Commit Graph

19396 Commits

Author SHA1 Message Date
Rasmus Villemoes
ee3043b2d7 ALSA: ctxfi: ct20k1reg: Fix typo in include guard
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22 11:24:57 +02:00
Rasmus Villemoes
3c25d04129 ALSA: hda: ca0132_regs.h: Fix typo in include guard
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22 11:24:11 +02:00
Clemens Ladisch
ddc64b278a ALSA: core: fix buffer overflow in snd_info_get_line()
snd_info_get_line() documents that its last parameter must be one
less than the buffer size, but this API design guarantees that
(literally) every caller gets it wrong.

Just change this parameter to have its obvious meaning.

Reported-by: Tommi Rantala <tt.rantala@gmail.com>
Cc: <stable@vger.kernel.org> # v2.2.26+
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22 07:21:05 +02:00
Libin Yang
ca2e7224d7 ALSA: hda/hdmi - apply Valleyview fix-ups to Cherryview display codec
Valleyview and Cherryview have the same behavior on display audio. So this patch
defines is_valleyview_plus() to include codecs for both Valleyview and its successor
Cherryview, and apply Valleyview fix-ups to Cherryview.

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-19 10:24:46 +02:00
Libin Yang
d35f64e748 ALSA: hda/hdmi - set depop_delay for haswell plus
Both Haswell and Broadwell need set depop_delay to 0. So apply this
setting to haswell plus.

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-19 10:24:37 +02:00
Hui Wang
f475371aa6 ALSA: hda - restore the gpio led after resume
On some HP laptops, the mute led is controlled by codec gpio.

When some machine resume from s3/s4, the codec gpio data will be
cleared to 0 by BIOS:
Before suspend:
  IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0
After resume:
  IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0

To skip the AFG node to enter D3 can't fix this problem.

A workaround is to restore the gpio data when the system resume
back from s3/s4. It is safe even on the machines without this
problem.

BugLink: https://bugs.launchpad.net/bugs/1358116
Tested-by: Franz Hsieh <franz.hsieh@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-19 07:57:46 +02:00
Takashi Iwai
f3ee07d8b6 ALSA: hda/realtek - Avoid setting wrong COEF on ALC269 & co
ALC269 & co have many vendor-specific setups with COEF verbs.
However, some verbs seem specific to some codec versions and they
result in the codec stalling.  Typically, such a case can be avoided
by checking the return value from reading a COEF.  If the return value
is -1, it implies that the COEF is invalid, thus it shouldn't be
written.

This patch adds the invalid COEF checks in appropriate places
accessing ALC269 and its variants.  The patch actually fixes the
resume problem on Acer AO725 laptop.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Tested-by: Francesco Muzio <muziofg@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-16 09:10:26 +02:00
Takashi Iwai
01d5500f35 ASoC: Fixes for v3.17
Nothing too exciting here, a bunch of driver fixes that came along since
 the initial pull request but none that really stand our and a warning
 fix in the core.
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Merge tag 'asoc-v3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.17

Nothing too exciting here, a bunch of driver fixes that came along since
the initial pull request but none that really stand our and a warning
fix in the core.
2014-08-16 09:10:19 +02:00
Mark Brown
7c063edee6 Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', 'asoc/fix/fsl-esai', 'asoc/fix/intel', 'asoc/fix/mcasp' and 'asoc/fix/pxa' into asoc-linus 2014-08-15 12:51:29 +01:00
Mark Brown
395d33bb16 Merge remote-tracking branch 'asoc/fix/pcm512x' into asoc-linus 2014-08-15 12:51:29 +01:00
Takashi Iwai
61074c1a2d ALSA: hda - Set TLV_DB_SCALE_MUTE bit for cx5051 vmaster
Conexnat HD-audio driver has a workaround for cx5051 (aka CX20561)
chip to add fake mute controls to each amp (commit 3868137e).  This
implies the minimum-as-mute TLV bit in TLV for each corresponding
control.  Meanwhile we build the virtual master from these, but the
TLV bit is missing, even though the slaves have it.

This patch simply adds the missing TLV_DB_SCALE_MUTE bit for vmaster,
as already done in patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-14 11:39:57 +02:00
Daniel Mack
9301503af0 ASoC: pxa-ssp: drop SNDRV_PCM_FMTBIT_S24_LE
This mode is unsupported, as the DMA controller can't do zero-padding
of samples.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-08-13 21:12:03 +01:00
Shengjiu Wang
769091ee18 ASoC: fsl-esai: Revert .xlate_tdm_slot_mask() support
This reverts commit a603c8ee52.

fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask().
fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit
to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled
bit to 0.
For esai when the bit value is 1, the slot is enabled, when the bit value is 0,
the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will
work abnormally. So revert this patch, make the esai use default function.

Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-13 19:50:00 +01:00
Jyri Sarha
8813543ecb ASoC: mcasp: Fix implicit BLCK divider setting
The implicit BLCK divider setting was broken by "ASoC: mcasp: don't
override bclk divider if it was provided by the machine"-patch. After
the BCLK divider is implicitly set for the first time the
mcasp->bclk_div gets a non zero value and the implicit setting is
"turned off".

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-12 23:04:08 +01:00
Nikesh Oswal
d114e5f73b ASoC: arizona: Fix TDM slot length handling in arizona_hw_params
TDM slot length was set same as word length, regardless of the value
received in set_tdm_slot. This patch sets the TDM slot length correctly
as received in set_tdm_slot DAI callback

Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-12 22:44:21 +01:00
Mark Brown
1c6d36805f ASoC: pcm512x: Correct Digital Playback control names
The source type should come before the direction specifier according to
ControlNames.txt.

Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-12 10:31:17 +01:00
Geert Uytterhoeven
6912831623 ASoC: dapm: Fix uninitialized variable in snd_soc_dapm_get_enum_double()
If soc_dapm_read() fails, reg_val will be uninitialized, and bogus
values will be written later:

sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double':
sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized]
  unsigned int reg_val, val;
               ^

Return early on error to fix this.

Introduced by commit ce0fc93ae5 ("ASoC:
Add DAPM support at the component level").

Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
2014-08-11 20:01:13 +01:00
Jarkko Nikula
b80d19c166 ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset
There is no need to restore and restart PCM streams in case ADSP didn't
reach reset and power off state during system suspend/resume cycle. In that
case stream is still active but paused and firmware doesn't allow allocating
a new stream before paused stream is freed.

ADSP remains active in case suspend sequence didn't go to suspend_late
stage. This can happen when either suspend sequence is aborted by a wakeup
or by letting only devices suspend by "echo devices >/sys/power/pm_test".

Currently stream restoring fails in these suspend cases. Fix this by adding
a flag that indicates is complete stream reinitialization needed or is it
enough to resume paused stream. Flag is set when we know that ADSP reached
suspend_late.

Initial fix to this issue came from Fang Yang. I modified it a little and
forward ported it to top of two other suspend/resume patches from me.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Cc: yang fang <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-11 13:40:24 +01:00
Jarkko Nikula
9246539bdd ASoC: Intel: Wait Baytrail ADSP boot at resume_early stage
Remove sst_byt_pcm_dev_resume() and move waiting of firmware boot into
sst_byt_pcm_dev_resume_early(). Now suspend_late and resume_early phases are
in sync with each other so that we know that ADSP was put into reset and was
unpowered after suspend_late and is ready to resume IO after resume_early
during resume stage in sst_byt_pcm_trigger().

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-11 13:40:23 +01:00
Jarkko Nikula
27d3f02689 ASoC: Intel: Merge Baytrail ADSP suspend_noirq into suspend_late
Merge DSP reset and cleanup sequence in sst_byt_pcm_dev_suspend_noirq()
into sst_byt_pcm_dev_suspend_late(). First their order was wrong by first
unloading firmware modules in suspend_late and then taking DSP into reset
in suspend_noirq. Second ACPI has put device into OFF state already during
suspend_late so trying to reset the DSP is a no-op at suspend_noirq stage.

Fix these by moving DSP reset and cleanup into
sst_byt_pcm_dev_suspend_late() before firmware unloading.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-11 13:40:23 +01:00
Takashi Iwai
e24aa0a4c5 ALSA: hda/ca0132 - Don't try loading firmware at resume when already failed
CA0132 driver tries to reload the firmware at resume.  Usually this
works since the firmware loader core caches the firmware contents by
itself.  However, if the driver failed to load the firmwares
(e.g. missing files), reloading the firmware at resume goes through
the actual file loading code path, and triggers a kernel WARNING like:

 WARNING: CPU: 10 PID:11371 at drivers/base/firmware_class.c:1105 _request_firmware+0x9ab/0x9d0()

For avoiding this situation, this patch makes CA0132 skipping the f/w
loading at resume when it failed at probe time.

Reported-and-tested-by: Janek Kozicki <cosurgi@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10 13:30:08 +02:00
Gabriele Mazzotta
de3da4f696 ALSA: hda - Fix pop noises on reboot for Dell XPS 13 9333
If nid 0x15 (Headphone Playback Switch) is in D3 and headphones are
plugged in when the laptop reboots, a pop noise is generated.
Prevent this by keeping nid 0x15 in D0 when headphones are plugged in.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10 13:28:51 +02:00
Gabriele Mazzotta
f38663ab5c ALSA: hda - Set internal mic as default input source on Dell XPS 13 9333
If the laptop is powered on with a jack plugged in, independently on what
is plugged, the jack is treated as a microphone jack.

Initialize the capture source so that by default jacks are treated as
headphones jacks. This will also prevent pop noises on boot in case
headphones are plugged in since setting/unsetting mic-in as input source
causes a pop noise.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10 13:28:39 +02:00
Clemens Ladisch
53da5ebfef ALSA: usb-audio: fix BOSS ME-25 MIDI regression
The BOSS ME-25 turns out not to have any useful descriptors in its MIDI
interface, so its needs a quirk entry after all.

Reported-and-tested-by: Kees van Veen <kees.vanveen@gmail.com>
Fixes: 8e5ced83dd ("ALSA: usb-audio: remove superfluous Roland quirks")
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10 13:28:04 +02:00
Takashi Iwai
875f0dd799 ALSA: hda - Fix parsing of CMI8888 codec
CMI8888 codec chip has a boost amp (only) on the headphone pin, and
this confuses the generic parser, which tends to pick up the most
outside amp.  This results in the wrong volume setup, as the driver
complains like:
  hda_codec: Mismatching dB step for vmaster slave (-100!=1000)

For avoiding this problem, rule out the amp on NID 0x10 and create
"Headphone Amp" volume control manually instead.

Note that this patch still doesn't fix all problems yet.  The sound
output from the line out seems still too low.  It will be fixed in
another patch (hopefully).

Reported-and-tested-by: Vincent Lejeune <vljn@ovi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-07 10:11:29 +02:00
Takashi Iwai
c563f473ac ALSA: hda - Fix probing and stuttering on CMI8888 HD-audio controller
ASUS Phoebus with CMI8888 HD-audio chip (PCI id 13f6:5011) doesn't
work with HD-audio driver as is because of some weird nature.  For
making DMA properly working, we need to disable MSI.  The position
report buffer doesn't work, thus we need to force reading LPIB
instead.  And yet, the codec CORB/RIRB communication gives errors
unless we disable the snooping (caching).

In this patch, all these workarounds are added as a quirk for the
device.  The HD-audio *codec* chip needs yet another workaround, but
it'll be provided in the succeeding patch.

Reported-and-tested-by: Vincent Lejeune <vljn@ovi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-07 10:11:26 +02:00
Kailang Yang
f7ae9ba0e8 ALSA: hda/realtek - Fixed ALC286/ALC288 recording delay for Headset Mic
It will be recording voice delay for resume back recording for Headset Mic.
This alc286 will quickly open Headset Mic, to prevent avoid recording files are missing.
The issue was fixed. This is follow ALC286 programing guide.

[fix build error, add static and renamed the function by tiwai]

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-07 10:08:21 +02:00
Jarkko Nikula
9b11eb44ef ASoC: Intel: Update Baytrail ADSP firmware name
Update the initial Baytrail ADSP firmware file name with the one that is now
in linux-firmware.git. Please see linux-firmware.git commit 7551a3a78453
("fw_sst_0f28: Add firmware for Intel Baytrail SST DSP").

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-06 20:36:13 +01:00
Himangi Saraogi
68a9edd18c sound: oss: Remove typedefs wanc_info and wavnc_port_info
The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedefs for wanc_info and
wavnc_port_info.

A simplified version of the Coccinelle semantic patch that finds the case is:

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i

Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-06 16:42:40 +02:00
Himangi Saraogi
80e7bbac48 sound: oss: uart401: Remove typedef uart401_devc
The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedef for uart401_devc.

The following Coccinelle semantic patch detects the case.

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i

Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-06 16:42:29 +02:00
Adam Goode
a509574e5e ALSA: usb-audio: Whitespace cleanups for sound/usb/midi.*
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05 20:08:00 +02:00
Adam Goode
f7881e5e8e ALSA: usb-audio: Respond to suspend and resume callbacks for MIDI input
sound/usb/card.c registers USB suspend and resume but did not previously
kill the input URBs. This means that USB MIDI devices left open across
suspend/resume had non-functional input (output still usually worked,
but it looks like that is another issue). Before this change, we would
get ESHUTDOWN for each of the input URBs at suspend time, killing input.

Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05 20:08:00 +02:00
Himangi Saraogi
81cb6b6be5 sound/oss/pss: Remove typedefs pss_mixerdata and pss_confdata
The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedefs for pss_mixerdata
and pss_confdata.

The following Coccinelle semantic patch is used to make the change.

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i

Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05 20:08:00 +02:00
Himangi Saraogi
8b0276d281 sound/oss/opl3: Remove typedef opl_devinfo
This typedef is unnecessary and should just be removed as they are
never used.

The following Coccinelle semantic patch detects the case.

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i

Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05 20:07:59 +02:00
Takashi Sakamoto
30225ed5ca ALSA: fireworks: fix specifiers in format strings for propper output
Use %d for loop counter and %X for device capabilities. This is a
supplemental patch for Hans Wennborg's patch.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05 09:15:20 +02:00
Takashi Iwai
e9e3bdffe3 ASoC: Updates for v3.17
This has been a pretty exciting release in terms of the framework, we've
 finally got support for multiple CODECs attached to a single DAI link
 which has been something there's been interest in as long as I've been
 working on ASoC.  A big thanks to Benoit and Misael for their work on
 this.
 
 Otherwise it's been a fairly standard release for development, including
 more componentisation work from Lars-Peter and a good selection of both
 CODEC and CPU drivers.
 
  - Support for multiple CODECs attached to a single DAI, enabling
    systems with for example multiple DAC/speaker drivers on a single
    link, contributed by Benoit Cousson based on work from Misael Lopez
    Cruz.
  - Support for byte controls larger than 256 bytes based on the use of
    TLVs contributed by Omair Mohammed Abdullah.
  - More componentisation work from Lars-Peter Clausen.
  - The remainder of the conversions of CODEC drivers to params_width()
  - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
    RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
    TAS2552.
  - Lots of updates and fixes, especially to the DaVinci, Intel,
    Freescale, Realtek, and rcar drivers.
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Merge tag 'asoc-v3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v3.17

This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC.  A big thanks to Benoit and Misael for their work on
this.

Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.

 - Support for multiple CODECs attached to a single DAI, enabling
   systems with for example multiple DAC/speaker drivers on a single
   link, contributed by Benoit Cousson based on work from Misael Lopez
   Cruz.
 - Support for byte controls larger than 256 bytes based on the use of
   TLVs contributed by Omair Mohammed Abdullah.
 - More componentisation work from Lars-Peter Clausen.
 - The remainder of the conversions of CODEC drivers to params_width()
 - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
   RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
   TAS2552.
 - Lots of updates and fixes, especially to the DaVinci, Intel,
   Freescale, Realtek, and rcar drivers.
2014-08-04 18:28:21 +02:00
Mark Brown
ae34a78c43 Merge remote-tracking branch 'asoc/topic/wm8985' into asoc-next 2014-08-04 16:32:25 +01:00
Mark Brown
e99a866c4c Merge remote-tracking branches 'asoc/topic/tlv320aic3x', 'asoc/topic/width', 'asoc/topic/wm0010', 'asoc/topic/wm8904' and 'asoc/topic/wm8962' into asoc-next 2014-08-04 16:32:18 +01:00
Mark Brown
c5e64c7636 Merge remote-tracking branches 'asoc/topic/tlv', 'asoc/topic/tlv320aic23', 'asoc/topic/tlv320aic31xx' and 'asoc/topic/tlv320aic32x4' into asoc-next 2014-08-04 16:32:12 +01:00
Mark Brown
3510a69495 Merge remote-tracking branches 'asoc/topic/sigmadsp', 'asoc/topic/sirf', 'asoc/topic/spdif', 'asoc/topic/tas2552' and 'asoc/topic/tas5086' into asoc-next 2014-08-04 16:32:07 +01:00
Mark Brown
c1bd940afd Merge remote-tracking branches 'asoc/topic/rt5670', 'asoc/topic/rt5677', 'asoc/topic/s6000', 'asoc/topic/samsung' and 'asoc/topic/sh-fsi' into asoc-next 2014-08-04 16:32:01 +01:00
Mark Brown
b0ae27c477 Merge remote-tracking branches 'asoc/topic/rl6231', 'asoc/topic/rockchip', 'asoc/topic/rt286', 'asoc/topic/rt5640' and 'asoc/topic/rt5645' into asoc-next 2014-08-04 16:31:55 +01:00
Mark Brown
6aa63a25c0 Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/oom' and 'asoc/topic/pxa' into asoc-next 2014-08-04 16:31:50 +01:00
Mark Brown
a6ce305207 Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood', 'asoc/topic/max98090' and 'asoc/topic/mc13783' into asoc-next 2014-08-04 16:31:45 +01:00
Mark Brown
e7177999dc Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-asrc', 'asoc/topic/fsl-spdif' and 'asoc/topic/imx-audmux' into asoc-next 2014-08-04 16:31:40 +01:00
Mark Brown
01954a7b67 Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs42l56', 'asoc/topic/cs42xx8', 'asoc/topic/cx20442' and 'asoc/topic/davinci' into asoc-next 2014-08-04 16:31:33 +01:00
Mark Brown
62e951ea65 Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/cirrus' and 'asoc/topic/cleanup' into asoc-next 2014-08-04 16:31:29 +01:00
Mark Brown
a486f183da Merge remote-tracking branches 'asoc/topic/adau1977', 'asoc/topic/ak4642', 'asoc/topic/ak5386' and 'asoc/topic/arizona' into asoc-next 2014-08-04 16:31:23 +01:00
Mark Brown
2fd5373467 Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next 2014-08-04 16:31:20 +01:00
Mark Brown
0e76ee41fc Merge remote-tracking branch 'asoc/topic/pcm1792' into asoc-next 2014-08-04 16:31:19 +01:00