops { > > - int (*build_3d) (struct snd_ac97 *ac97); > > - int (*build_specific) (struct snd_ac97 *ac97); > > - int (*build_spdif) (struct snd_ac97 *ac97); > > - int (*build_post_spdif) (struct snd_ac97 *ac97); > > + int (* const build_3d) (struct snd_ac97 *ac97); > > + int (* const build_specific) (struct snd_ac97 *ac97); > > + int (* const build_spdif) (struct snd_ac97 *ac97); > > + int (* const build_post_spdif) (struct snd_ac97 *ac97); > > #ifdef CONFIG_PM > > - void (*suspend) (struct snd_ac97 *ac97); > > - void (*resume) (struct snd_ac97 *ac97); > > + void (* const suspend) (struct snd_ac97 *ac97); > > + void (* const resume) (struct snd_ac97 *ac97); > > #endif > > - void (*update_jacks) (struct snd_ac97 *ac97); /* for jack-sharing */ > > + /* for jack-sharing */ > > + void (* const update_jacks) (struct snd_ac97 *ac97); > > Can we please not do this. > > You made whole ops structure RO, this is enough. Yes. Hanno, could you fix and resubmit? thanks, Takashi

2 2

[alsa-devel] [PATCH] ASoC: tegra: Remove TEGRA_I2S_AUDIO from Kconfig
by Stephen Warren 14 Jan '11

14 Jan '11

That config variable doesn't exist in the mainline kernel, and hence the dependency shouldn't either. In linux-tegra-2.6.36, the dependency did exist to avoid a conflict with the old non-ALSA Tegra I2S driver. However, this isn't and won't be upstreamed. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Kconfig | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 5514f1c..66b504f 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -8,7 +8,6 @@ config SND_TEGRA_SOC config SND_TEGRA_SOC_I2S tristate depends on SND_TEGRA_SOC - depends on !TEGRA_I2S_AUDIO default m help Say Y or M if you want to add support for codecs attached to the -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: tegra: Add DAPM widgets/routes for Harmony
by Stephen Warren 14 Jan '11

14 Jan '11

With this change, I can capture from a microphone plugged into the mic jack on Harmony (after unmuting Left Input PGA, and maybe turning up the gain there too). Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Question: Harmony has a mono mic attached to the left input channel. However, the I2S controller only supports stereo. Is it possible to duplicate the mic input to both channels in ALSA? I know the WM8903 can do it via the WM8903_AUDIO_INTERFACE_0 register, but I'm not sure how to request that in ALSA-speak. It looks like this same setup applies to many other platforms, but I couldn't see how the issue was solved, if at all. sound/soc/tegra/harmony.c | 31 +++++++++++++++++++++++++++++++ 1 files changed, 31 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index bf0dbaf..b160b71 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -107,6 +107,36 @@ static struct snd_soc_ops harmony_asoc_ops = { .hw_params = harmony_asoc_hw_params, }; +static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route harmony_audio_map[] = { + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "HPOUTL"}, + {"Mic Bias", NULL, "Mic Jack"}, + {"IN1L", NULL, "Mic Bias"}, +}; + +static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets, + ARRAY_SIZE(harmony_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, harmony_audio_map, + ARRAY_SIZE(harmony_audio_map)); + + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_sync(dapm); + + return 0; +} + static struct snd_soc_dai_link harmony_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", @@ -114,6 +144,7 @@ static struct snd_soc_dai_link harmony_wm8903_dai = { .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", + .init = harmony_asoc_init, .ops = &harmony_asoc_ops, }; -- 1.7.0.4

3 3

[alsa-devel] question on snd_soc_jack_report
by Harsha, Priya 14 Jan '11

14 Jan '11

Hi, I am using snd_soc_jack_report and I send the parameter as follows to report headphones. struct snd_soc_jack *soc_jack; snd_soc_jack_report (soc_jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); I don't find the reporting to be happening. When I looked at the snd_jack_report code, I found that jack->type needs to be set and that was not being set. So when I added the following line in my code, it started reporting. soc_jack->jack->type = SND_JACK_HEADPHONE; Is it expected that the driver sets the 'type' of the instance of 'snd_jack' inside snd_soc_jack structure. I was actually thinking that snd_soc_jack_report would need to do this. I did not find any other drivers doing that. Is there something that I am missing? Also, to report button press events, I had to add the following line of code soc_jack->jack->key[0] = BTN_0; Is this also an expected thing to be set. -Harsha

2 1

[alsa-devel] [PATCH] ALSA: hda: Disable odd numbered channels on HDMI controllers.
by Nitin Daga 14 Jan '11

14 Jan '11

Added code in hda_intel.c to disable odd numbered channels not supported by HDMI controllers. Signed-off-by: Nitin Daga <ndaga(a)nvidia.com> Acked-By: Stephen Warren <swarren(a)nvidia.com> --- pci/hda/hda_intel.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/pci/hda/hda_intel.c b/pci/hda/hda_intel.c index a78ea34..293c685 100644 --- a/pci/hda/hda_intel.c +++ b/pci/hda/hda_intel.c @@ -1539,6 +1539,16 @@ struct azx_pcm { struct hda_pcm_stream *hinfo[2]; }; +static unsigned int channels_2_4_6_8[] = { + 2, 4, 6, 8 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_4_6_8_channels = { + .count = ARRAY_SIZE(channels_2_4_6_8), + .list = channels_2_4_6_8, + .mask = 0, +}; + static int azx_pcm_open(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); @@ -1566,6 +1576,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 128); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_constraints_2_4_6_8_channels); snd_hda_power_up(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { -- 1.7.0.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

3 4

[alsa-devel] Changes on Realtek fix-up handling
by Takashi Iwai 14 Jan '11

14 Jan '11

Hi, there were slight changes in the handling of fix-up's in HD-audio patch_realtek.c. Now, the fixup struct is like: struct alc_fixup { int type; bool chained; int chain_id; union { unsigned int sku; const struct alc_pincfg *pins; const struct hda_verb *verbs; void (*func)(struct hda_codec *codec, const struct alc_fixup *fix, int action); } v; }; and each fixup entry has only one type, either sku, pins, verbs or func specified by the type field. If chained is set, the entry corresponding to chain_id is evaluated at next, so fixup entries can be reused if an additional change is required to the existing one. The callback function can take "action" instead of "pre_init" flag. This can be either ALC_FIXUP_ACT_PRE_PROBE, ALC_FIXUP_ACT_PROBE or AL_FIXUP_ACT_INIT. The first two are the cases pre_init=1 and pre_init=0, i.e. the fix-up is handled during probing only once. The last one means that the fix-up is called in the init callback, i.e. at initialization during the first loading and PM resume. My plan is to move from the quirk entries to fixup entries with auto-parse mode in more cases, and the above changes are of help for that. thanks, Takashi

1 0

[alsa-devel] [PATCH 1/4] ASoC: soc core move card cleanup from soc_remove()
by Koul, Vinod 14 Jan '11

14 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> In soc_remove() the card resources are cleaned up. This can also be done in card_unregister() This patch move this cleanup into a new function and calls it from card_unregister. This paves way for further work to allow card registartion from machine. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/soc-core.c | 51 ++++++++++++++++++++++++++++--------------------- 1 files changed, 29 insertions(+), 22 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30d76e8..318c3a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1890,37 +1890,42 @@ static int soc_probe(struct platform_device *pdev) return 0; } -/* removes a socdev */ -static int soc_remove(struct platform_device *pdev) +static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct platform_device *pdev = to_platform_device(card->dev); int i; - if (card->instantiated) { + /* make sure any delayed work runs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + flush_delayed_work_sync(&rtd->delayed_work); + } - /* make sure any delayed work runs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - flush_delayed_work_sync(&rtd->delayed_work); - } + /* remove auxiliary devices */ + for (i = 0; i < card->num_aux_devs; i++) + soc_remove_aux_dev(card, i); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + /* remove and free each DAI */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i); - /* remove and free each DAI */ - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + soc_cleanup_card_debugfs(card); - soc_cleanup_card_debugfs(card); + /* remove the card */ + if (card->remove) + card->remove(pdev); - /* remove the card */ - if (card->remove) - card->remove(pdev); + kfree(card->rtd); + snd_card_free(card->snd_card); + return 0; + +} + +/* removes a socdev */ +static int soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); - kfree(card->rtd); - snd_card_free(card->snd_card); - } snd_soc_unregister_card(card); return 0; } @@ -3153,6 +3158,8 @@ static int snd_soc_register_card(struct snd_soc_card *card) */ static int snd_soc_unregister_card(struct snd_soc_card *card) { + if (card->instantiated) + soc_cleanup_card_resources(card); mutex_lock(&client_mutex); list_del(&card->list); mutex_unlock(&client_mutex); -- 1.7.2.3

3 2

[alsa-devel] [PATCH 3/4] ASoC: soc core add inline to handle card list initialzation
by Koul, Vinod 13 Jan '11

13 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> Currently the soc_probe initializes the card hence it does the card list initialzation. But if machines directly register the card they would need to do these steps, so putting them as inline would save lot of code This patch adds an inline to do list initialzation Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <harsha.priya(a)intel.com> --- include/sound/soc.h | 10 ++++++++++ sound/soc/soc-core.c | 7 +------ 2 files changed, 11 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index accb8a1..541ddfa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -779,6 +779,16 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) return dev_get_drvdata(&rtd->dev); } +static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) +{ + INIT_LIST_HEAD(&card->dai_dev_list); + INIT_LIST_HEAD(&card->codec_dev_list); + INIT_LIST_HEAD(&card->platform_dev_list); + INIT_LIST_HEAD(&card->widgets); + INIT_LIST_HEAD(&card->paths); + INIT_LIST_HEAD(&card->dapm_list); +} + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9c5e7cf..8305712 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1872,12 +1872,7 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - INIT_LIST_HEAD(&card->dai_dev_list); - INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); - INIT_LIST_HEAD(&card->widgets); - INIT_LIST_HEAD(&card->paths); - INIT_LIST_HEAD(&card->dapm_list); + snd_soc_initialize_card_lists(card); ret = snd_soc_register_card(card); if (ret != 0) { -- 1.7.2.3

1 0

[alsa-devel] [PATCH 2/4] ASoC: soc core move the card debugfs initialization
by Koul, Vinod 13 Jan '11

13 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> The card debugfs initialization is done in soc_probe but would be better if it is done when the card in registered This patch moves the debugfs initialization to register_card() Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <harsha.priya(a)intel.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 318c3a7..9c5e7cf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1879,8 +1879,6 @@ static int soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); - soc_init_card_debugfs(card); - ret = snd_soc_register_card(card); if (ret != 0) { dev_err(&pdev->dev, "Failed to register card\n"); @@ -3123,6 +3121,8 @@ static int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + soc_init_card_debugfs(card); + card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * (card->num_links + card->num_aux_devs), GFP_KERNEL); -- 1.7.2.3

1 0

[alsa-devel] [PATCH] ASoC: Blackfin TDM: fix missed snd_soc_dai_get_drvdata update
by Mike Frysinger 13 Jan '11

13 Jan '11

One spot was missed in this driver when converting from snd_soc_dai.private_data to snd_soc_dai_get_drvdata. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-tdm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 1251239..b2cf239 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -210,7 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = dai->private_data; + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); if (!dai->active) return 0; -- 1.7.4.rc1

4 7

[alsa-devel] [PATCH] ASoC: Make register_card API available to drivers
by Koul, Vinod 13 Jan '11

13 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> The register_card is an internal API which soc_probe uses to register the card. In order to create multiple cards we need to create multiple instances of soc-audio device. If we exposes the register card API for drivers to use, then machine driver can register their cards directly This patch makes register_card and unregister_card API available to machine drivers. This also maintains compatibility with older method of registration which can be marked depreciated and removed eventually. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- include/sound/soc.h | 2 + sound/soc/soc-core.c | 90 +++++++++++++++++++++++++++---------------------- 2 files changed, 52 insertions(+), 40 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d609232..b515eb6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -258,6 +258,8 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_register_card(struct snd_soc_card *card); +int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cbac50b..0bd07f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -58,8 +58,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int snd_soc_register_card(struct snd_soc_card *card); -static int snd_soc_unregister_card(struct snd_soc_card *card); static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* @@ -1870,16 +1868,12 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; + /* no card, so machine driver will register explictly */ + if (!card) + return 0; + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - INIT_LIST_HEAD(&card->dai_dev_list); - INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); - INIT_LIST_HEAD(&card->widgets); - INIT_LIST_HEAD(&card->paths); - INIT_LIST_HEAD(&card->dapm_list); - - soc_init_card_debugfs(card); ret = snd_soc_register_card(card); if (ret != 0) { @@ -1890,37 +1884,46 @@ static int soc_probe(struct platform_device *pdev) return 0; } -/* removes a socdev */ -static int soc_remove(struct platform_device *pdev) +static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct platform_device *pdev = to_platform_device(card->dev); int i; - if (card->instantiated) { + /* make sure any delayed work runs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + flush_delayed_work_sync(&rtd->delayed_work); + } - /* make sure any delayed work runs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - flush_delayed_work_sync(&rtd->delayed_work); - } + /* remove auxiliary devices */ + for (i = 0; i < card->num_aux_devs; i++) + soc_remove_aux_dev(card, i); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + /* remove and free each DAI */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i); - /* remove and free each DAI */ - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + soc_cleanup_card_debugfs(card); - soc_cleanup_card_debugfs(card); + /* remove the card */ + if (card->remove) + card->remove(pdev); + + kfree(card->rtd); + snd_card_free(card->snd_card); + return 0; - /* remove the card */ - if (card->remove) - card->remove(pdev); +} + +/* removes a socdev */ +int soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + + if (!card) + return 0; - kfree(card->rtd); - snd_card_free(card->snd_card); - } snd_soc_unregister_card(card); return 0; } @@ -3107,17 +3110,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); * * @card: Card to register * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_register_card(struct snd_soc_card *card) +int snd_soc_register_card(struct snd_soc_card *card) { int i; if (!card->name || !card->dev) return -EINVAL; + INIT_LIST_HEAD(&card->dai_dev_list); + INIT_LIST_HEAD(&card->codec_dev_list); + INIT_LIST_HEAD(&card->platform_dev_list); + INIT_LIST_HEAD(&card->widgets); + INIT_LIST_HEAD(&card->paths); + INIT_LIST_HEAD(&card->dapm_list); + + soc_init_card_debugfs(card); + card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * (card->num_links + card->num_aux_devs), GFP_KERNEL); @@ -3141,25 +3150,26 @@ static int snd_soc_register_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core * * @card: Card to unregister * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_unregister_card(struct snd_soc_card *card) +int snd_soc_unregister_card(struct snd_soc_card *card) { mutex_lock(&client_mutex); list_del(&card->list); mutex_unlock(&client_mutex); + if (card->instantiated) + soc_cleanup_card_resources(card); dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); return 0; } +EXPORT_SYMBOL_GPL(snd_soc_unregister_card); /* * Simplify DAI link configuration by removing ".-1" from device names -- 1.7.2.3

2 8

[alsa-devel] [PATCH] ASoC: Blackfin: fix DAI/SPORT config dependency issues
by Mike Frysinger 13 Jan '11

13 Jan '11

From: Barry Song <barry.song(a)analog.com> While I2S/TDM/AC97 DAI is built-in, others are compiled as modules, SND_BF5XX_SOC_SPORT will be module, then DAI can't get some symbols. Except that, SND_BF5XX_AC97 depends on SND_BF5XX_SOC_AC97 too. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/Kconfig | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 3abeedd..ae40359 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,6 +1,7 @@ config SND_BF5XX_I2S tristate "SoC I2S Audio for the ADI BF5xx chip" depends on BLACKFIN + select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -35,6 +36,7 @@ config SND_BFIN_AD73311_SE config SND_BF5XX_TDM tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" depends on (BLACKFIN && SND_SOC) + select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in TDM @@ -61,6 +63,10 @@ config SND_BF5XX_SOC_AD193X config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_AC97 help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in slot 16 @@ -122,17 +128,12 @@ config SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_I2S tristate - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_TDM tristate - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_AC97 tristate - select AC97_BUS - select SND_SOC_AC97_BUS - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" -- 1.7.4.rc1

3 3

[alsa-devel] [PATCH] ASoC: Blackfin AC97: fix build error after multi-component update
by Mike Frysinger 13 Jan '11

13 Jan '11

We need to tweak how we query the active capture/playback state after the recent overhauls of common code. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ac97.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c5f856e..ffbac26 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -260,9 +260,9 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) return 0; - if (dai->capture.active) + if (dai->capture_active) sport_rx_stop(sport); - if (dai->playback.active) + if (dai->playback_active) sport_tx_stop(sport); return 0; } -- 1.7.4.rc1

3 3

[alsa-devel] [PATCH v2 1/3] ASoC: soc-cache: Add support for default readable()/volatile() functions
by Dimitris Papastamos 13 Jan '11

13 Jan '11

For common scenarios, device drivers can provide a table of all the registers that are at least either readable/writable/volatile. The idea is that if a register lookup fails, all of its read/write/vol members will be zero and will be treated as default. This also reduces the size of the register access array. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 22 ++++++++++++++++++++++ sound/soc/soc-cache.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d609232..b8acf99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,6 +276,10 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -367,6 +371,22 @@ int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** + * struct snd_soc_reg_access - Describes whether a given register is + * readable, writable or volatile. + * + * @reg: the register number + * @read: whether this register is readable + * @write: whether this register is writable + * @vol: whether this register is volatile + */ +struct snd_soc_reg_access { + u16 reg; + u16 read; + u16 write; + u16 vol; +}; + +/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -515,6 +535,8 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; + short reg_access_size; + const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1a36b36..d97a59f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1603,3 +1603,52 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); + +static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int min, max, index; + + codec_drv = codec->driver; + min = 0; + max = codec_drv->reg_access_size - 1; + do { + index = (min + max) / 2; + if (codec_drv->reg_access_default[index].reg == reg) + return index; + if (codec_drv->reg_access_default[index].reg < reg) + min = index + 1; + else + max = index; + } while (min <= max); + return -1; +} + +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].vol; +} +EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); + +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].read; +} +EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); -- 1.7.3.5

3 4

[alsa-devel] [PATCH] ASoC: tegra: s/IS_ERR_OR_NULL/IS_ERR/ for clk_get_sys
by Stephen Warren 13 Jan '11

13 Jan '11

A recent discussion on linux-arm-kernel noted that the value returned by clk_get_sys is an opaque token, and not strictly a pointer; it is meaningful only to the clock API, clients should not dereference the value, and the clock API must accept any non-IS_ERR value it returned. Hence, only IS_ERR is appropriate to interpret the result, not IS_ERR_OR_NULL. I checked that clk_get_sys in both ASoC's for-next and Tegra's for-next do behave as described; NULL is not returned in the case of error. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 20 +++++++++----------- sound/soc/tegra/tegra_i2s.c | 2 +- 2 files changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 711ab7f..cfe2ea8 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -113,35 +113,33 @@ int tegra_asoc_utils_init(void) int ret; clk_pll_a = clk_get_sys(NULL, "pll_a"); - if (IS_ERR_OR_NULL(clk_pll_a)) { + if (IS_ERR(clk_pll_a)) { pr_err(PREFIX "Can't retrieve clk pll_a\n"); ret = PTR_ERR(clk_pll_a); goto err; } clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); - if (IS_ERR_OR_NULL(clk_pll_a_out0)) { + if (IS_ERR(clk_pll_a_out0)) { pr_err(PREFIX "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(clk_pll_a_out0); - goto err; + goto err_put_pll_a; } clk_cdev1 = clk_get_sys(NULL, "cdev1"); - if (IS_ERR_OR_NULL(clk_cdev1)) { + if (IS_ERR(clk_cdev1)) { pr_err(PREFIX "Can't retrieve clk cdev1\n"); ret = PTR_ERR(clk_cdev1); - goto err; + goto err_put_pll_a_out0; } return 0; +err_put_pll_a_out0: + clk_put(clk_pll_a_out0); +err_put_pll_a: + clk_put(clk_pll_a); err: - if (!IS_ERR_OR_NULL(clk_cdev1)) - clk_put(clk_cdev1); - if (!IS_ERR_OR_NULL(clk_pll_a_out0)) - clk_put(clk_pll_a_out0); - if (!IS_ERR_OR_NULL(clk_pll_a)) - clk_put(clk_pll_a); return ret; } diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 9b7a22a..75d3bfe 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -385,7 +385,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id); i2s->clk_i2s = clk_get_sys(clk_name, NULL); - if (IS_ERR_OR_NULL(i2s->clk_i2s)) { + if (IS_ERR(i2s->clk_i2s)) { pr_err("Can't retrieve i2s clock\n"); ret = PTR_ERR(i2s->clk_i2s); goto err_free; -- 1.7.0.4

3 2

[alsa-devel] [RFC] [ALSA] ASoC documentation updates
by Seungwhan Youn 13 Jan '11

13 Jan '11

This patch is only for RFC purpose of ASoC documentation updates which match with current ASoC codes with documents. Mostly modify features are modified to be sync with changes after multi-component patches. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- To Mr. Liam Girdwood and Mark Brown I think this modification is not enough to cover current ASoC features, and I hope that it will be a good time to introduce recently changes, e.g. multi-component model, new linkage system of ASoC DAIs, and so on. But if you have some other idea, or other reason to delay this, please ignore this RFC. And also if there are something that I can help, please let me know. Thanks, Claude --- Documentation/sound/alsa/soc/codec.txt | 45 +++++++++++++--------------- Documentation/sound/alsa/soc/machine.txt | 38 ++++++------------------ Documentation/sound/alsa/soc/platform.txt | 12 ++++++- 3 files changed, 40 insertions(+), 55 deletions(-) diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 37ba3a7..bce23a4 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -27,42 +27,38 @@ ASoC Codec driver breakdown 1 - Codec DAI and PCM configuration ----------------------------------- -Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +Each codec driver must have a struct snd_soc_dai_driver to define its DAI and PCM capabilities and operations. This struct is exported so that it can be registered with the core by your machine driver. e.g. -struct snd_soc_codec_dai wm8731_dai = { - .name = "WM8731", - /* playback capabilities */ +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + +struct snd_soc_dai_driver wm8731_dai = { + .name = "wm8731-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - /* capture capabilities */ .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - /* pcm operations - see section 4 below */ - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - }, - /* DAI operations - see DAI.txt */ - .dai_ops = { - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, + .symmetric_rates = 1, }; -EXPORT_SYMBOL_GPL(wm8731_dai); 2 - Codec control IO @@ -186,13 +182,14 @@ when the mute is applied or freed. i.e. -static int wm8974_mute(struct snd_soc_codec *codec, - struct snd_soc_codec_dai *dai, int mute) +static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; - if(mute) - wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + + if (mute) + snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); else - wm8974_write(codec, WM8974_DAC, mute_reg); + snd_soc_write(codec, WM8974_DAC, mute_reg); return 0; } diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 2524c75..3e2ec9c 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -12,6 +12,8 @@ the following struct:- struct snd_soc_card { char *name; + ... + int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -22,12 +24,13 @@ struct snd_soc_card { int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev); - /* machine stream operations */ - struct snd_soc_ops *ops; + ... /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + + ... }; probe()/remove() @@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs and DMA is suspended and resumed. Optional. -Machine operations ------------------- -The machine specific audio operations can be set here. Again this is optional. - - Machine DAI Configuration ------------------------- The machine DAI configuration glues all the codec and CPU DAIs together. It can @@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai = &pxa_i2s_dai, - .codec_dai = &wm8731_dai, + .cpu_dai_name = "pxa-is2-dai", + .codec_dai_name = "wm8731-hifi", + .platform_name = "pxa-pcm-audio", + .codec_name = "wm8713-codec.0-001a", .init = corgi_wm8731_init, .ops = &corgi_ops, }; @@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = { }; -Machine Audio Subsystem ------------------------ - -The machine soc device glues the platform, machine and codec driver together. -Private data can also be set here. e.g. - -/* corgi audio private data */ -static struct wm8731_setup_data corgi_wm8731_setup = { - .i2c_address = 0x1b, -}; - -/* corgi audio subsystem */ -static struct snd_soc_device corgi_snd_devdata = { - .machine = &snd_soc_corgi, - .platform = &pxa2xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &corgi_wm8731_setup, -}; - - Machine Power Map ----------------- diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index 06d8359..d57efad 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -20,9 +20,10 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -The platform driver exports its DMA functionality via struct snd_soc_platform:- +The platform driver exports its DMA functionality via struct +snd_soc_platform_driver:- -struct snd_soc_platform { +struct snd_soc_platform_driver { char *name; int (*probe)(struct platform_device *pdev); @@ -34,6 +35,13 @@ struct snd_soc_platform { int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); + /* + * For platform caused delay reporting. + * Optional. + */ + snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, + struct snd_soc_dai *); + /* platform stream ops */ struct snd_pcm_ops *pcm_ops; }; -- 1.6.2.5

3 2

[alsa-devel] [PATCH] ASoC: Blackfin TDM: use external frame syncs
by Mike Frysinger 13 Jan '11

13 Jan '11

From: Barry Song <barry.song(a)analog.com> We don't want to use internal frame syncs otherwise we sometimes get out of sync, so don't enable them when setting up the SPORT. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-tdm.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index b2cf239..5515ac9 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -235,13 +235,13 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) ret = -EBUSY; } - ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0); + ret = sport_config_rx(sport, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; } - ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0); + ret = sport_config_tx(sport, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; @@ -303,14 +303,14 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) goto sport_config_err; } - ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0); + ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; goto sport_config_err; } - ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0); + ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- 1.7.4.rc1

3 6

[alsa-devel] [PATCH 1/3] ASoC: soc-cache: Add support for default readable()/volatile() functions
by Dimitris Papastamos 13 Jan '11

13 Jan '11

For common scenarios, device drivers can provide a table of all the registers that are at least either readable/writable/volatile. The idea is that if a register lookup fails, all of its read/write/vol members will be zero and will be treated as default. This also reduces the size of the register access array. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 22 ++++++++++++++++++++++ sound/soc/soc-cache.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d609232..b8acf99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,6 +276,10 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -367,6 +371,22 @@ int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** + * struct snd_soc_reg_access - Describes whether a given register is + * readable, writable or volatile. + * + * @reg: the register number + * @read: whether this register is readable + * @write: whether this register is writable + * @vol: whether this register is volatile + */ +struct snd_soc_reg_access { + u16 reg; + u16 read; + u16 write; + u16 vol; +}; + +/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -515,6 +535,8 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; + short reg_access_size; + const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1a36b36..d97a59f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1603,3 +1603,52 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); + +static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int min, max, index; + + codec_drv = codec->driver; + min = 0; + max = codec_drv->reg_access_size - 1; + do { + index = (min + max) / 2; + if (codec_drv->reg_access_default[index].reg == reg) + return index; + if (codec_drv->reg_access_default[index].reg < reg) + min = index + 1; + else + max = index; + } while (min <= max); + return -1; +} + +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].vol; +} +EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); + +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].read; +} +EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); -- 1.7.3.5

1 2

[alsa-devel] registering multiple cards
by Koul, Vinod 13 Jan '11

13 Jan '11

Hi, Currently snd_soc_register_card is internal soc-core API and is called from soc-probe. And probe always assumes it is a single device, so how do we add up two cards? I saw a note on this that it will exposed so that drivers can register. What is that plan to do that, and meanwhile what can be done to solve this? ~Vinod

2 2

[alsa-devel] [PATCH] powerpc/85xx: fix compatible properties of the P1022DS DMA nodes used for audio
by Timur Tabi 13 Jan '11

13 Jan '11

In order to prevent the fsl_dma driver from claiming the DMA channels that the P1022DS audio driver needs, the compatible properties for those nodes must say "fsl,ssi-dma-channel" instead of "fsl,eloplus-dma-channel". Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, please pick up this patch for 2.6.37. arch/powerpc/boot/dts/p1022ds.dts | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/powerpc/boot/dts/p1022ds.dts b/arch/powerpc/boot/dts/p1022ds.dts index 2bbecbb..69422eb 100644 --- a/arch/powerpc/boot/dts/p1022ds.dts +++ b/arch/powerpc/boot/dts/p1022ds.dts @@ -291,13 +291,13 @@ ranges = <0x0 0xc100 0x200>; cell-index = <1>; dma00: dma-channel@0 { - compatible = "fsl,eloplus-dma-channel"; + compatible = "fsl,ssi-dma-channel"; reg = <0x0 0x80>; cell-index = <0>; interrupts = <76 2>; }; dma01: dma-channel@80 { - compatible = "fsl,eloplus-dma-channel"; + compatible = "fsl,ssi-dma-channel"; reg = <0x80 0x80>; cell-index = <1>; interrupts = <77 2>; -- 1.7.2.3

2 1

[alsa-devel] [PATCH 0/2] ASoC: sh: fsi-codecs: add/modify FSI selection
by Kuninori Morimoto 13 Jan '11

13 Jan '11

Dear Mark, Liam Kuninori Morimoto (2): ASoC: sh: fsi-ak4642: fixup platform device id ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection 1st patch fixup platform device id. 2nd patch add FSI PortA/B selection to FSI-HDMI These patches are tested on Ecovec, AP4EVB, Mackerel board, and Paul's rmobile-latest + Mark's for-2.6.38 branch. Best regards -- Kuninori Morimoto

3 4

[alsa-devel] [PATCH] ALSA: hda: Disable 4/6 channels on some NVIDIA GPUs.
by Nitin Daga 12 Jan '11

12 Jan '11

Added hardware constraint in patch_hdmi.c to disable channels 4/6 which are not supported by some older NVIDIA GPUs. Signed-off-by: Nitin Daga <ndaga(a)nvidia.com> Acked-By: Stephen Warren <swarren(a)nvidia.com> --- pci/hda/patch_hdmi.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 42 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_hdmi.c b/pci/hda/patch_hdmi.c index d3e49aa..ddde59b 100644 --- a/pci/hda/patch_hdmi.c +++ b/pci/hda/patch_hdmi.c @@ -1166,11 +1166,53 @@ static int nvhdmi_7x_init(struct hda_codec *codec) return 0; } +static unsigned int channels_2_6_8[] = { + 2, 6, 8 +}; + +static unsigned int channels_2_8[] = { + 2, 8 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_6_8_channels = { + .count = ARRAY_SIZE(channels_2_6_8), + .list = channels_2_6_8, + .mask = 0, +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_8_channels = { + .count = ARRAY_SIZE(channels_2_8), + .list = channels_2_8, + .mask = 0, +}; + static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; + struct snd_pcm_hw_constraint_list *hw_constraints_channels = NULL; + + switch (codec->preset->id) { + case 0x10de0002: + case 0x10de0003: + case 0x10de0005: + case 0x10de0006: + hw_constraints_channels = &hw_constraints_2_8_channels; + break; + case 0x10de0007: + hw_constraints_channels = &hw_constraints_2_6_8_channels; + break; + default: + break; + } + + if (hw_constraints_channels != NULL) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + hw_constraints_channels); + } + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -- 1.7.0.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

4 7

[alsa-devel] Adding a "capture" device naming scheme
by Colin Guthrie 12 Jan '11

12 Jan '11

Hi, As has been pointed out a few times, it's incorrect to use "front:" for capture, but can we have an official name/wrapper (I'm unsure of the terminology here) for this so we don't use hw: for recording? I'd be in favour of "capture:" or "input:" as these both seem quite logical and easy to understand. It would be used thusly: arecord -f S16_LE -c 2 -D capture:CARD=Intel as opposed to the incorrect: arecord -f S16_LE -c 2 -D front:CARD=Intel or the low level: arecord -f S16_LE -c 2 -D hw:CARD=Intel Keeping it wrapped up allows for e.g. softvol or other things to be plugged in as appropriate in the config and thus future proofs things a little. If someone could point me roughly in the right direction I'll make a patch. I suspect it'll be easy enough to cook up the relevant config needed (and I suspect it's just config rather than code that needs patched, but I could be wrong). It would be nice to get this into a release (alsa-lib?) soon so that it can propigate and we can require it. Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

4 10

[alsa-devel] [PATCH] ASoC: soc-cache: Add reg_size as a member to snd_soc_codec
by Dimitris Papastamos 12 Jan '11

12 Jan '11

Simplify the use of reg_size, by calculating it once and storing it in the codec structure for later reference. The value of reg_size is reg_cache_size * reg_word_size. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 26 ++++++++------------------ sound/soc/soc-core.c | 1 + 3 files changed, 10 insertions(+), 18 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c477058..d609232 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -459,6 +459,7 @@ struct snd_soc_codec { struct list_head card_list; int num_dai; enum snd_soc_compress_type compress_type; + size_t reg_size; /* reg_cache_size * reg_word_size */ /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 2f1d596..1a36b36 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1100,34 +1100,28 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, unsigned int reg) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; return (reg * codec_drv->reg_word_size) / - DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); + DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, unsigned int reg) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - return reg % (DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()) / + return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) / codec_drv->reg_word_size); } static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - return DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); + return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) @@ -1287,7 +1281,7 @@ static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; - size_t reg_size, bmp_size; + size_t bmp_size; const struct snd_soc_codec_driver *codec_drv; int ret, tofree, i, blksize, blkcount; const char *p, *end; @@ -1295,7 +1289,6 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) ret = 0; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; /* * If we have not been given a default register cache @@ -1307,8 +1300,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) tofree = 1; if (!codec->reg_def_copy) { - codec->reg_def_copy = kzalloc(reg_size, - GFP_KERNEL); + codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL); if (!codec->reg_def_copy) return -ENOMEM; } @@ -1356,7 +1348,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) blksize = snd_soc_lzo_get_blksize(codec); p = codec->reg_def_copy; - end = codec->reg_def_copy + reg_size; + end = codec->reg_def_copy + codec->reg_size; /* compress the register map and fill the lzo blocks */ for (i = 0; i < blkcount; ++i, p += blksize) { lzo_blocks[i]->src = p; @@ -1441,16 +1433,14 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (codec->reg_def_copy) codec->reg_cache = kmemdup(codec->reg_def_copy, - reg_size, GFP_KERNEL); + codec->reg_size, GFP_KERNEL); else - codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); + codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL); if (!codec->reg_cache) return -ENOMEM; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 04475c1..cbac50b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3500,6 +3500,7 @@ int snd_soc_register_codec(struct device *dev, /* allocate CODEC register cache */ if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + codec->reg_size = reg_size; /* it is necessary to make a copy of the default register cache * because in the case of using a compression type that requires * the default register cache to be marked as __devinitconst the -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache
by Dimitris Papastamos 12 Jan '11

12 Jan '11

With the addition of the cache fallback functionality, it is necessary to ensure that if the register defaults cache was marked as __devinitconst and the LZO compression is not compiled in the kernel, the fallback to flat compression will still use a copy of the defaults cache. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 16 ++++------------ 1 files changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index b0d86cc..2f1d596 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1400,8 +1400,8 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; - if (codec_drv->reg_cache_default) - if (snd_soc_get_cache_val(codec_drv->reg_cache_default, + if (codec->reg_def_copy) + if (snd_soc_get_cache_val(codec->reg_def_copy, i, codec_drv->reg_word_size) == val) continue; ret = snd_soc_write(codec, i, val); @@ -1446,16 +1446,8 @@ static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) codec_drv = codec->driver; reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - /* - * for flat compression, we don't need to keep a copy of the - * original defaults register cache as it will definitely not - * be marked as __devinitconst - */ - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - - if (codec_drv->reg_cache_default) - codec->reg_cache = kmemdup(codec_drv->reg_cache_default, + if (codec->reg_def_copy) + codec->reg_cache = kmemdup(codec->reg_def_copy, reg_size, GFP_KERNEL); else codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); -- 1.7.3.5

3 3

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 12 Jan '11

12 Jan '11

The following changes since commit 2f55eaab56ea083498034c18e09d356594081fca: Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-01-12 07:20:30 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Dimitris Papastamos (1): ASoC: soc-cache: Fix invalid memory access during snd_soc_lzo_cache_sync() Guennadi Liakhovetski (1): ASoC: fix migor audio build Timur Tabi (1): ASoC: cs4270: use the built-in register cache support Vasily Khoruzhick (1): ASoC: RX1950: Enable Mic Jack during glue driver init sound/soc/codecs/cs4270.c | 161 +++++++++++------------------------- sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/sh/migor.c | 2 +- sound/soc/soc-cache.c | 2 +- 4 files changed, 51 insertions(+), 115 deletions(-)

2 1

[alsa-devel] Fix section mismatch in wm8995.c
by Takashi Iwai 12 Jan '11

12 Jan '11

FYI, I noticed the section mismatch in wm8995.c, and fixed now on topic/asoc branch with the patch below. thanks, Takashi === From 03cfbdf9f7a1d392146718f67e50fa9ab2844f22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Tue, 11 Jan 2011 17:58:26 +0100 Subject: [PATCH] ASoC: Fix section mismatch in wm8995.c __devinitconst can't be used for data referred in driver struct. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm8995.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 3d2110c..6045cbd 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -30,7 +30,7 @@ #include "wm8995.h" -static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] __devinitconst = { +static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = { [0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b, [24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0, [28] = 0x000f, [32] = 0x0005, [33] = 0x0005, [40] = 0x0003, -- 1.7.3.4

3 14

[alsa-devel] Missing patches for 2.6.38 merge?
by Takashi Iwai 12 Jan '11

12 Jan '11

Hi, I've been slowly recovered from the long off, and gathering the patches for pushing to 2.6.38 merge. I'm going to send a pull request tomorrow or the next day. The below is the list of commits for now. If anyone has a patch to be merged but missing here, please let me know (at best resend to me). Mark, Liam, do you guys have still many pending updates? If so, please forward quickly. Also, if a brief summary of this merge in ASoC area is ready, it'd be helpful. thanks, Takashi === Alexander Sverdlin (1): ASoC: EP93xx: sampling rate range extended Andreas Mohr (7): ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer ALSA: azt3328: cosmetics, minor updates ALSA: azt3328: _setfmt() update ALSA: azt3328: cosmetics: use a helper variable for codec setup ALSA: azt3328: use a helper variable to remove one indirection in hotpath ALSA: azt3328: use proper private_data hookup for codec identification ALSA: azt3328: improve snd_azf3328_codec_setdmaa() Anisse Astier (1): ALSA: hda - factorize an automute_mic realtek quirk function Anssi Hannula (1): ALSA: hda - use generic hdmi parser for ATI R6xx codec Arnaud Patard (3): ASoC: add support for alc562[123] codecs ASoC: kirkwood: Add audio support to hp t5325 thin clients ASoC: Add support for OpenRD Ultimate Axel Lin (21): ASoC: Fix compile error if CONFIG_DEBUG_FS is not configured ASoC: Fix incorrect kfree in ad1836_probe error path ASoC: Fix incorrect kfree in ad193x_probe error path ASoC: Fix incorrect kfree in aic3x_probe error path ASoC: Fix incorrect kfree in wm8731_probe error path ASoC: Fix incorrect kfree in wm8962_probe error path ASoC: Move kfree(twl4030) to twl4030_soc_remove() ASoC: Fix a memory leak in alc5623_i2c_probe error path ASoC: neo1973_gta02_wm8753: fix wrong parameter for snd_soc_register_dai and snd_soc_unregister_dai ASoC: Fix compile error for smartq_wm8987.c ASoC: smdk_wm9713: fix resource leak in smdk_init error path ASoC: Fix resource leak in goni_wm8994.c ASoC: Fix resource leak in neo1973_gta02_init() error path ASoC: Add missing gpio_free(S3C64XX_GPK(12)) in smartq_exit() ASoC: Fix resource leak in smdk_spdif.c ASoC: Simplify pm860x_probe error handling ASoC: ak4535: Improve readability for setting mute ASoC: Remove unused aic3x_i2c_init and aic3x_i2c_exit functions ASoC: Fix inconsistent meaning of default case while checking alc5623->id ASoC: Fix snd_soc_instantiate_card error path ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove() Brian Bloniarz (1): ALSA: ice1712 - working M-Audio Delta 66E support Clemens Ladisch (46): ALSA: oxygen: merge HiFier driver into snd-oxygen ALSA: oxygen: add Kuroutoshikou CMI8787-HG2PCI support ALSA: oxygen: reorganize PCI IDs ALSA: oxygen: add HiFier Serenade support ALSA: pcm: support for period wakeup disabling ALSA: hda-intel: support for period wakeup disabling ALSA: oxygen: support for period wakeup disabling ALSA: pcm: detect xruns in no-period-wakeup mode ALSA: pcm: optimize xrun detection in no-period-wakeup mode ALSA: timer: automatically load the high-resolution timer ALSA: virtuoso: initialize unknown GPIO bits ALSA: virtuoso: add HDMI enable switch for HDAV1.3 ALSA: virtuoso: fix front panel routing for D1/DX/ST(X) ALSA: oxygen: allow to dump codec registers ALSA: oxygen: show correct package ID ALSA: oxygen: update hardware comments ALSA: oxygen: allow different number of PCM and mixer channels ALSA: oxygen: remove oxygen_model::private_data field ALSA: oxygen: fix SPI clocks slower than 6.25 MHz ALSA: virtuoso: do not use fast I2C speed ALSA: virtuoso: wait for PCM1796 clock to become stable ALSA: virtuoso: change PCM1796 format to I2S ALSA: virtuoso: remove non-working controls on Essence ST Deluxe ALSA: virtuoso: configure correct master clock frequency on the CS2000 ALSA: virtuoso: use lower master clock with H6 daughterboard ALSA: virtuoso: handle DAC oversampling automatically ALSA: virtuoso: use headphone gain setting only on front DAC ALSA: oxygen: simplify model-specific MCLK handling ALSA: virtuoso: reduce MCLK in double rate modes ALSA: oxygen: add more PCI IDs ALSA: oxygen: add X-Meridian 2G support ALSA: oxygen: add Xonar DG support ALSA: virtuoso: add Xonar HDAV1.3 Slim support ALSA: core, oxygen, virtuoso: add an enum control info helper ALSA: bt87x: use enum control info helper ALSA: cmipci: use enum control info helper ALSA: ymfpci: use enum control info helper ALSA: usb-audio: use enum control info helper ALSA: usb-audio: add Edirol SD-90 PCM support MAINTAINERS: add myself ALSA: oxygen: add digital input validity check switch ALSA: oxygen: X-Meridian: add S/PDIF source selection ALSA: oxygen: do not show chip revision in card longname ALSA: oxygen: add some card names ALSA: oxygen: fix CD/MIDI for X-Meridian (2G) ALSA: oxygen: add S/PDIF source selection for Claro cards Daniel T Chen (3): ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J ALSA: hda: Use vostro model quirk for Dell Vostro 1014 David Henningsson (9): ALSA: HDA: Create mixers on ALC887 ALSA: HDA: Remove unconnected PCM devices for Intel HDMI ALSA: HDA: Rename "Int Mic" to "Internal Mic" ALSA: HDA: Rename "Ext Mic" and "External Mic" to "Mic" ALSA: HDA: Rename "e-Mic" and "i-Mic" to "Mic" and "Internal Mic" ALSA: HDA: Add internal mic for IDT 92HD88B ALSA: HDA: Rename "Mic Boost" to "Mic Boost Volume" ALSA: HDA: Fix volume control indices for Mics (Realtek) ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx Dimitris Papastamos (22): ASoC: soc-cache: Remove unnecessary debugging info ASoC: WM8770: Initial driver ASoC: soc-cache: Use BUG_ON() for unsupported hw_read() calls ASoC: Remove unneeded use of address-of operator ASoC: soc-cache: Add support for flat register caching ASoC: soc-cache: Add support for LZO register caching ASoC: soc-cache: Add support for rbtree based register caching ASoC: soc-cache: Add error checking in the *_cache_sync functions ASoC: soc-cache: Ensure consistent cache naming ASoC: soc-cache: Fix memory overflow in LZO initialization ASoC: Add compress_type as a member to snd_soc_codec ASoC: Change the base value of compress_type ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default ASoC: soc-core: Allow machine drivers to override compress_type ASoC: soc-cache: Add optional cache name member to snd_soc_cache_ops ASoC: soc-core: Replace use of strncpy() with strlcpy() ASoC: soc-core: Remove useless inline function construct ASoC: soc-core: Fix null pointer dereference ASoC: soc-cache: A few minor stylistic changes ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type ASoC: Add initial WM8995 driver Florian Faber (1): ALSA: hdsp - Add support for RPM io box Francois Mazard (1): ASoC: twl6040: Fix analog Mic L & R mux controls Harsha Priya (1): ASoC: Fix the device references to codec and platform drivers Jarkko Nikula (23): ASoC: Fix SND_SOC_ALL_CODECS typo for alc5623 ASoC: tlv320aic3x: Use gpio_is_valid in checking for valid gpio_reset ASoC: tlv320aic3x: Add support to shared common reset line ASoC: Add sound card directory under debugfs/asoc/ ASoC: Move codec debugfs directories under parent card directory ASoC: Move pop time from DAPM context to sound card ASoC: tpa6130a2: Revisit power-up sequence ASoC: Update DAPM debug and error prints ASoC: Convert pop_dbg to use dev_info ASoC: Add optional name_prefix for codec kcontrol, widget and route names ASoC: Fix compile breakage in jz4740.c and smartq_wm8987.c ASoC: Do not include soc-dapm.h ASoC: Do not include soc-dai.h ASoC: Remove cyclic dependency between soc.h and soc-dapm.h/soc-dai.h ASoC: Add support for optional auxiliary dailess codecs ASoC: Fix build failure in soc-core.c ASoC: Don't oops in soc_probe_aux_dev in case of missing codec ASoC: Merge common code in DAI link and auxiliary codec probing/removal ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38 ASoC: Move DAPM paths from DAPM context to snd_soc_card ASoC: Move widgets from DAPM context to snd_soc_card ASoC: Extend DAPM to handle power changes on cross-device paths ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers Jassi Brar (38): ARM: S5PV310: Add platform devices for PL330 DMACs ARM: SAMSUNG: Rename s3c64xx I2S platform device ARM: SAMSUNG: Prepare audio header for new I2S driver ARM: S3C64XX: Upgrade platform device for I2S ARM: S5P64X0: Upgrade platform device for I2S ARM: S5P6442: Upgrade platform device for I2S ARM: S5PC100: Upgrade platform device for I2S ARM: S5PV210: Upgrade platform device for I2S ARM: S5PV310: Add audio platform devices ASoC: Samsung: Remove redundant AQUILA driver ASoC: Samsung: Rename DMA device ARM: Samsung: Define common audio-dma device ASoC: Samsung: Rename ASoC DMA driver ASoC: Samsung: Rename AC97 platform device ASoC: Samsung: Rename AC97 driver ASoC: Samsung: Rename PCM driver ASoC: Samsung: Generalize DMA driver namespace ASoC: Samsung: Rename s3c64xx I2S device ASoC: Samsung: Add common I2S driver ASoC: SMARTQ: Move to use new I2S driver ASoC: GONI: Move to use new I2S driver ASoC: SMDK64XX: Move to use new I2S driver ASoC: S3C64XX: Remove obsoleted I2S drivers ASoC: SMDK64XX: Rename for other platforms ASoC: SMDK_WM8580: Enable for SMDKC100 ASoC: Samsung: Generalize Kconfig symbols ASoC: Samsung: Rename from s3c24xx to samsung ASoC: WM8580: Debug interface index ASoC: Samsung: Set default rclk source rate ASoC: Samsung: I2S: Flush FIFO after stop ASoC: SMDK_WM8580: Make I2S0 as default dai ASoC: SMDK6440: Enable I2S device ASoC: SMDK6450: Enable I2S device ASoC: SMDK6442: Enable I2S device ASoC: SMDKV210: Enable I2S device ASoC: SMDKC110: Enable I2S device ASoC: SMDKV310: Enable AC97 device ASoC: SMDKV310: Add I2S support Jeffrin Jose (1): sound: Fixed line limit issue in sound/ac97_bus.c Jesper Juhl (1): ALSA: Don't leak in sound/core/oss/pcm_oss.c::snd_pcm_hw_param_near() Joe Perches (2): ASoC: Remove unnecessary semicolons ALSA: ml403-ac97cr: Use vsprintf extension %pR for struct resource Jorge Eduardo Candelaria (6): ASoC: sdp4430: Enable FM stereo pins ASoC: twl6040: Add jack support for headset and handset ASoC: sdp4430: Add Jack support ASoC: twl6040: Add headset and handset mux controls ASoC: twl6040: Clear interrupt status at boot time ASoC: twl6040: Increase timeout for power up Karsten Wiese (2): ALSA: snd-usb-us122l: Fix MIDI output ALSA: snd-usb-us122l: Fix missing NULL checks Kay Sievers (1): ALSA: support module on-demand loading for seq and timer Kuninori Morimoto (13): ASoC: sh: fsi-ak4642: Add FSI port and ak464x selection ASoC: sh: fsi-ak4642: tidyup unnecessary variables ASoC: sh: fsi-ak4642: midify card name ASoC: sh: fsi: clean up SPDIF defines ASoC: sh: fsi: remove fsi_master_write ASoC: sh: fsi: change fsi->mst_ctrl to master->a/b_mclk ASoC: sh: fsi: remove runtime register check from fsi_reg_xxx ASoC: sh: fsi: remove runtime register check from fsi_master_xxx ASoC: sh: fsi: move fsi_irq_enable function to fsi_dai_trigger ASoC: sh: fsi: Add over/under run counter ASoC: sh: fsi: modify improper dependent fbdev: sh_mobile_hdmi: fixup compile error ASoC: sh: fsi-da7210: remove unnecessary format settings Lars-Peter Clausen (1): ASoC: codecs: Remove unused reg_cache fields from device structs Liam Girdwood (1): ASoC: Decouple DAPM from CODECs Margarita Olaya Cabrera (1): ASoC: twl6040: Add ramp up/down volume for HS and HF Mario Becroft (1): ASoC: Optimise WM9081 FLL performance Mark Brown (69): ASoC: Store DC offset correction for wm_hubs devices in class W mode ASoC: Convert soc-jack code to use request_any_context_irq() ASoC: Push snd_soc_write() and snd_soc_read() into the source file ASoC: Remove register write trace from WM8994 ASoC: Use pm_wakeup_event() in WM8962 jack detection ASoC: Factor out boiler plate for DAPM event generation ASoC: Add trace events for ASoC register read/write ASoC: Add DAPM trace events ASoC: Reset WM8962 with default ID value ASoC: Prevent system suspend while debouncing wakeup capable GPIO jacks ASoC: Use DAPM context rather than CODEC when constructing sequences ASoC: Provide ADC left/right channel source selection on WM8994 ASoC: Move WM8994 read/write access data into separate file ASoC: Add initial WM8737 driver ASoC: Make SMDK WM8580 driver unloadable ASoC: Update MAINTAINERS for Samsung driver move ASoC: Add WM8737 ALC support ARM: Samsung: Make sure ASoC DMA device is prototyped ASoC: Remove redundant hw_write initialisation in WM8737 mfd: Add initial WM8958 support gpiolib: Add WM8958 support to the WM8994 driver ASoC: Use DC servo startup mode when not doing DCS correction ASoC: Initial WM8958 audio configuration ASoC: Implement support for enhanced AIF3 on WM8958 ASoC: Add WM8958 Multi-band compressor support ASoC: Add WM8958 microphone detection support ASoC: Convert WM8994 to use soc-cache.c cache functions ASoC: Enable rbtree compression for WM8994/58 register cache ASoC: Implement runtime PM for WM8994/58 ASoC: Correct event flags for WM8958 AIF DACs ASoC: Allow user-specified WM8958 multiband compressor configurations ASoC: Provide WM8731 microphone boost TLV information ASoC: Split WM8731 enumeration array into individual enums ASoC: Automatically manage WM8731 deemphasis ASoC: Tune performance of WM8958 revision A ASoC: Apostrophe patrol in soc-core.c ASoC: Make the DAI ops constant in the DAI structure ASoC: Move active copy of CODEC read and write into runtime structure ASoC: Constify struct snd_soc_codec_driver ASoC: When disabling WM8994 FLL force a source selection ASoC: Add post-CODEC bias level callback for machine driver ASoC: Add trace events for jack detection ASoC: Support WM8994 mono AIF configurations ASoC: Implement WM8994/58 DAC and ADC oversampling control ASoC: Implement WM8903 high pass filter support ASoC: Implement WM8903 oversampling rate controls ASoC: Remove open coded symmetry implementation from WM8903 ASoC: Automatically manage WM8903 deemphasis rate ASoC: Fix widgets for WM8994/58 AIF2 source control ASoC: Fix AC'97 registration unwind regulator: Update LDO2 for WM8958 ASoC: Explicitly clear WM8993 ramp controls on power down ASoC: Remove unused DAPM_DOUBLE control types ASoC: Do DAPM control updates in the middle of DAPM sequences ASoC: Remove some unused defines from WM8903 ASoC: Fix WM8994/58 3D stereo control definitions ASoC: If we can't find a cache compression type default to flat ASoC: Make LZO cache compression optional ASoC: Use delayed work to debounce WM8350 jack IRQs ASoC: Trace Wolfson jack detection IRQs ASoC: Add jack IRQ trace to 88pm860x driver ASoC: ifdef out trace points from modules for x86 ASoC: Remove incorrect WM8903 erratum workaround ASoC: Remove WM8995 write sequencer bitfield definitions ASoC: Fix typo in x86 workaround ASoC: Fix double comment start ASoC: One more x86 typo fix ASoC: Yet more x86 tracepoint workarounds ASoC: Change Samsung Kconfig from ASOC_ to SND_SOC_ Maurus Cuelenaere (1): ASoC: Fix SmartQ compilation Misael Lopez Cruz (2): ASoC: twl6040: Enable plug detection interrupts ASoC: twl6040: Use correct offset for LineInAmp Right Nitin Daga (1): ALSA: hda: Disable 4/6 channels on some NVIDIA GPUs. Olaya, Margarita (9): ASoC: dapm: Add output driver widget ASoC: twl6040: Update twl IO macro ASoC: twl6040: Modify the IRQ handler ASoC: twl6040: Restore bias level at resume ASoC: twl6040: Fix PCM error handling ops ASoC: twl6040: Support other sample rates. ASoC: twl6040: Enable automatic power for phoenix 1.1 ASoC: twl6040: Fix TLV dB step values for gains ASoC: twl6040: Set default gains to minimun value Peter Ujfalusi (13): ASoC: TWL4030: Use usleep_range when appropriate ASoC: tlv320dac33: Do not enable the codec in init_chip ASoC: tlv320dac33: Avoid multiple soft power up ASoC: tpa6130a2: Simplify power state management ASoC: tpa6130a2: Defer SW enable from power enable ASoC: tpa6130a2: Use one event handler for PGA_E ASoC: tpa6130a2: Add stereo DAPM path ASoC: tpa6130a2: Replace DAPM code with direct interface ASoC: tlv320dac33: Rename outpup amplifier widget ASoC: tlv320dac33: Move DAC LR power on to a supply widget ASoC: tlv320dac33: Fix compillation error ASoC: tlv320dac33: Power down digital parts, when not needed ASoC: TWL4030: Fix 24bit support Sascha Hauer (1): ASoC: i.MX: we can do mono Seungwhan Youn (4): ASoC: SAMSUNG: Fix initial return value ASoC: SAMSUNG: Remove duplicated snd_card on smdk_spdif ASoC: Remove unnecessary structure definitions ASoC: SAMSUNG: Debug wrong parameter Stephen Rothwell (1): ALSA: include delay.h for msleep in Xonar DG support Takashi Iwai (14): ALSA: hda - Add a generic fixup callback for Realtek codecs ALSA: hda - Remove dead md2 quirk code ALSA: hda - Clean up cxt5066 port-D handling & co ALSA: hda - Enable beep for IDT92HD87 / 88 codecs ALSA: hda - Fix beep-tone on IDT 92HD87/88 codecs ALSA: hda - Mute speakers when line-out jack is plugged with Conexant auto mode ASoC: Fix merge errors with flush_scheduled_work() removal ALSA: hda - Clean up dead code in patch_realtek.c ASoC: don't pass the string as the format arguemtn for dev_info() ALSA: hda - Fix multi-headphone handling for Realtek codecs ALSA: hda - Add support for multiple headphone/speaker controls for Realtek ASoC: Fix section mismatch in wm8995.c ALSA: hda - Don't refer ELD when unplugged ALSA: hda - Add static_hdmi_pcm option to HDMI codec parser Tejun Heo (1): sound: don't use flush_scheduled_work() Todd Broch (2): ALSA: hda: Add modelname lookup and fixup for realtek codecs ALSA: hda: Add fixup for mario system Uk Kim (1): ASoC: Add ADC high pass filter support to WM8994 Vasiliy Kulikov (1): ALSA: snd-atmel-abdac: test wrong variable Vasily Khoruzhick (1): ASoC: Add HP iPAQ H1940 support --- Documentation/sound/alsa/ALSA-Configuration.txt | 20 +- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - MAINTAINERS | 45 +- arch/arm/mach-kirkwood/openrd-setup.c | 2 +- arch/arm/mach-s3c2410/mach-h1940.c | 2 +- arch/arm/mach-s3c2440/mach-rx1950.c | 2 +- arch/arm/mach-s3c64xx/clock.c | 2 +- arch/arm/mach-s3c64xx/dev-audio.c | 75 +- arch/arm/mach-s3c64xx/mach-smdk6410.c | 2 +- arch/arm/mach-s5p6442/dev-audio.c | 44 +- arch/arm/mach-s5p64x0/clock-s5p6440.c | 2 +- arch/arm/mach-s5p64x0/clock-s5p6450.c | 2 +- arch/arm/mach-s5p64x0/dev-audio.c | 55 +- arch/arm/mach-s5pc100/dev-audio.c | 56 +- arch/arm/mach-s5pc100/mach-smdkc100.c | 2 + arch/arm/mach-s5pv210/clock.c | 10 +- arch/arm/mach-s5pv210/dev-audio.c | 58 +- arch/arm/mach-s5pv310/Kconfig | 1 + arch/arm/mach-s5pv310/Makefile | 3 +- arch/arm/mach-s5pv310/dev-audio.c | 364 ++ arch/arm/mach-s5pv310/dma.c | 168 + arch/arm/mach-s5pv310/include/mach/dma.h | 26 + arch/arm/mach-s5pv310/include/mach/irqs.h | 3 + arch/arm/mach-s5pv310/include/mach/map.h | 21 + arch/arm/mach-shmobile/board-ap4evb.c | 5 + arch/arm/plat-s3c24xx/devs.c | 19 +- arch/arm/plat-samsung/Makefile | 1 + arch/arm/plat-samsung/dev-asocdma.c | 25 + arch/arm/plat-samsung/include/plat/audio.h | 24 + arch/arm/plat-samsung/include/plat/devs.h | 11 +- arch/sh/boards/mach-se/7724/setup.c | 5 + drivers/gpio/wm8994-gpio.c | 24 + drivers/mfd/wm8994-core.c | 93 +- drivers/regulator/wm8994-regulator.c | 23 +- drivers/video/sh_mobile_hdmi.c | 1 + include/linux/mfd/wm8994/core.h | 8 + include/linux/mfd/wm8994/pdata.h | 17 + include/linux/mfd/wm8994/registers.h | 302 ++ include/sound/alc5623.h | 15 + include/sound/asound.h | 3 + include/sound/control.h | 4 +- include/sound/hdsp.h | 1 + include/sound/minors.h | 4 +- include/sound/pcm.h | 1 + include/sound/soc-dai.h | 4 +- include/sound/soc-dapm.h | 99 +- include/sound/soc.h | 116 +- include/trace/events/asoc.h | 235 ++ sound/ac97_bus.c | 4 +- sound/aoa/codecs/onyx.c | 1 - sound/aoa/core/gpio-feature.c | 7 +- sound/aoa/core/gpio-pmf.c | 7 +- sound/core/control.c | 28 +- sound/core/oss/pcm_oss.c | 4 +- sound/core/pcm_lib.c | 22 + sound/core/pcm_native.c | 3 + sound/core/seq/seq.c | 4 + sound/core/sound.c | 18 +- sound/core/timer.c | 7 +- sound/drivers/ml403-ac97cr.c | 4 +- sound/i2c/other/ak4113.c | 5 +- sound/i2c/other/ak4114.c | 5 +- sound/pci/Kconfig | 23 +- sound/pci/ac97/ac97_codec.c | 6 +- sound/pci/azt3328.c | 406 +- sound/pci/bt87x.c | 10 +- sound/pci/cmipci.c | 25 +- sound/pci/hda/hda_codec.c | 3 + sound/pci/hda/hda_intel.c | 9 +- sound/pci/hda/patch_analog.c | 72 +- sound/pci/hda/patch_conexant.c | 108 +- sound/pci/hda/patch_hdmi.c | 92 +- sound/pci/hda/patch_realtek.c | 669 ++-- sound/pci/hda/patch_sigmatel.c | 15 +- sound/pci/hda/patch_via.c | 3 +- sound/pci/ice1712/delta.c | 49 +- sound/pci/ice1712/delta.h | 11 + sound/pci/oxygen/Makefile | 4 +- sound/pci/oxygen/cs4245.h | 107 + sound/pci/oxygen/hifier.c | 239 -- sound/pci/oxygen/oxygen.c | 356 ++- sound/pci/oxygen/oxygen.h | 19 +- sound/pci/oxygen/oxygen_io.c | 4 +- sound/pci/oxygen/oxygen_lib.c | 71 +- sound/pci/oxygen/oxygen_mixer.c | 110 +- sound/pci/oxygen/oxygen_pcm.c | 55 +- sound/pci/oxygen/oxygen_regs.h | 16 +- sound/pci/oxygen/xonar.h | 2 + sound/pci/oxygen/xonar_cs43xx.c | 84 +- sound/pci/oxygen/xonar_dg.c | 572 +++ sound/pci/oxygen/xonar_dg.h | 8 + sound/pci/oxygen/xonar_hdmi.c | 2 +- sound/pci/oxygen/xonar_lib.c | 6 +- sound/pci/oxygen/xonar_pcm179x.c | 473 ++-- sound/pci/oxygen/xonar_wm87x6.c | 317 ++- sound/pci/rme9652/hdsp.c | 538 +++- sound/pci/ymfpci/ymfpci_main.c | 12 +- sound/soc/Kconfig | 17 +- sound/soc/Makefile | 2 +- sound/soc/atmel/playpaq_wm8510.c | 12 +- sound/soc/atmel/sam9g20_wm8731.c | 18 +- sound/soc/atmel/snd-soc-afeb9260.c | 14 +- sound/soc/au1x/db1200.c | 1 - sound/soc/blackfin/bf5xx-ad1836.c | 1 - sound/soc/blackfin/bf5xx-ad193x.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/codecs/88pm860x-codec.c | 30 +- sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ad1836.c | 7 +- sound/soc/codecs/ad193x.c | 8 +- sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/ak4535.c | 20 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 11 +- sound/soc/codecs/alc5623.c | 1117 +++++ sound/soc/codecs/alc5623.h | 161 + sound/soc/codecs/cq93vc.c | 4 +- sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs42l51.c | 7 +- sound/soc/codecs/cx20442.c | 16 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/jz4740.c | 11 +- sound/soc/codecs/max98088.c | 13 +- sound/soc/codecs/ssm2602.c | 10 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tlv320aic23.c | 10 +- sound/soc/codecs/tlv320aic26.c | 5 +- sound/soc/codecs/tlv320aic3x.c | 71 +- sound/soc/codecs/tlv320dac33.c | 73 +- sound/soc/codecs/tpa6130a2.c | 100 +- sound/soc/codecs/tpa6130a2.h | 1 + sound/soc/codecs/twl4030.c | 56 +- sound/soc/codecs/twl6040.c | 855 ++++- sound/soc/codecs/twl6040.h | 8 + sound/soc/codecs/uda134x.c | 3 +- sound/soc/codecs/uda1380.c | 15 +- sound/soc/codecs/wl1273.c | 3 +- sound/soc/codecs/wm2000.c | 6 +- sound/soc/codecs/wm8350.c | 105 +- sound/soc/codecs/wm8400.c | 12 +- sound/soc/codecs/wm8510.c | 12 +- sound/soc/codecs/wm8523.c | 12 +- sound/soc/codecs/wm8580.c | 23 +- sound/soc/codecs/wm8711.c | 11 +- sound/soc/codecs/wm8728.c | 12 +- sound/soc/codecs/wm8731.c | 102 +- sound/soc/codecs/wm8737.c | 754 ++++ sound/soc/codecs/wm8737.h | 322 ++ sound/soc/codecs/wm8741.c | 10 +- sound/soc/codecs/wm8750.c | 13 +- sound/soc/codecs/wm8753.c | 49 +- sound/soc/codecs/wm8770.c | 749 ++++ sound/soc/codecs/wm8770.h | 51 + sound/soc/codecs/wm8776.c | 10 +- sound/soc/codecs/wm8804.c | 7 +- sound/soc/codecs/wm8900.c | 13 +- sound/soc/codecs/wm8903.c | 222 +- sound/soc/codecs/wm8903.h | 25 - sound/soc/codecs/wm8904.c | 36 +- sound/soc/codecs/wm8940.c | 8 +- sound/soc/codecs/wm8955.c | 12 +- sound/soc/codecs/wm8960.c | 29 +- sound/soc/codecs/wm8961.c | 13 +- sound/soc/codecs/wm8962.c | 41 +- sound/soc/codecs/wm8971.c | 30 +- sound/soc/codecs/wm8974.c | 13 +- sound/soc/codecs/wm8978.c | 13 +- sound/soc/codecs/wm8985.c | 12 +- sound/soc/codecs/wm8988.c | 11 +- sound/soc/codecs/wm8990.c | 12 +- sound/soc/codecs/wm8993.c | 21 +- sound/soc/codecs/wm8994-tables.c | 3147 +++++++++++++++ sound/soc/codecs/wm8994.c | 2544 ++++-------- sound/soc/codecs/wm8994.h | 14 + sound/soc/codecs/wm8995.c | 1818 +++++++++ sound/soc/codecs/wm8995.h | 4269 ++++++++++++++++++++ sound/soc/codecs/wm9081.c | 15 +- sound/soc/codecs/wm9090.c | 18 +- sound/soc/codecs/wm9705.c | 7 +- sound/soc/codecs/wm9712.c | 10 +- sound/soc/codecs/wm9713.c | 9 +- sound/soc/codecs/wm_hubs.c | 108 +- sound/soc/codecs/wm_hubs.h | 3 + sound/soc/davinci/davinci-evm.c | 23 +- sound/soc/davinci/davinci-sffsdr.c | 1 - sound/soc/ep93xx/ep93xx-i2s.c | 4 +- sound/soc/ep93xx/ep93xx-pcm.c | 4 +- sound/soc/ep93xx/snappercl15.c | 6 +- sound/soc/imx/eukrea-tlv320.c | 1 - sound/soc/imx/imx-ssi.c | 4 +- sound/soc/imx/phycore-ac97.c | 1 - sound/soc/imx/wm1133-ev1.c | 8 +- sound/soc/jz4740/jz4740-i2s.c | 1 - sound/soc/jz4740/qi_lb60.c | 14 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 141 + sound/soc/nuc900/nuc900-audio.c | 1 - sound/soc/omap/am3517evm.c | 14 +- sound/soc/omap/ams-delta.c | 82 +- sound/soc/omap/igep0020.c | 1 - sound/soc/omap/n810.c | 44 +- sound/soc/omap/omap2evm.c | 1 - sound/soc/omap/omap3beagle.c | 1 - sound/soc/omap/omap3evm.c | 1 - sound/soc/omap/omap3pandora.c | 45 +- sound/soc/omap/osk5912.c | 14 +- sound/soc/omap/overo.c | 1 - sound/soc/omap/rx51.c | 27 +- sound/soc/omap/sdp3430.c | 44 +- sound/soc/omap/sdp4430.c | 57 +- sound/soc/omap/zoom2.c | 36 +- sound/soc/pxa/corgi.c | 52 +- sound/soc/pxa/e740_wm9705.c | 30 +- sound/soc/pxa/e750_wm9705.c | 30 +- sound/soc/pxa/e800_wm9712.c | 8 +- sound/soc/pxa/em-x270.c | 1 - sound/soc/pxa/magician.c | 36 +- sound/soc/pxa/mioa701_wm9713.c | 18 +- sound/soc/pxa/palm27x.c | 34 +- sound/soc/pxa/poodle.c | 26 +- sound/soc/pxa/raumfeld.c | 1 - sound/soc/pxa/saarb.c | 18 +- sound/soc/pxa/spitz.c | 70 +- sound/soc/pxa/tavorevb3.c | 18 +- sound/soc/pxa/tosa.c | 38 +- sound/soc/pxa/z2.c | 16 +- sound/soc/pxa/zylonite.c | 12 +- sound/soc/s3c24xx/Kconfig | 171 - sound/soc/s3c24xx/Makefile | 55 - sound/soc/s3c24xx/aquila_wm8994.c | 295 -- sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 230 -- sound/soc/s3c24xx/s3c64xx-i2s.c | 242 -- sound/soc/s3c24xx/s3c64xx-i2s.h | 41 - sound/soc/s6000/s6105-ipcam.c | 42 +- sound/soc/samsung/Kconfig | 171 + sound/soc/samsung/Makefile | 55 + sound/soc/{s3c24xx/s3c-ac97.c => samsung/ac97.c} | 22 +- sound/soc/{s3c24xx/s3c-ac97.h => samsung/ac97.h} | 8 +- sound/soc/{s3c24xx/s3c-dma.c => samsung/dma.c} | 134 +- sound/soc/{s3c24xx/s3c-dma.h => samsung/dma.h} | 2 +- sound/soc/{s3c24xx => samsung}/goni_wm8994.c | 66 +- .../rx1950_uda1380.c => samsung/h1940_uda1380.c} | 249 +- sound/soc/samsung/i2s.c | 1258 ++++++ sound/soc/samsung/i2s.h | 29 + sound/soc/{s3c24xx => samsung}/jive_wm8750.c | 26 +- sound/soc/{s3c24xx => samsung}/lm4857.h | 0 sound/soc/{s3c24xx => samsung}/ln2440sbc_alc650.c | 9 +- .../{s3c24xx => samsung}/neo1973_gta02_wm8753.c | 72 +- sound/soc/{s3c24xx => samsung}/neo1973_wm8753.c | 130 +- sound/soc/{s3c24xx/s3c-pcm.c => samsung/pcm.c} | 6 +- sound/soc/{s3c24xx/s3c-pcm.h => samsung/pcm.h} | 2 +- sound/soc/{s3c24xx => samsung}/regs-i2s-v2.h | 0 sound/soc/{s3c24xx => samsung}/rx1950_uda1380.c | 16 +- sound/soc/{s3c24xx => samsung}/s3c-i2s-v2.c | 4 +- sound/soc/{s3c24xx => samsung}/s3c-i2s-v2.h | 2 +- sound/soc/{s3c24xx => samsung}/s3c2412-i2s.c | 4 +- sound/soc/{s3c24xx => samsung}/s3c2412-i2s.h | 2 +- sound/soc/{s3c24xx => samsung}/s3c24xx-i2s.c | 2 +- sound/soc/{s3c24xx => samsung}/s3c24xx-i2s.h | 0 sound/soc/{s3c24xx => samsung}/s3c24xx_simtec.c | 5 +- sound/soc/{s3c24xx => samsung}/s3c24xx_simtec.h | 2 +- .../{s3c24xx => samsung}/s3c24xx_simtec_hermes.c | 24 +- .../s3c24xx_simtec_tlv320aic23.c | 22 +- sound/soc/{s3c24xx => samsung}/s3c24xx_uda134x.c | 5 +- sound/soc/{s3c24xx => samsung}/smartq_wm8987.c | 64 +- sound/soc/{s3c24xx => samsung}/smdk2443_wm9710.c | 9 +- sound/soc/{s3c24xx => samsung}/smdk_spdif.c | 21 +- .../smdk64xx_wm8580.c => samsung/smdk_wm8580.c} | 174 +- sound/soc/samsung/smdk_wm8994.c | 176 + sound/soc/{s3c24xx => samsung}/smdk_wm9713.c | 26 +- sound/soc/{s3c24xx => samsung}/spdif.c | 4 +- sound/soc/{s3c24xx => samsung}/spdif.h | 2 +- sound/soc/sh/Kconfig | 4 +- sound/soc/sh/fsi-ak4642.c | 140 +- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi.c | 273 +- sound/soc/sh/migor.c | 6 +- sound/soc/sh/sh7760-ac97.c | 3 +- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sh/siu_pcm.c | 2 +- sound/soc/soc-cache.c | 1065 +++++- sound/soc/soc-core.c | 575 ++- sound/soc/soc-dapm.c | 860 +++-- sound/soc/soc-jack.c | 31 +- sound/usb/format.c | 5 +- sound/usb/midi.c | 19 +- sound/usb/mixer.c | 11 +- sound/usb/quirks-table.h | 4 +- sound/usb/usx2y/us122l.c | 41 +- 294 files changed, 24957 insertions(+), 7039 deletions(-) create mode 100644 arch/arm/mach-s5pv310/dev-audio.c create mode 100644 arch/arm/mach-s5pv310/dma.c create mode 100644 arch/arm/mach-s5pv310/include/mach/dma.h create mode 100644 arch/arm/plat-samsung/dev-asocdma.c create mode 100644 include/sound/alc5623.h create mode 100644 include/trace/events/asoc.h create mode 100644 sound/pci/oxygen/cs4245.h delete mode 100644 sound/pci/oxygen/hifier.c create mode 100644 sound/pci/oxygen/xonar_dg.c create mode 100644 sound/pci/oxygen/xonar_dg.h create mode 100644 sound/soc/codecs/alc5623.c create mode 100644 sound/soc/codecs/alc5623.h create mode 100644 sound/soc/codecs/wm8737.c create mode 100644 sound/soc/codecs/wm8737.h create mode 100644 sound/soc/codecs/wm8770.c create mode 100644 sound/soc/codecs/wm8770.h create mode 100644 sound/soc/codecs/wm8994-tables.c create mode 100644 sound/soc/codecs/wm8995.c create mode 100644 sound/soc/codecs/wm8995.h create mode 100644 sound/soc/kirkwood/kirkwood-t5325.c delete mode 100644 sound/soc/s3c24xx/Kconfig delete mode 100644 sound/soc/s3c24xx/Makefile delete mode 100644 sound/soc/s3c24xx/aquila_wm8994.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h create mode 100644 sound/soc/samsung/Kconfig create mode 100644 sound/soc/samsung/Makefile rename sound/soc/{s3c24xx/s3c-ac97.c => samsung/ac97.c} (96%) rename sound/soc/{s3c24xx/s3c-ac97.h => samsung/ac97.h} (73%) rename sound/soc/{s3c24xx/s3c-dma.c => samsung/dma.c} (74%) rename sound/soc/{s3c24xx/s3c-dma.h => samsung/dma.h} (97%) rename sound/soc/{s3c24xx => samsung}/goni_wm8994.c (84%) copy sound/soc/{s3c24xx/rx1950_uda1380.c => samsung/h1940_uda1380.c} (56%) create mode 100644 sound/soc/samsung/i2s.c create mode 100644 sound/soc/samsung/i2s.h rename sound/soc/{s3c24xx => samsung}/jive_wm8750.c (88%) rename sound/soc/{s3c24xx => samsung}/lm4857.h (100%) rename sound/soc/{s3c24xx => samsung}/ln2440sbc_alc650.c (92%) rename sound/soc/{s3c24xx => samsung}/neo1973_gta02_wm8753.c (88%) rename sound/soc/{s3c24xx => samsung}/neo1973_wm8753.c (83%) rename sound/soc/{s3c24xx/s3c-pcm.c => samsung/pcm.c} (99%) rename sound/soc/{s3c24xx/s3c-pcm.h => samsung/pcm.h} (99%) rename sound/soc/{s3c24xx => samsung}/regs-i2s-v2.h (100%) rename sound/soc/{s3c24xx => samsung}/rx1950_uda1380.c (95%) rename sound/soc/{s3c24xx => samsung}/s3c-i2s-v2.c (99%) rename sound/soc/{s3c24xx => samsung}/s3c-i2s-v2.h (98%) rename sound/soc/{s3c24xx => samsung}/s3c2412-i2s.c (98%) rename sound/soc/{s3c24xx => samsung}/s3c2412-i2s.h (95%) rename sound/soc/{s3c24xx => samsung}/s3c24xx-i2s.c (99%) rename sound/soc/{s3c24xx => samsung}/s3c24xx-i2s.h (100%) rename sound/soc/{s3c24xx => samsung}/s3c24xx_simtec.c (99%) rename sound/soc/{s3c24xx => samsung}/s3c24xx_simtec.h (94%) rename sound/soc/{s3c24xx => samsung}/s3c24xx_simtec_hermes.c (87%) rename sound/soc/{s3c24xx => samsung}/s3c24xx_simtec_tlv320aic23.c (86%) rename sound/soc/{s3c24xx => samsung}/s3c24xx_uda134x.c (99%) rename sound/soc/{s3c24xx => samsung}/smartq_wm8987.c (82%) rename sound/soc/{s3c24xx => samsung}/smdk2443_wm9710.c (92%) rename sound/soc/{s3c24xx => samsung}/smdk_spdif.c (95%) rename sound/soc/{s3c24xx/smdk64xx_wm8580.c => samsung/smdk_wm8580.c} (55%) create mode 100644 sound/soc/samsung/smdk_wm8994.c rename sound/soc/{s3c24xx => samsung}/smdk_wm9713.c (87%) rename sound/soc/{s3c24xx => samsung}/spdif.c (99%) rename sound/soc/{s3c24xx => samsung}/spdif.h (94%)

4 4

[alsa-devel] [GIT PULL] ASoC Final updates for 2.6.38
by Liam Girdwood 12 Jan '11

12 Jan '11

Later than usual since I've been on the road lately. The following changes since commit 991e02b4469c2e92cc98aa5b04fbde981671b74c: Merge branch 'for-2.6.38' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2010-12-17 16:43:17 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.38 David Lambert (1): ASoC: DMIC codec: Adding a generic DMIC codec Jorge Eduardo Candelaria (1): ASoC: mcbsp: Add McBSP support for OMAP4 Olaya, Margarita (1): ASoC: twl6040: Convert HF and HS drivers to use DAPM OUT_DRV widget Peter Ujfalusi (5): ASoC: tlv320dac33: Remove manual FIFO configuration ASoC: tlv320dac33: Some cleanup for 32/24 bit support ASoC: tlv320dac33: Add 32/24 bit audio support ASoC: tlv320dac33: Add DAPM selection for LOM invert ASoC: tpa6130a2: Fix compiler warning sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dmic.c | 81 ++++++++++++ sound/soc/codecs/tlv320dac33.c | 269 ++++++++++++++++------------------------ sound/soc/codecs/tpa6130a2.c | 1 - sound/soc/codecs/twl6040.c | 8 +- sound/soc/omap/omap-mcbsp.c | 35 +++++- sound/soc/omap/omap-mcbsp.h | 4 + 8 files changed, 234 insertions(+), 169 deletions(-) create mode 100644 sound/soc/codecs/dmic.c

2 1

[alsa-devel] snd_hda_intel: regression from kernel 2.6.36 -> .37 (Microphone)
by Wolfgang Scheicher 12 Jan '11

12 Jan '11

I used kernel 2.6.36 from debian experimental for quite some time now, without any problem: http://www.alsa-project.org/db/?f=11290fe50066e9bcd6a244f2c83b1c66c9dd201b Upgrading to 2.6.37 from debian experimental left me unable to use my microphone: http://www.alsa-project.org/db/?f=0f29e6de98ea9fefae5c6e2adb43ff8f8eede697 I also noticed some (inconsisitent?) changes in the mixer channel names, like "Mic" became "Rear Mic", while "Mic Boost" remained. Wolfgang Scheicher

2 1

[alsa-devel] alsa virtual sound card (with networked output)
by Metyl Methylius 12 Jan '11

12 Jan '11

Hello everyone, I was wondering if its possible to just create a "virtual" sound card using alsa which will just sent its audio stream to other computer (using TCP/IP or UDP for example). On the other end there would be some kind of server (or just nc -u -l -p 31337 | aplay ). I want to do it on alsa layer because other aps using alsa will get immediate support. It should be something like described in http://alsa.opensrc.org/index.php/Network but with transparent support for all alsa based applications. I dont care about syncronization or transmission delay. Any pointers as how this could be done in alsa ? metyl -- This mess was sent using 100% recycled electrons. If you want to get rid of this mess just turn off your computer,

4 5

[alsa-devel] [PATCH v2 0/4] Adding OMAP DMIC driver to kernel
by David Lambert 12 Jan '11

12 Jan '11

This is a patch series to add the OMAP Digital Microphone driver for OMAP4. It includes the driver, a generic DMIC codec, platform devices, as well as HWMOD entries for OMAP44xx chipsets. David Lambert (4): ASoC: DMIC: Adding the OMAP DMIC driver ASoC: DMIC codec: Adding a generic DMIC codec OMAP4: hwmod: add entries for DMIC driver OMAP4: DMIC: Add DMIC codec platform devices arch/arm/mach-omap2/board-4430sdp.c | 11 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 91 +++++ arch/arm/plat-omap/devices.c | 35 ++ arch/arm/plat-omap/include/plat/dmic.h | 82 ++++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dmic.c | 81 ++++ sound/soc/omap/Kconfig | 4 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 553 ++++++++++++++++++++++++++++ 10 files changed, 864 insertions(+), 0 deletions(-) create mode 100644 arch/arm/plat-omap/include/plat/dmic.h create mode 100644 sound/soc/codecs/dmic.c create mode 100644 sound/soc/omap/omap-dmic.c

4 10

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add DAPM selection for LOM invert
by Peter Ujfalusi 12 Jan '11

12 Jan '11

The L/R LOM line can be invertined side of the corresponding DAC, or inverted from the corresponding LOP. Add control for user space to select the source of the LOM inversion. When only the analog bypass is enabled, and the LOM is inverted from DAC output, we need to power the corresponding DAC. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 55 ++++++++++++++++++++++++++++++++++++--- 1 files changed, 50 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 13d521c..71d7be8 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -298,7 +298,6 @@ static void dac33_init_chip(struct snd_soc_codec *codec) if (unlikely(!dac33->chip_power)) return; - /* 44-46: DAC Control Registers */ /* A : DAC sample rate Fsref/1.5 */ dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0)); /* B : DAC src=normal, not muted */ @@ -321,6 +320,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_read_reg_cache(codec, DAC33_LINEL_TO_LLO_VOL)); dac33_write(codec, DAC33_LINER_TO_RLO_VOL, dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL)); + + dac33_write(codec, DAC33_OUT_AMP_CTRL, + dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL)); + } static inline int dac33_read_id(struct snd_soc_codec *codec) @@ -523,6 +526,25 @@ static const struct snd_kcontrol_new dac33_dapm_abypassl_control = static const struct snd_kcontrol_new dac33_dapm_abypassr_control = SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1); +/* LOP L/R invert selection */ +static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"}; + +static const struct soc_enum dac33_left_lom_enum = + SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3, + ARRAY_SIZE(dac33_lr_lom_texts), + dac33_lr_lom_texts); + +static const struct snd_kcontrol_new dac33_dapm_left_lom_control = +SOC_DAPM_ENUM("Route", dac33_left_lom_enum); + +static const struct soc_enum dac33_right_lom_enum = + SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2, + ARRAY_SIZE(dac33_lr_lom_texts), + dac33_lr_lom_texts); + +static const struct snd_kcontrol_new dac33_dapm_right_lom_control = +SOC_DAPM_ENUM("Route", dac33_right_lom_enum); + static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LEFT_LO"), SND_SOC_DAPM_OUTPUT("RIGHT_LO"), @@ -539,6 +561,18 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, &dac33_dapm_abypassr_control), + SND_SOC_DAPM_MUX("Left LOM Inverted From", SND_SOC_NOPM, 0, 0, + &dac33_dapm_left_lom_control), + SND_SOC_DAPM_MUX("Right LOM Inverted From", SND_SOC_NOPM, 0, 0, + &dac33_dapm_right_lom_control), + /* + * For DAPM path, when only the anlog bypass path is enabled, and the + * LOP inverted from the corresponding DAC side. + * This is needed, so we can attach the DAC power supply in this case. + */ + SND_SOC_DAPM_PGA("Left Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amplifier", DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amplifier", @@ -561,11 +595,22 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Output Left Amplifier", NULL, "DACL"}, {"Output Right Amplifier", NULL, "DACR"}, - {"Output Left Amplifier", NULL, "Analog Left Bypass"}, - {"Output Right Amplifier", NULL, "Analog Right Bypass"}, + {"Left Bypass PGA", NULL, "Analog Left Bypass"}, + {"Right Bypass PGA", NULL, "Analog Right Bypass"}, + + {"Left LOM Inverted From", "DAC", "Left Bypass PGA"}, + {"Right LOM Inverted From", "DAC", "Right Bypass PGA"}, + {"Left LOM Inverted From", "LOP", "Analog Left Bypass"}, + {"Right LOM Inverted From", "LOP", "Analog Right Bypass"}, + + {"Output Left Amplifier", NULL, "Left LOM Inverted From"}, + {"Output Right Amplifier", NULL, "Right LOM Inverted From"}, + + {"DACL", NULL, "Left DAC Power"}, + {"DACR", NULL, "Right DAC Power"}, - {"Output Left Amplifier", NULL, "Left DAC Power"}, - {"Output Right Amplifier", NULL, "Right DAC Power"}, + {"Left Bypass PGA", NULL, "Left DAC Power"}, + {"Right Bypass PGA", NULL, "Right DAC Power"}, /* output */ {"LEFT_LO", NULL, "Output Left Amplifier"}, -- 1.7.4.rc1

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[alsa-devel] [PATCH] ASoC: tpa6130a2: Fix compiler warning
by Peter Ujfalusi 12 Jan '11

12 Jan '11

sound/soc/codecs/tpa6130a2.c: In function 'tpa6130a2_add_controls': sound/soc/codecs/tpa6130a2.c:342: warning: unused variable 'dapm' Introduced by commit 39646871a47fd8808c08de0ce7d7ca8393af2805 ("ASoC: tpa6130a2: Replace DAPM code with direct interface"). The DAPM code has been removed from the driver, but the dapm struct remained. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0a99f31..1f1ac81 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -339,7 +339,6 @@ EXPORT_SYMBOL_GPL(tpa6130a2_stereo_enable); int tpa6130a2_add_controls(struct snd_soc_codec *codec) { struct tpa6130a2_data *data; - struct snd_soc_dapm_context *dapm = &codec->dapm; if (tpa6130a2_client == NULL) return -ENODEV; -- 1.7.3.4

3 2

[alsa-devel] [PATCH] ASoC: Fix resource leak if soc_register_ac97_dai_link failed
by Axel Lin 12 Jan '11

12 Jan '11

Properly free the resources in the case of soc_register_ac97_dai_link failure. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against Linus tree. Regards, Axel sound/soc/soc-core.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 441285a..d7f80c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1619,12 +1619,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #ifdef CONFIG_SND_SOC_AC97_BUS /* register any AC97 codecs */ for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); - goto probe_dai_err; - } + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + goto probe_dai_err; } + } #endif card->instantiated = 1; -- 1.7.0.4

3 2

[alsa-devel] [PATCH] configure.in: Fix xmlto detection
by Ozan Çağlayan 11 Jan '11

11 Jan '11

xmlto was never detected when alsaconf is disabled leading to a missing alsactl_init man page. --- configure.in | 12 +++++------- 1 files changed, 5 insertions(+), 7 deletions(-) diff --git a/configure.in b/configure.in index 2d6c80c..561ca46 100644 --- a/configure.in +++ b/configure.in @@ -102,13 +102,11 @@ AC_ARG_ENABLE(alsaloop, AM_CONDITIONAL(ALSALOOP, test x$alsaloop = xtrue) xmlto="" -if test x"$alsaconf" = xtrue; then - AC_ARG_ENABLE(xmlto, - AS_HELP_STRING([--disable-xmlto], [Disable man page creation via xmlto]), - xmlto="$enableval", xmlto="yes") - if test "$xmlto" = "yes"; then - AC_CHECK_PROG([xmlto], [xmlto], [yes]) - fi +AC_ARG_ENABLE(xmlto, + AS_HELP_STRING([--disable-xmlto], [Disable man page creation via xmlto]), + xmlto="$enableval", xmlto="yes") +if test "$xmlto" = "yes"; then + AC_CHECK_PROG([xmlto], [xmlto], [yes]) fi AM_CONDITIONAL(USE_XMLTO, test x"$xmlto" = xyes) -- 1.7.3.4

2 1

[alsa-devel] [PATCH] snd-usb-us122l: Fix MIDI output
by Karsten Wiese 11 Jan '11

11 Jan '11

The US-122L always reads 9 bytes per urb unless they are set to 0xFD. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/midi.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 25bce7e..b1f9ec9 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -850,8 +850,8 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, return; } - memset(urb->transfer_buffer + count, 0xFD, 9 - count); - urb->transfer_buffer_length = count; + memset(urb->transfer_buffer + count, 0xFD, ep->max_transfer - count); + urb->transfer_buffer_length = ep->max_transfer; } static struct usb_protocol_ops snd_usbmidi_122l_ops = { @@ -1295,6 +1295,13 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ ep->max_transfer = 4; break; + /* + * Some devices only work with 9 bytes packet size: + */ + case USB_ID(0x0644, 0x800E): /* Tascam US-122L */ + case USB_ID(0x0644, 0x800F): /* Tascam US-144 */ + ep->max_transfer = 9; + break; } for (i = 0; i < OUTPUT_URBS; ++i) { buffer = usb_alloc_coherent(umidi->dev, -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: soc-core: Simplify compress_type overriding functionality
by Dimitris Papastamos 11 Jan '11

11 Jan '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2ff708a..52aaef9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1748,6 +1748,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) list_for_each_entry(codec, &codec_list, list) { if (codec->cache_init) continue; + /* by default we don't override the compress_type */ + compress_type = 0; /* check to see if we need to override the compress_type */ for (i = 0; i < card->num_configs; ++i) { codec_conf = &card->codec_conf[i]; @@ -1758,18 +1760,6 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) break; } } - if (i == card->num_configs) { - /* no need to override the compress_type so - * go ahead and do the standard thing */ - ret = snd_soc_init_codec_cache(codec, 0); - if (ret < 0) { - mutex_unlock(&card->mutex); - return; - } - continue; - } - /* override the compress_type with the one supplied in - * the machine driver */ ret = snd_soc_init_codec_cache(codec, compress_type); if (ret < 0) { mutex_unlock(&card->mutex); -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ASoC: soc-cache: Clean up the cache manipulation code
by Dimitris Papastamos 11 Jan '11

11 Jan '11

Use Takashi's clean up code to make the cache manipulation code more readable. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 230 +++++++++++++++++------------------------------- 1 files changed, 82 insertions(+), 148 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8c2a21a..b0d86cc 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -761,6 +761,49 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); +static bool snd_soc_set_cache_val(void *base, unsigned int idx, + unsigned int val, unsigned int word_size) +{ + switch (word_size) { + case 1: { + u8 *cache = base; + if (cache[idx] == val) + return true; + cache[idx] = val; + break; + } + case 2: { + u16 *cache = base; + if (cache[idx] == val) + return true; + cache[idx] = val; + break; + } + default: + BUG(); + } + return false; +} + +static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, + unsigned int word_size) +{ + switch (word_size) { + case 1: { + const u8 *cache = base; + return cache[idx]; + } + case 2: { + const u16 *cache = base; + return cache[idx]; + } + default: + BUG(); + } + /* unreachable */ + return -1; +} + struct snd_soc_rbtree_node { struct rb_node node; unsigned int reg; @@ -924,7 +967,12 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) { + struct snd_soc_rbtree_node *rbtree_node; struct snd_soc_rbtree_ctx *rbtree_ctx; + unsigned int val; + unsigned int word_size; + int i; + int ret; codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL); if (!codec->reg_cache) @@ -936,53 +984,25 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) if (!codec->reg_def_copy) return 0; -/* - * populate the rbtree with the initialized registers. All other - * registers will be inserted into the tree when they are first written. - * - * The reasoning behind this, is that we need to step through and - * dereference the cache in u8/u16 increments without sacrificing - * portability. This could also be done using memcpy() but that would - * be slightly more cryptic. - */ -#define snd_soc_rbtree_populate(cache) \ -({ \ - int ret, i; \ - struct snd_soc_rbtree_node *rbtree_node; \ - \ - ret = 0; \ - cache = codec->reg_def_copy; \ - for (i = 0; i < codec->driver->reg_cache_size; ++i) { \ - if (!cache[i]) \ - continue; \ - rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); \ - if (!rbtree_node) { \ - ret = -ENOMEM; \ - snd_soc_cache_exit(codec); \ - break; \ - } \ - rbtree_node->reg = i; \ - rbtree_node->value = cache[i]; \ - rbtree_node->defval = cache[i]; \ - snd_soc_rbtree_insert(&rbtree_ctx->root, \ - rbtree_node); \ - } \ - ret; \ -}) - - switch (codec->driver->reg_word_size) { - case 1: { - const u8 *cache; - - return snd_soc_rbtree_populate(cache); - } - case 2: { - const u16 *cache; - - return snd_soc_rbtree_populate(cache); - } - default: - BUG(); + /* + * populate the rbtree with the initialized registers. All other + * registers will be inserted when they are first modified. + */ + word_size = codec->driver->reg_word_size; + for (i = 0; i < codec->driver->reg_cache_size; ++i) { + val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size); + if (!val) + continue; + rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); + if (!rbtree_node) { + ret = -ENOMEM; + snd_soc_cache_exit(codec); + break; + } + rbtree_node->reg = i; + rbtree_node->value = val; + rbtree_node->defval = val; + snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node); } return 0; @@ -1165,29 +1185,10 @@ static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, } /* write the new value to the cache */ - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - cache = lzo_block->dst; - if (cache[blkpos] == value) { - kfree(lzo_block->dst); - goto out; - } - cache[blkpos] = value; - } - break; - case 2: { - u16 *cache; - cache = lzo_block->dst; - if (cache[blkpos] == value) { - kfree(lzo_block->dst); - goto out; - } - cache[blkpos] = value; - } - break; - default: - BUG(); + if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value, + codec->driver->reg_word_size)) { + kfree(lzo_block->dst); + goto out; } /* prepare the source to be the decompressed block */ @@ -1241,25 +1242,10 @@ static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, /* decompress the block */ ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret >= 0) { + if (ret >= 0) /* fetch the value from the cache */ - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - cache = lzo_block->dst; - *value = cache[blkpos]; - } - break; - case 2: { - u16 *cache; - cache = lzo_block->dst; - *value = cache[blkpos]; - } - break; - default: - BUG(); - } - } + *value = snd_soc_get_cache_val(lzo_block->dst, blkpos, + codec->driver->reg_word_size); kfree(lzo_block->dst); /* restore the pointer and length of the compressed block */ @@ -1414,28 +1400,10 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; - if (codec_drv->reg_cache_default) { - switch (codec_drv->reg_word_size) { - case 1: { - const u8 *cache; - - cache = codec_drv->reg_cache_default; - if (cache[i] == val) - continue; - } - break; - case 2: { - const u16 *cache; - - cache = codec_drv->reg_cache_default; - if (cache[i] == val) - continue; - } - break; - default: - BUG(); - } - } + if (codec_drv->reg_cache_default) + if (snd_soc_get_cache_val(codec_drv->reg_cache_default, + i, codec_drv->reg_word_size) == val) + continue; ret = snd_soc_write(codec, i, val); if (ret) return ret; @@ -1448,50 +1416,16 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) static int snd_soc_flat_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - - cache = codec->reg_cache; - cache[reg] = value; - } - break; - case 2: { - u16 *cache; - - cache = codec->reg_cache; - cache[reg] = value; - } - break; - default: - BUG(); - } - + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); return 0; } static int snd_soc_flat_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value) { - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - - cache = codec->reg_cache; - *value = cache[reg]; - } - break; - case 2: { - u16 *cache; - - cache = codec->reg_cache; - *value = cache[reg]; - } - break; - default: - BUG(); - } - + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); return 0; } -- 1.7.3.5

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[alsa-devel] [PATCH] ASoC: soc-cache: Fix invalid memory access during snd_soc_lzo_cache_sync()
by Dimitris Papastamos 11 Jan '11

11 Jan '11

The size of the lzo syncing bitmap was incorrectly set to the size of the cache times the word size, however, the correct size is the size of the cache. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a9ebc07..8c2a21a 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1361,7 +1361,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) goto err; } lzo_blocks[i]->sync_bmp = sync_bmp; - lzo_blocks[i]->sync_bmp_nbits = reg_size; + lzo_blocks[i]->sync_bmp_nbits = bmp_size; /* alloc the working space for the compressed block */ ret = snd_soc_lzo_prepare(lzo_blocks[i]); if (ret < 0) -- 1.7.3.5

3 2

[alsa-devel] [PATCH] ucm: fix switch device & modifier when no transition is found
by Liam Girdwood 11 Jan '11

11 Jan '11

Fix some logic bugs in switch device and switch modifier when transition sequences are not found. Also fix check for new device. Reported-by: w0806.kim(a)samsung.com Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- src/ucm/main.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7305042..030c9b1 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1305,7 +1305,7 @@ static int switch_device(snd_use_case_mgr_t *uc_mgr, if (xold == NULL) return -ENOENT; xnew = find_device(uc_mgr->active_verb, new_device); - if (xold == NULL) + if (xnew == NULL) return -ENOENT; err = 0; list_for_each(pos, &xold->transition_list) { @@ -1327,7 +1327,7 @@ static int switch_device(snd_use_case_mgr_t *uc_mgr, err = set_device(uc_mgr, xold, 0); if (err < 0) return err; - err = set_device(uc_mgr, xnew, 0); + err = set_device(uc_mgr, xnew, 1); if (err < 0) return err; } @@ -1379,7 +1379,7 @@ static int switch_modifier(snd_use_case_mgr_t *uc_mgr, err = set_modifier(uc_mgr, xold, 0); if (err < 0) return err; - err = set_modifier(uc_mgr, xnew, 0); + err = set_modifier(uc_mgr, xnew, 1); if (err < 0) return err; } -- 1.7.1

1 0

[alsa-devel] HDA - hdmi_show_short_audio_desc puts wrong rates in eld#N
by Anssi Hannula 11 Jan '11

11 Jan '11

Hi! Just noticed that the sample rates reported in eld#X.Y are wrong. They are printed by hdmi_show_short_audio_desc() in hda_eld.c. That function uses the snd_print_pcm_rates() function to convert the rates bitmap to a string. However, snd_print_pcm_rates() takes a HDA rates bitmap, not a generic SNDRV_ one. (BTW, an interesting function name for an internal HDA thing...) This results in several errors. For example, my 32000 44100 48000 88200 96000 176400 192000 becomes 44100 48000 88200 176400 192000 384000. -- Anssi Hannula

1 1

[alsa-devel] UCM updates and fixes
by Liam Girdwood 11 Jan '11

11 Jan '11

Jarolsav, This fixes the remaining issues I can currently see in the alsa-lib ucm branch. It also includes a patch to execute commands, although I dont know if this is what you had in mind for exec. I've also done quite a bit of testing now. What is your merge plan for UCM ? We really need UCM merged soon as there are many users who depend on it. Thanks Liam

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