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Re: [alsa-devel] [PATCH] ALSA: ASoc: Add regulator support to CS4270 codec driver
by Mark Brown 07 Dec '09

07 Dec '09

On Sat, Nov 28, 2009 at 07:46:22PM -0700, Tabi Timur-B04825 wrote: > Have you tested it on a Freescale MPC8610 HPCD? Are you sure it still > compiles and builds on any PowerPC system? I can do a compile test on PowerPC (and that's one of the platforms that gets tested with linux-next too) but I don't have hardware to do runtime tests with. Timur, would you be OK with compile tests for now - I'd not expect to see 2.6.33 go out this year so if there are any runtime issues it should be possible to fix them once you're back in the office?

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[alsa-devel] plans to support mixer widget on hda?
by Alexey Fisher 07 Dec '09

07 Dec '09

hallo, are there any plans to support mixer widget? I found it's really nice. Currently i use it with hda-verb for karaoke, to record skype call or just to make some live mix for online radio. Alexey

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[alsa-devel] About TASCAM US-122 mkII support
by Alberto Martinez de la Ossa 06 Dec '09

06 Dec '09

Hello, I have just got the TASCAM US-122 mkI <http://www.tascam.com/products/us-122mkII.html>I audio/midi interface, which seemed to me a great product and a good deal. The problem is that I use a linux system (ubuntu-studio karmic) for music production and the interface can not be recognized when plugged. Among other reasons I chose this device because I thought it would work in linux: There were alsa drivers for its predecessors, the us-122 <http://alsa.opensrc.org/Tascam_US-122> and the us-122l <http://www.pps.jussieu.fr/%7Esmimram/tascam/> models. The latter, thanks to Karsten Wiese who wrote a patch for the us-122l version, which was latter included by default in newer alsa-firmware releases. Unfortunately, the new model (US-122 mkII), the one I have, is not supported yet. I have even tried to get it replaced by the previous us-122l model, but it is not available anymore in the music store -> http://www.justmusic.de On the other hand, the "linux specialist" of the music store told me that a new linux driver for the us-122 mkII is been prepared, although he doesn't know when it will be ready. Being you the experts I have found in the web in regards with ALSA and TASCAM usb interfaces, I have decided to directly ask you about this. I would really appreciate any information about the topic. Thanks in advance for your attention. Best regards,

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Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 06 Dec '09

06 Dec '09

Hi Raul, in my previous mail I put some snd_pcm_ioplug_set_param_minmax code in the wrong callback function. The corrected code should be below: #include <stdio.h> #include <stdint.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <pthread.h> #include <time.h> #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #define FILE_PERM 0644 #define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0])) #define DEBUG #ifdef DEBUG #define DBG(f, ...) \ fprintf(stderr, "%s:%i %i "f"\n", __FUNCTION__, __LINE__, getpid(), ## __VA_ARGS__); #else #define DBG(f, ...) #endif struct play_info { snd_pcm_ioplug_t io; char *filename; int file_fd; snd_pcm_sframes_t hw_ptr; }; static int snd_pcm_play_close(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_start(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_stop(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static snd_pcm_sframes_t snd_pcm_play_pointer(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("ptr: %lu", data->hw_ptr); return data->hw_ptr; } static snd_pcm_sframes_t snd_pcm_play_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { DBG("aborting with -EINVAL ..."); return -EINVAL; } static int snd_pcm_play_prepare(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("hw_ptr -> 0"); data->hw_ptr=0; return 0; } static int snd_pcm_play_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) { return 0; } static int snd_pcm_play_poll_descriptors_count(snd_pcm_ioplug_t *io) { DBG(""); return 1; } static int snd_pcm_play_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd, unsigned int space) { DBG(""); return 1; } static int snd_pcm_play_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { DBG(""); return 0; } static int snd_pcm_play_delay(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp) { DBG(""); return 0; } static snd_pcm_ioplug_callback_t play_pcm_callback = { .close = snd_pcm_play_close, .start = snd_pcm_play_start, .stop = snd_pcm_play_stop, .pointer = snd_pcm_play_pointer, .transfer = snd_pcm_play_transfer, .prepare = snd_pcm_play_prepare, .hw_params = snd_pcm_play_hw_params, .poll_descriptors_count = snd_pcm_play_poll_descriptors_count, .poll_descriptors = snd_pcm_play_poll_descriptors, .poll_revents = snd_pcm_play_poll_revents, .delay = snd_pcm_play_delay, }; static int play_hw_constraint(struct play_info * pcm) { snd_pcm_ioplug_t *io = &pcm->io; static const snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; static const unsigned int format_list[] = { SND_PCM_FORMAT_S16_LE, }; int err; DBG(""); /* access type */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) return err; /* supported formats */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(format_list), format_list); if (err < 0) return err; /* supported channels */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_CHANNELS, 1, 1); if (err < 0) return err; /* supported rate */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE, 8000, 8000); if (err < 0) return err; /* supported block size */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 156*2, 164*2); if (err < 0) return err; err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 200); if (err < 0) return err; return 0; } SND_PCM_PLUGIN_DEFINE_FUNC(play) { snd_config_iterator_t i, next; const char *filename=NULL; int err; struct play_info *play=NULL; if (stream != SND_PCM_STREAM_PLAYBACK) { SNDERR("play plugin can only be used for playback\n"); return -1; } snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "filename") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_STRING) { SNDERR("invalid type for %s\n", id); return -EINVAL; } if (snd_config_get_string(n, &filename)<0) { SNDERR("could not get filename for %s\n", id); return -EINVAL; } continue; } SNDERR("Unknown field %s", id); return -EINVAL; } if (!filename) { SNDERR("no filename defined or play plugin\n"); return -EINVAL; } if ((play = calloc(1, sizeof(*play))) == NULL) return -ENOMEM; play->io.version = SND_PCM_IOPLUG_VERSION; play->io.name = "ALSA play Plugin"; play->io.callback = &play_pcm_callback; play->io.private_data = play; play->io.mmap_rw = 0; if ((err = snd_pcm_ioplug_create(&play->io, name, stream, mode))<0) { free(play); return err; } err = play_hw_constraint(play); if (err < 0) { snd_pcm_ioplug_delete(&play->io); free(play); return err; } *pcmp = play->io.pcm; return err; } SND_PCM_PLUGIN_SYMBOL(play);

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Re: [alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 06 Dec '09

06 Dec '09

Hi Raul, Raul Xiong wrote: > Hi Stefan > > This is because your pointer callback function is not implemented properly. > "pointer" callback function should return the right position of the > playback/capture buffer, apps over alsa framework will call > snd_pcm_avail_update to get the position. > Also, you should call snd_pcm_ioplug_set_param_minmax in your > play_hw_constraint function to define the buffer size relevant parameters. Well, I did what you suggested but the start() callback is still not called: $ aplay -D play test_8khz_16LE_mono.wav Playing WAVE 'test_8khz_16LE_mono.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono snd_pcm_play_hw_params:84 24610 snd_pcm_play_prepare:71 24610 hw_ptr -> 0 snd_pcm_play_pointer:54 24610 ptr: 0 snd_pcm_play_transfer:63 24610 aborting with -EINVAL ... aplay: pcm_write:1442: write error: Invalid argument Also, when transferring PCM samples from the ALSA buffer to somewhere else, it seems logical to me that the initial value of the hw_ptr has to be 0, right ? As you can see above, the situation is still exactly the same and the start callback is never called at all. The code for the plugin is below. cheers, stefan #include <stdio.h> #include <stdint.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <pthread.h> #include <time.h> #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #define FILE_PERM 0644 #define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0])) #define DEBUG #ifdef DEBUG #define DBG(f, ...) \ fprintf(stderr, "%s:%i %i "f"\n", __FUNCTION__, __LINE__, getpid(), ## __VA_ARGS__); #else #define DBG(f, ...) #endif struct play_info { snd_pcm_ioplug_t io; char *filename; int file_fd; snd_pcm_sframes_t hw_ptr; snd_pcm_hw_params_t *hw_params; }; static int snd_pcm_play_close(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_start(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static int snd_pcm_play_stop(snd_pcm_ioplug_t *io) { DBG(""); return 0; } static snd_pcm_sframes_t snd_pcm_play_pointer(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("ptr: %lu", data->hw_ptr); return data->hw_ptr; } static snd_pcm_sframes_t snd_pcm_play_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { DBG("aborting with -EINVAL ..."); return -EINVAL; } static int snd_pcm_play_prepare(snd_pcm_ioplug_t *io) { struct play_info *data = io->private_data; DBG("hw_ptr -> 0"); data->hw_ptr=0; return 0; } static int snd_pcm_play_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) { snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; unsigned int format_list[] = { SND_PCM_FORMAT_S16 }; int err; DBG(""); /* access type */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) return err; /* supported formats */ err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(format_list), format_list); if (err < 0) return err; /* supported channels */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_CHANNELS, 1, 1); if (err < 0) return err; /* supported rate */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_RATE, 8000, 8000); if (err < 0) return err; /* supported block size */ err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 156*2, 164*2); if (err < 0) return err; err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 200); if (err < 0) return err; return 0; } static int snd_pcm_play_poll_descriptors_count(snd_pcm_ioplug_t *io) { DBG(""); return 1; } static int snd_pcm_play_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd, unsigned int space) { DBG(""); return 1; } static int snd_pcm_play_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { DBG(""); return 0; } static int snd_pcm_play_delay(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp) { DBG(""); return 0; } static snd_pcm_ioplug_callback_t play_pcm_callback = { .close = snd_pcm_play_close, .start = snd_pcm_play_start, .stop = snd_pcm_play_stop, .pointer = snd_pcm_play_pointer, .transfer = snd_pcm_play_transfer, .prepare = snd_pcm_play_prepare, .hw_params = snd_pcm_play_hw_params, .poll_descriptors_count = snd_pcm_play_poll_descriptors_count, .poll_descriptors = snd_pcm_play_poll_descriptors, .poll_revents = snd_pcm_play_poll_revents, .delay = snd_pcm_play_delay, }; static int play_hw_constraint(struct play_info * pcm) { snd_pcm_ioplug_t *io = &pcm->io; static const snd_pcm_access_t access_list[] = { SND_PCM_ACCESS_RW_INTERLEAVED }; static const unsigned int formats[] = { SND_PCM_FORMAT_S16_LE, }; int err; err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list); if (err < 0) { SNDERR("cannot set HW access constraint list: %s", snd_strerror(err)); return err; } err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, ARRAY_SIZE(formats), formats); if (err < 0) { SNDERR("cannot set HW format constraint list: %s", snd_strerror(err)); return err; } return 0; } SND_PCM_PLUGIN_DEFINE_FUNC(play) { snd_config_iterator_t i, next; const char *filename=NULL; int err; struct play_info *play=NULL; if (stream != SND_PCM_STREAM_PLAYBACK) { SNDERR("play plugin can only be used for playback\n"); return -1; } snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "filename") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_STRING) { SNDERR("invalid type for %s\n", id); return -EINVAL; } if (snd_config_get_string(n, &filename)<0) { SNDERR("could not get filename for %s\n", id); return -EINVAL; } continue; } SNDERR("Unknown field %s", id); return -EINVAL; } if (!filename) { SNDERR("no filename defined or play plugin\n"); return -EINVAL; } if ((play = calloc(1, sizeof(*play))) == NULL) return -ENOMEM; play->io.version = SND_PCM_IOPLUG_VERSION; play->io.name = "ALSA play Plugin"; play->io.callback = &play_pcm_callback; play->io.private_data = play; play->io.mmap_rw = 0; if ((err = snd_pcm_ioplug_create(&play->io, name, stream, mode))<0) { free(play); return err; } err = play_hw_constraint(play); if (err < 0) { snd_pcm_ioplug_delete(&play->io); free(play); return err; } *pcmp = play->io.pcm; return err; } SND_PCM_PLUGIN_SYMBOL(play);

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[alsa-devel] ControlNames: Input Source/ Capture Source/ Jack Mode?
by Alexey Fisher 06 Dec '09

06 Dec '09

Hallo, i have question about controlnames, or if i correctly understand it: Capture Source - you can capture only from one of two (or more) sources. On switching mic a or mic b will be enabled or disabled. /--(o)--mic b cap switch--- \--(x)--mic a Input Source - same as capture source but more verbose /---(Mute/Vol)--Mic a input switch--(a(x)/b(o))-- \---(Mute/Vol)--Mic b more complicated setup: +---(Mute/Vol)--Mic a input switch a-(a(x)/b(o))--+ +---(Mute/Vol)--Mic b input switch b-(a(x)/b(o))--+ jack mode - you can choice to put VREF on the line or not. /---(Mute/Vol)-(mode)-Mic a input switch--(a(x)/b(o))-- \---(Mute/Vol)-(mode)-Mic b i hope i understand it correctly. Now is my situation: +---------+--(Mute/Rec)-(mode)-Mic a input switch a-(a(x)/b(o)/mix(o))--+ | +---------+--(Mute/Rec)-(mode)-Mic b input switch b-(a(x)/b(o)/mix(o))--+ | +--mixer--+<---(Mute/Vol for Mic a/b) are there any way to make alsamixer less verbose but still productive?

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[alsa-devel] ALSA on Alienware M17x
by Adam Gray 06 Dec '09

06 Dec '09

Hi guys, I'm having a lot of trouble getting sound out of my Alienware M17x. I've attached the output from alsa-info. I'm running kernel version 2.6.28 with ALSA driver version 1.0.18. There's no sound at all from the speakers or the headphones. In alsamixer, all channels appear un-muted and turned up full. If you need any further information please let me know. Regards, Adam Gray

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[alsa-devel] when i plugin my headphone the sound come out from speaker and headphone
by Michael Domanski 05 Dec '09

05 Dec '09

Hi, When I plugin my headphone the sound come out from speaker and headphone. The speaker and headphone working in the same time. I want my speaker to mute when headphone is plugged. http://www.alsa-project.org/db/?f=841de9447634e7e35ae0171f968c70de253259af mixer@mdomansky ~ $ sudo lspci |grep -i "multi\|audio\|040[0-3]" 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) mixer@mdomansky ~ $ sudo rmmod snd-hda-intel mixer@mdomansky ~ $ sudo modprobe snd-hda-intel model=hp WARNING: Deprecated config file /etc/modprobe.conf, all config files belong into /etc/modprobe.d/. WARNING: All config files need .conf: /etc/modprobe.d/alsa, it will be ignored in a future release. mixer@mdomansky ~ $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x92400000 irq 22 mixer@mdomansky ~ $ speaker-test -c2 -D hw:0,0 -t wav -l1 ^^^^^ sounds ok ^^^^^^ -- Sincerely, Michael +7 904 512 59 36 MasterWeb.PRO

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 04 Dec '09

04 Dec '09

The following changes since commit a649d1fcc9bd2299cb06b6594fabb429fa50f174: Daniel Mack (1): ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Very last minute but these would qualify as bug fixes during the release cycle so I feel comfortable merging them now. Manuel Lauss (2): ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path sound/soc/au1x/dbdma2.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-)

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[alsa-devel] What's in sound git tree for 2.6.33
by Takashi Iwai 04 Dec '09

04 Dec '09

Hi, here is the annual report of sound subsystem for 2.6.33. For 2.6.33, sound bits aren't crazy huge like 2.6.31 or 2.6.32. HD-audio stuff is getting stabilized, but still lots of changes. There are many small fixes here and there, including (not surprisingly) even ISA drivers. Large amounts of changes are found for the support of new VIA HD-audio codecs, Intel HDMI codecs, and a few new drivers and board-supports like Infrasonic Quartet support. In ALSA core side, there are a couple of important fixes. The all "PC Speaker" and "PC Beep" mixer elements are renamed to "Beep". This was a confusion for a long time, and hopefully more understandable now. And, a long-standing PCM mmap problem on PPC and MIPS non-coherent architectures is fixed -- or better to say, Oops is worked around. We are still looking for a saner solution, but it'll be likely an issue for 2.6.34. The biggest change in HD-audio is that we enabled MSI as default now. So far, there is no regression report by this. But, if you see any IRQ issues newly on the new kernel, please report. For ASoC, this has been another fairly quiet release, too. Aside from the addition of virtual mux support to DAPM and some further preparatory work for multi-CODEC cards the majority of changes have been driver updates, including: - New drivers for ADS117x, AK4671, TLV320DAC33, TPA6130A2, WM8711 and WM8727. - Substantial improvements to DMA performance and reliability on MPC5200 and DaVinci. - Capture support for the FSI port on SH. Due to cross tree merge issues some updates to the TWL4030 MFD code are also included, enabling sharing of the CODEC hardware by support for the VIBRA interface of the devices. The tree is at: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus The short log and diffstat are below. Takashi === Akinobu Mita (2): ALSA: ice1712: Use bitrev8 ALSA: sound: usbmidi: Use hweight16 Aleksey Kunitskiy (1): ALSA: ice1724 - Patch for suspend/resume for ESI Juli@ Alexey Fisher (1): ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT Anuj Aggarwal (3): ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal ASoC: Adding OMAP3517 / AM3517 EVM support in ASOC ASoC: Modifying Kconfig/Makefile for AM3517 EVM Barry Song (4): ASoC: Provide API for reordering channels ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 ASoC: Blackfin I2S: use dai state rather than local counter ASoC: move setting ac97 platformdata earlier than ac97 read/write Ben Dooks (1): ASoC: S3C: Remove <plat/audio.h> Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Clemens Ladisch (23): sound: oxygen: do not try to restore nonexistent EEPROM sound: oxygen: fix for PI7C9X110 compatibility sound: virtuoso: split virtuoso.c sound: virtuoso: fix Xonar Essence ST support sound: oxygen: better defaults for upmixing control sound: oxygen: add stereo upmixing to center/LFE channels sound: oxygen: more hardware documentation sound: oxygen: cache codec registers sound: virtuoso: add headphone impedance control sound: oxygen: allow custom MCLK rates sound: virtuoso: add PCM1796 oversampling control sound: oxygen: add digital filter control sound: oxygen: add high-pass filter control sound: oxygen: fix input monitor control names control: remove snd_konctrol_volatile::owner_pid field control: use reference-counted pid sound: rawmidi: fix opened substreams count sound: pcm: record a substream's owner process sound: rawmidi: record a substream's owner process sound: usb-audio: add Roland UA-1G support sound: usb-audio: allow switching altsetting on Roland USB MIDI devices sound: usb: make the USB MIDI module more independent sound: usxxx: cleanup chip field Dan Carpenter (1): ALSA: remove unnecessary null check Daniel J Blueman (1): mfd: twl4030: fix ELF section mismatch... Daniel Mack (3): ASoC: CS4270: export de-emphasis filter as ALSA control ALSA: ARM: add Raumfeld audio support ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API Daniel T Chen (2): ALSA: intel8x0: Mute External Amplifier by default for Gateway 4525GZ ALSA: hda: Fix max PCM level to 0 dB for Fujitsu-Siemens laptops using CX20549 (Venice) Eero Nurkkala (2): ASoC: remove io_mutex ASoC: refactor snd_soc_update_bits() Einar Rünkaru (1): ALSA: hda - Make Dell Vostro 1015n mic and speaker switching work Enric Balletbò i Serra (1): ASoC: Add support for IGEP v2 Graeme Gregory (3): ASoC: pxa-ssp increase max_channels to 8 ASoC: ADS117x ADC driver ASoC: omap-mcbsp - add support for upto 16 channels. Grant Likely (6): ASoC/mpc5200: Track DMA position by period number instead of bytes ASoC/mpc5200: get rid of the appl_ptr tracking nonsense ASoC/mpc5200: Improve printk debug output for trigger ASoC/mpc5200: add to_psc_dma_stream() helper ASoC/mpc5200: fix enable/disable of AC97 slots ASoC: mpc5200: remove duplicate identical IRQ handler Igor Grinberg (1): ASoC: finally enable support for eXeda and CM-X300 Janusz Krzysztofik (2): ASoC: Amstrad Delta minor cleanups ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1 Jari Vanhala (1): ASoC: TWL4030: Vibra motor stop fix when it is driven with audio Jarkko Nikula (1): ASoC: Pandora: Pass SRG input clock frequency to the OMAP McBSP DAI Jaroslav Kysela (10): ALSA: rename "PC Speaker" and "PC Beep" controls to "Beep" ALSA: rename "PC Speaker" controls to "Speaker" ALSA: hda - proc - show which I/O NID is associated to PCM device ALSA: hda_intel: Digital PC Beep - change behaviour for input layer ALSA: hda_intel: Digital PC Beep - delay input device unregistration ALSA: hda: beep - add missing cancel_delayed_work ALSA: hda - add beep_mode module parameter ALSA: hda - proc - introduce Control: lines to show mixer<->NID assignment ALSA: hda - proc - add support for dynamic controls to mixer<->NID mapping ALSA: hda - move snd_hda_pcm_type_name from hda_codec.h to hda_local.h Jassi (2): ASoC: S3C I2S LRCLK polarity option. ASoC: Return correct codec clock in s3c64xx-i2s Jassi Brar (7): ASoC: Fixed arguments passed to SMDK64xx set_pll ASoC: Rename s3c24xx_pcm prefix to s3c_dma ASoC: Rename 's3c24xx-pcm' driver to 's3c-dma' ARM: S3C64XX: Define PCM Controller base registers ARM: S3C64XX: Added platform data header ARM: S3C64XX: Defined PCM controller platform devices ASoC: Added the CPU driver for PCM controllers Joonyoung Shim (4): ASoC: Factor out I2C 8 bit address 8 bit data I/O ASoC: AK4671: add ak4671 codec driver ASoC: Add jack_status_check callback function for GPIO jacks ASoC: Rename controls with a / in wm_hubs Julia Lawall (1): ASoC: Move dereference after NULL test Kailang Yang (1): ALSA: hda - Add ALC661/259, ALC892/888VD support Krzysztof Helt (24): ALSA: sscape: add supoort for SPEA Media FX/Reveal SC-600 ALSA: sscape: convert to firmware loader framework ALSA: sscape: remove MIDI instances counting with limit ULONG_MAX ALSA: sscape: force AD1848 codec mode on old Soundscape ALSA: sscape: coding style fixes ALSA: wss: convert CS4231 mixer to dB scale ALSA: wss: reuse CS4231 controls for AD1848 ALSA: snd_dma_pointer workaround for chipsets with buggy DMA sound: remove OSS Ensoniq SoundScape driver ALSA: es18xx: remove snd_card pointer from snd_es18xx structure ALSA: es18xx: remove snd_audiodrive structure ALSA: cs4236: detect chip in one pass ALSA: es18xx: code improvements ALSA: cs4236: update control names ALSA: opti-miro: use variables directly in the probe function ALSA: cs4236: add dB scale for all volume controls ALSA: opti-miro: remove snd_card pointer from snd_miro structure ALSA: opti-miro: make miro.h header available outside the alsa directory ALSA: opti-miro: expose ACI mixer to outside drivers ALSA: opti-miro: fix OOPS if hardware is not detected media/radio: New driver for the radio FM module on Miro PCM20 sound card ALSA: opti-miro: separate comon probing code ALSA: opti-miro: add PnP detection ALSA: opti9xx: remove snd_opti9xx fields Kuninori Morimoto (4): ASoC: sh: FSI: Remove DMA support ASoC: sh: FSI: Add capture support sh: ms7724se: Add runtime PM support for FSI ASoC: sh: fsi: Add runtime PM support Logan Li (1): ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Lydia Wang (30): ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro ALSA: HDA VIA: Change get_codec_type argument to hda_codec type ALSA: HDA VIA: Add VT1708B-CE codec support. ALSA: HDA VIA: Limit VT1702 AA-Path max volume ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type ALSA: HDA VIA: Add low current mode for power saving. ALSA: HDA VIA: Remove unused argument of via_new_analog_input ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls ALSA: HDA VIA: Rewrite via_independent_hp_put ALSA: HDA VIA: Add smart5.1 function. ALSA: HDA VIA: When changing input source, update power state. ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. ALSA: HDA VIA: Refresh front playback mute in via_hp_automute. ALSA: HDA VIA: Add Jack detect feature for VT1708. ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function. ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls. ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls. ALSA: HDA VIA: Replace MIC_BOOST_VOLUME. ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb ALSA: HDA VIA: Add VT1718S support. ALSA: HDA VIA: Add VT1828S and VT2020 support. ALSA: HDA VIA: Add VT1716S support. ALSA: HDA VIA: Add VT2002P support. ALSA: HDA VIA: Add VT1812 support. ALSA: HDA VIA: rename vt1708_control_templates[]. ALSA: HDA VIA: Change PW4 connect select default to to MW0. ALSA: HDA VIA: comments: update copyright, changeset, etc. ALSA: HDA VIA: Only cosmetic changes ALSA: VIA HDA: Add support for VT1818S. Manuel Lauss (5): ASoC: au1x: psc-ac97: verify correct codec register was read ASoC: au1x: psc-ac97: reorganize timeouts ASoC: au1x: convert to platform drivers. ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path Marin Mitov (1): ASoC: Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK Mark Brown (26): ASoC: Update WM8711 to driver model registration method ASoC: Factor out WM8711 cache I/O ASoC: Add SPI support to WM8711 ASoC: WM8711 minor cleanups ASoC: Add TLV information to WM8711 ASoC: Add source argument to PLL configuration ASoC: Allow per-route connectedness checks for supplies ASoC: Add S3C64xx IIS CDCLK source selection ASoC: Convert soc-cache to use C99 style initialisers for the table ASoC: Clean up WM8974 PLL configuration ASoC: Factor out analogue platform data from WM8993 ASoC: Simplify code for DAPM widget updates ASoC: Push DAPM enumeration register change test out ASoC: Add virtual enumeration support for DAPM muxes ASoC: Minor fixups to tpa6130a2 driver ASoC: Remove snd_soc_suspend_device() ASoC: Add regulator support for WM8731 ASoC: Minor SMDK64xx WM8580 cleanups ASoC: Staticise wm8727 driver structure ASoC: Move sysfs and debugfs functions to head of soc-core.c ASoC: Factor out snd_soc_init_card() ASoC: Update ads117x to current APIs ASoC: Add bit clock rate calculator utility functions ASoC: Remove redundant snd_soc_dapm_new_widgets() calls ASoC: Add BCLK calculation utility for TDM mode too S3C64XX: Staticise platform data for PCM devices Miguel Aguilar (1): ASoC: Davinci: Add audio codec support for DM365 EVM Mike Arthur (1): ASoC: Add WM8711 CODEC driver Mike Rapoport (1): ASoC: OMAP: enable Overo driver for CM-T35 Neil Jones (1): ASoC: Add support for the WM8727 DAC. Nicolas Ferre (1): ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek Ondrej Zary (2): ALSA: tea575x-tuner: fix mute snd-fm801: autodetect SF64-PCR (tuner-only) card Pavel Hofman (7): ALSA: ak4114 - fix errors in output selector bits ALSA: ak4620 support, codec regs listed in proc ALSA: ak4113 support ALSA: ice1724 - adding GPIO routines for mask and direction ALSA: ice1724 - pro-rate-locking makes sense only for internal clock mode ALSA: ice1724 - Support for multiple external clock types ALSA: ice1724 - Infrasonic Quartet support Peter Ujfalusi (21): ASoC: add support for multiple cards/codecs in debugfs ASoC: Improve the debugfs hierarchy ASoC: TPA6130A2 amplifier driver ASoC: TPA6130A2: Make tpa6130a2_power as static ASoC: Codec driver for Texas Instruments tlv320dac33 codec ASoC: TWL4030: Only update the needed bits in *set_dai_sysclk ASoC: tlv320dac33: typo fix in the header MFD: twl4030: add twl4030_codec MFD as a new child to the core OMAP: Platform support for twl4030_codec MFD ASoC: TWL4030: use the twl4030-codec.h for register descriptions ASoC: TWL4030: Driver registration via twl4030_codec MFD ASoC: TWL4030: Remove bypass tracking ASoC: TWL4030: Change codec_muted to apll_enabled ASoC: TWL4030: Change APLL powering sequence ASoC: TWL4030: Add APLL supply for the capture path ASoC: TWL4030: Make sure, that the codec is powered on startup MFD: TWL4030: Add audio_mclk to the codec platform data OMAP: Configure audio_mclk for twl4030-codec MFD MFD: twl4030-codec: APLL_INFREQ handling in the MFD driver ASoC: TWL4030: Do not modify the APLL_CTL register ASoC: tlv320dac33: Change RT wq to singlethread wq Rafael Ignacio Zurita (1): ALSA: sh: add SuperH DAC audio driver for ALSA V4 Roel Kluin (3): ALSA: Cleanup redundant tests on unsigned sound: OSS: fix error return in dma_ioctl() sound: OSS: keep index within bounds of midi_devs[] Russell King (1): ALSA: AACI cleanup Stas Sergeev (1): ALSA: snd-pcsp: add nopcm mode Stephen Rothwell (1): sound: use semicolons to end statements Takashi Iwai (41): ALSA: hda - Enable MSI as default ASoC: Fix build errors of wm8711.c with SPI ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c ALSA: sscape - Remove sscap_ioctl.h from include/sound/Kbuild ALSA: hda - Clean up name string creation in patch_realtek.c ALSA: sscape - Remove invalid __devinitdata to module parameters ALSA: hda - Add a proper ifdef to a debug code ALSA: hda - Switch to polling mode before disabling MSI ALSA: hda - Reset pins of IDT/STAC codecs at free ASoC: Remove dead code and labels ALSA: hda - Add reboot notifier to each codec ALSA: hda - Add missing export for snd_hda_bus_reboot_notify ALSA: hda - Add power on/off counter ALSA: hda - Don't access invalid substream in proc file ALSA: hda - Fix build error without CONFIG_SND_HDA_HWDEP=y ALSA: hda - Get rid of magic digits for subdev hack ALSA: hda - Add description of beep_mode in ALSA-Configuration.txt ALSA: hda - Update / add kerneldoc comments to exported functions ALSA: hda - Fix beep_mode option value ALSA: hda - Fix build errors with CONFIG_SND_HDA_INPUT_BEEP=n ALSA: hda - Disable default quirk for Sony VAIO with ALC262 codec ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect() ALSA: opti-miro: Fix missing semicolon ALSA: hda - Fix detection of dual headphones ALSA: hda - Fix mute-LED sync on HP laptops with IDT92HD83xxx codecs ALSA: hda - Change quirk for Acer Aspire 5930G ALSA: hda - Fix input and jack Kconfig depenencies Revert "ALSA: hda - Change quirk for Acer Aspire 5930G" ALSA: pcm - Use dma_mmap_coherent() if available ALSA: pcm - define snd_pcm_default_page_ops() ALSA: pcm - fix page conversion on non-coherent MIPS arch ALSA: pcm - fix page conversion on non-coherent PPC arch ALSA: Remove old DMA-mmap code from arm/devdma.c ALSA: hda - Don't trigger pin-sense for STAC/IDT codecs ASoC: Revert missing reset_err in wm97*.c ALSA: hda - Add a position_fix quirk for MSI Wind U115 ALSA: hda - Fix Cxt5047 test mode ALSA: hda - Add a pin-fix for FSC Amilo Pi1505 ALSA: hda - Add position_fix quirk for HP dv3 ALSA: aaci - Clean up duplicate code ALSA: hda - Fix memory leaks in the previous patch Timothy Knoll (1): sound: Kconfig typo fix Tobias Hansen (2): ALSA: snd-usb-us122l: add support for US-144 ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd Troy Kisky (5): ASoC: DaVinci: remove requirement that dma_params is 1st in structure ASoC: DaVinci: i2s, reduce underruns by combining into 1 element ASoC: DaVinci: pcm, rename variables in prep for ping/pong ASoC: DaVinci: pcm, fix underrun by using sram ASoC: DaVinci: use edma_pause, edma_resume Wu Fengguang (21): ALSA: hda - select IbexPeak handler for Calpella ALSA: hda - vectorize get_empty_pcm_device() ALSA: hda - allow up to 4 HDMI devices ALSA: hda - convert intelhdmi global references to local parameters ALSA: hda - remove intelhdmi dependency on multiout ALSA: hda - use pcm prepare/cleanup callbacks for intelhdmi ALSA: hda - reorder intelhdmi prepare/cleanup callbacks ALSA: hda - vectorize intelhdmi ALSA: hda - get intelhdmi max channels from widget caps ALSA: hda - auto parse intelhdmi cvt/pin configurations ALSA: hda - remove static intelhdmi configurations ALSA: intelhdmi - fix audio infoframe fill size ALSA: intelhdmi - fix channel mapping slot mask ALSA: intelhdmi - export monitor-presence and ELD-valid status ALSA: hda - introduce snd_hda_jack_detect() and snd_hda_pin_sense() ALSA: intelhdmi - probe for monitor/eld presence at module init time ALSA: intelhdmi - separate out infoframe checksum routine ALSA: intelhdmi - sticky infoframe ALSA: intelhdmi - sticky stream id and format ALSA: intelhdmi - sticky channel count ALSA: hda - show EPSS capability in proc Wu Zhangjin (1): ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency akpm(a)linux-foundation.org (1): sound/pci/hda/patch_via.c: work around gcc-4.0.2 ICE jassi brar (1): ASoC: Support WM8580 based audio subsystem on SMDK64xx machines Łukasz Wojniłowicz (1): ALSA: hda - 4930g mute lfe and side when pluging in headphones --- Documentation/sound/alsa/ALSA-Configuration.txt | 13 +- Documentation/sound/alsa/ControlNames.txt | 3 +- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + arch/arm/mach-davinci/include/mach/asp.h | 8 + arch/arm/mach-omap2/board-3430sdp.c | 10 + arch/arm/mach-omap2/board-omap3beagle.c | 10 + arch/arm/mach-omap2/board-omap3evm.c | 10 + arch/arm/mach-omap2/board-omap3pandora.c | 10 + arch/arm/mach-omap2/board-overo.c | 10 + arch/arm/mach-omap2/board-zoom2.c | 10 + arch/arm/mach-s3c6400/include/mach/map.h | 2 + arch/arm/plat-s3c/include/plat/audio.h | 48 +- arch/arm/plat-s3c/include/plat/devs.h | 3 + arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 2 + arch/arm/plat-s3c64xx/dev-audio.c | 101 +- arch/sh/boards/mach-hp6xx/setup.c | 55 + arch/sh/boards/mach-se/7724/setup.c | 3 + arch/sh/include/mach-common/mach/hp6xx.h | 4 + drivers/media/radio/Kconfig | 18 + drivers/media/radio/Makefile | 1 + drivers/media/radio/radio-miropcm20.c | 270 ++ drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 1 + drivers/mfd/twl4030-codec.c | 276 ++ drivers/mfd/twl4030-core.c | 16 +- include/linux/i2c/twl4030.h | 19 + include/linux/mfd/twl4030-codec.h | 272 ++ include/sound/Kbuild | 1 - sound/isa/opti9xx/miro.h => include/sound/aci.h | 23 +- include/sound/ak4113.h | 321 ++ include/sound/ak4114.h | 12 +- include/sound/ak4xxx-adda.h | 5 +- include/sound/control.h | 5 +- include/sound/cs4231-regs.h | 1 - include/sound/pcm.h | 3 + include/sound/rawmidi.h | 2 + include/sound/sh_dac_audio.h | 21 + include/sound/soc-dai.h | 14 +- include/sound/soc-dapm.h | 17 + include/sound/soc.h | 15 +- include/sound/sscape_ioctl.h | 21 - include/sound/tlv320dac33-plat.h | 20 + include/sound/tpa6130a2-plat.h | 30 + include/sound/wss.h | 1 - sound/Kconfig | 2 +- sound/arm/Makefile | 2 +- sound/arm/aaci.c | 35 +- sound/arm/devdma.c | 80 - sound/arm/devdma.h | 3 - sound/core/control.c | 9 +- sound/core/isadma.c | 10 +- sound/core/oss/mixer_oss.c | 4 +- sound/core/pcm.c | 6 +- sound/core/pcm_native.c | 83 +- sound/core/rawmidi.c | 17 +- sound/drivers/pcsp/pcsp.c | 32 +- sound/drivers/pcsp/pcsp.h | 2 +- sound/drivers/pcsp/pcsp_mixer.c | 35 +- sound/i2c/cs8427.c | 15 +- sound/i2c/other/Makefile | 3 +- sound/i2c/other/ak4113.c | 639 ++++ sound/i2c/other/ak4xxx-adda.c | 136 +- sound/i2c/other/tea575x-tuner.c | 2 +- sound/isa/Kconfig | 12 +- sound/isa/cmi8330.c | 4 +- sound/isa/cs423x/cs4236.c | 13 +- sound/isa/cs423x/cs4236_lib.c | 241 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 221 +- sound/isa/opti9xx/miro.c | 783 +++-- sound/isa/opti9xx/opti92x-ad1848.c | 110 +- sound/isa/sb/sb_mixer.c | 4 +- sound/isa/sscape.c | 727 ++--- sound/isa/wss/wss_lib.c | 105 +- sound/oss/Kconfig | 12 - sound/oss/Makefile | 1 - sound/oss/audio.c | 2 +- sound/oss/midi_synth.c | 2 +- sound/oss/mpu401.c | 2 +- sound/oss/sh_dac_audio.c | 3 - sound/oss/sscape.c | 1480 --------- sound/pci/Kconfig | 1 + sound/pci/ac97/ac97_codec.c | 6 +- sound/pci/ac97/ac97_patch.c | 12 +- sound/pci/azt3328.c | 4 +- sound/pci/ca0106/ca0106_mixer.c | 4 +- sound/pci/ca0106/ca0106_proc.c | 4 +- sound/pci/cmipci.c | 4 +- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +- sound/pci/emu10k1/emumixer.c | 4 +- sound/pci/emu10k1/emuproc.c | 4 +- sound/pci/emu10k1/io.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/fm801.c | 40 +- sound/pci/hda/Kconfig | 13 +- sound/pci/hda/hda_beep.c | 114 +- sound/pci/hda/hda_beep.h | 10 + sound/pci/hda/hda_codec.c | 607 +++- sound/pci/hda/hda_codec.h | 11 + sound/pci/hda/hda_eld.c | 20 +- sound/pci/hda/hda_generic.c | 17 +- sound/pci/hda/hda_hwdep.c | 38 + sound/pci/hda/hda_intel.c | 50 +- sound/pci/hda/hda_local.h | 69 +- sound/pci/hda/hda_proc.c | 70 + sound/pci/hda/patch_analog.c | 61 +- sound/pci/hda/patch_ca0110.c | 4 +- sound/pci/hda/patch_cirrus.c | 31 +- sound/pci/hda/patch_cmedia.c | 4 +- sound/pci/hda/patch_conexant.c | 189 +- sound/pci/hda/patch_intelhdmi.c | 488 +++- sound/pci/hda/patch_realtek.c | 440 ++-- sound/pci/hda/patch_sigmatel.c | 149 +- sound/pci/hda/patch_via.c | 3509 +++++++++++++++++++-- sound/pci/ice1712/Makefile | 2 +- sound/pci/ice1712/ice1712.c | 12 + sound/pci/ice1712/ice1712.h | 14 +- sound/pci/ice1712/ice1724.c | 103 +- sound/pci/ice1712/juli.c | 56 +- sound/pci/ice1712/quartet.c | 1130 +++++++ sound/pci/ice1712/quartet.h | 10 + sound/pci/intel8x0.c | 6 + sound/pci/oxygen/Makefile | 3 +- sound/pci/oxygen/cs2000.h | 83 + sound/pci/oxygen/hifier.c | 61 +- sound/pci/oxygen/oxygen.c | 248 ++- sound/pci/oxygen/oxygen.h | 5 + sound/pci/oxygen/oxygen_lib.c | 29 +- sound/pci/oxygen/oxygen_mixer.c | 52 +- sound/pci/oxygen/oxygen_pcm.c | 19 +- sound/pci/oxygen/virtuoso.c | 1105 +------- sound/pci/oxygen/xonar.h | 50 + sound/pci/oxygen/xonar_cs43xx.c | 434 +++ sound/pci/oxygen/xonar_hdmi.c | 128 + sound/pci/oxygen/xonar_lib.c | 132 + sound/pci/oxygen/xonar_pcm179x.c | 1115 +++++++ sound/ppc/awacs.c | 12 +- sound/ppc/burgundy.c | 8 +- sound/ppc/tumbler.c | 2 +- sound/sh/Kconfig | 8 + sound/sh/Makefile | 2 + sound/sh/sh_dac_audio.c | 453 +++ sound/soc/Makefile | 2 +- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/au1x/dbdma2.c | 115 +- sound/soc/au1x/psc-ac97.c | 243 +- sound/soc/au1x/psc-i2s.c | 189 +- sound/soc/au1x/psc.h | 7 +- sound/soc/blackfin/bf5xx-ad1836.c | 7 + sound/soc/blackfin/bf5xx-ad1938.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 15 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 45 +- sound/soc/blackfin/bf5xx-tdm.h | 11 + sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 12 + sound/soc/codecs/ac97.c | 3 - sound/soc/codecs/ad1836.c | 12 - sound/soc/codecs/ad1938.c | 12 - sound/soc/codecs/ad1980.c | 5 - sound/soc/codecs/ad73311.c | 8 - sound/soc/codecs/ads117x.c | 123 + sound/soc/codecs/ads117x.h | 13 + sound/soc/codecs/ak4104.c | 8 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4642.c | 9 - sound/soc/codecs/ak4671.c | 815 +++++ sound/soc/codecs/ak4671.h | 156 + sound/soc/codecs/cs4270.c | 28 +- sound/soc/codecs/cx20442.c | 12 - sound/soc/codecs/pcm3008.c | 9 - sound/soc/codecs/ssm2602.c | 9 - sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 11 +- sound/soc/codecs/tlv320aic26.c | 11 - sound/soc/codecs/tlv320aic3x.c | 11 - sound/soc/codecs/tlv320dac33.c | 1229 ++++++++ sound/soc/codecs/tlv320dac33.h | 267 ++ sound/soc/codecs/tpa6130a2.c | 463 +++ sound/soc/codecs/tpa6130a2.h | 61 + sound/soc/codecs/twl4030.c | 452 ++-- sound/soc/codecs/twl4030.h | 242 +-- sound/soc/codecs/uda134x.c | 9 - sound/soc/codecs/uda1380.c | 9 - sound/soc/codecs/wm8350.c | 32 +- sound/soc/codecs/wm8400.c | 32 +- sound/soc/codecs/wm8510.c | 14 +- sound/soc/codecs/wm8523.c | 26 - sound/soc/codecs/wm8580.c | 30 +- sound/soc/codecs/wm8711.c | 633 ++++ sound/soc/codecs/wm8711.h | 42 + sound/soc/codecs/wm8727.c | 135 + sound/soc/codecs/wm8727.h | 21 + sound/soc/codecs/wm8728.c | 10 - sound/soc/codecs/wm8731.c | 94 +- sound/soc/codecs/wm8750.c | 9 - sound/soc/codecs/wm8753.c | 49 +- sound/soc/codecs/wm8776.c | 43 - sound/soc/codecs/wm8900.c | 34 +- sound/soc/codecs/wm8903.c | 28 - sound/soc/codecs/wm8940.c | 28 +- sound/soc/codecs/wm8960.c | 30 +- sound/soc/codecs/wm8961.c | 27 - sound/soc/codecs/wm8971.c | 11 +- sound/soc/codecs/wm8974.c | 36 +- sound/soc/codecs/wm8988.c | 44 - sound/soc/codecs/wm8990.c | 14 +- sound/soc/codecs/wm8993.c | 49 +- sound/soc/codecs/wm9081.c | 27 - sound/soc/codecs/wm9705.c | 7 - sound/soc/codecs/wm9712.c | 7 - sound/soc/codecs/wm9713.c | 32 +- sound/soc/codecs/wm_hubs.c | 51 +- sound/soc/codecs/wm_hubs.h | 5 + sound/soc/davinci/Kconfig | 4 +- sound/soc/davinci/davinci-evm.c | 7 +- sound/soc/davinci/davinci-i2s.c | 85 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 5 - sound/soc/davinci/davinci-pcm.c | 571 +++- sound/soc/davinci/davinci-pcm.h | 2 + sound/soc/fsl/mpc5200_dma.c | 123 +- sound/soc/fsl/mpc5200_dma.h | 24 +- sound/soc/fsl/mpc5200_psc_ac97.c | 39 +- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/omap/Kconfig | 23 +- sound/soc/omap/Makefile | 4 + sound/soc/omap/am3517evm.c | 202 ++ sound/soc/omap/ams-delta.c | 4 +- sound/soc/omap/igep0020.c | 148 + sound/soc/omap/omap-mcbsp.c | 63 +- sound/soc/omap/omap3evm.c | 7 + sound/soc/omap/omap3pandora.c | 24 +- sound/soc/omap/overo.c | 4 +- sound/soc/pxa/Kconfig | 12 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 20 +- sound/soc/pxa/raumfeld.c | 335 ++ sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/Kconfig | 12 + sound/soc/s3c24xx/Makefile | 6 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 10 +- sound/soc/s3c24xx/neo1973_wm8753.c | 11 +- sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} | 88 +- sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} | 8 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 35 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +- sound/soc/s3c24xx/s3c-pcm.c | 552 ++++ sound/soc/s3c24xx/s3c-pcm.h | 123 + sound/soc/s3c24xx/s3c2412-i2s.c | 7 +- sound/soc/s3c24xx/s3c2443-ac97.c | 13 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 26 +- sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 268 ++ sound/soc/s6000/s6000-pcm.c | 4 +- sound/soc/sh/Kconfig | 1 - sound/soc/sh/fsi.c | 271 +- sound/soc/soc-cache.c | 46 +- sound/soc/soc-core.c | 566 ++-- sound/soc/soc-dapm.c | 135 +- sound/soc/soc-jack.c | 6 +- sound/soc/soc-utils.c | 74 + sound/usb/usbaudio.c | 38 +- sound/usb/usbaudio.h | 7 +- sound/usb/usbmidi.c | 208 +- sound/usb/usbmixer_maps.c | 23 + sound/usb/usbquirks.h | 23 + sound/usb/usx2y/us122l.c | 135 +- sound/usb/usx2y/us122l.h | 4 +- sound/usb/usx2y/usX2Yhwdep.c | 8 +- sound/usb/usx2y/usbusx2y.c | 28 +- sound/usb/usx2y/usbusx2y.h | 6 +- sound/usb/usx2y/usbusx2yaudio.c | 34 +- sound/usb/usx2y/usx2yhwdeppcm.c | 8 +- 285 files changed, 21108 insertions(+), 7537 deletions(-) create mode 100644 drivers/media/radio/radio-miropcm20.c create mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 include/linux/mfd/twl4030-codec.h rename sound/isa/opti9xx/miro.h => include/sound/aci.h (85%) create mode 100644 include/sound/ak4113.h create mode 100644 include/sound/sh_dac_audio.h delete mode 100644 include/sound/sscape_ioctl.h create mode 100644 include/sound/tlv320dac33-plat.h create mode 100644 include/sound/tpa6130a2-plat.h delete mode 100644 sound/arm/devdma.c delete mode 100644 sound/arm/devdma.h create mode 100644 sound/i2c/other/ak4113.c delete mode 100644 sound/oss/sscape.c create mode 100644 sound/pci/ice1712/quartet.c create mode 100644 sound/pci/ice1712/quartet.h create mode 100644 sound/pci/oxygen/cs2000.h create mode 100644 sound/pci/oxygen/xonar.h create mode 100644 sound/pci/oxygen/xonar_cs43xx.c create mode 100644 sound/pci/oxygen/xonar_hdmi.c create mode 100644 sound/pci/oxygen/xonar_lib.c create mode 100644 sound/pci/oxygen/xonar_pcm179x.c create mode 100644 sound/sh/sh_dac_audio.c create mode 100644 sound/soc/codecs/ads117x.c create mode 100644 sound/soc/codecs/ads117x.h create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h create mode 100644 sound/soc/codecs/tlv320dac33.c create mode 100644 sound/soc/codecs/tlv320dac33.h create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h create mode 100644 sound/soc/codecs/wm8727.c create mode 100644 sound/soc/codecs/wm8727.h create mode 100644 sound/soc/omap/am3517evm.c create mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/pxa/raumfeld.c rename sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} (82%) rename sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} (87%) create mode 100644 sound/soc/s3c24xx/s3c-pcm.c create mode 100644 sound/soc/s3c24xx/s3c-pcm.h create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c create mode 100644 sound/soc/soc-utils.c

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