Commit Graph

11782 Commits

Author SHA1 Message Date
Takashi Iwai
5c845beb42 ALSA: Don't prompt for CONFIG_SND_COMPRESS_OFFLOAD
CONFIG_SND_COMPRESS_OFFLOAD is an item to be selected by the dirver
just like CONFIG_SND_PCM, and no need to prompt for explicit
selection.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-13 09:53:53 +01:00
David Henningsson
b01de4fb40 ALSA: HDA: Use LPIB position fix for Macbook Pro 7,1
Several users have reported "choppy" audio under the 3.2 kernel,
and that changing position_fix to 1 has resolved their problem.
The chip is an nVidia Corporation MCP89 High Definition Audio,
[10de:0d94] (rev a2).

Cc: stable@kernel.org (v3.2+)
BugLink: https://bugs.launchpad.net/bugs/909419
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-13 09:50:23 +01:00
Takashi Iwai
9e4ce164ee Merge branch 'topic/hda' into for-linus 2012-01-12 09:59:18 +01:00
Takashi Iwai
627b79628f Merge branch 'topic/misc' into for-linus 2012-01-12 09:59:14 +01:00
Takashi Iwai
29abceb67f Merge branch 'for-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc 2012-01-12 09:48:20 +01:00
Takashi Iwai
d6b2450797 Merge branch 'for-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc into topic/asoc 2012-01-11 15:30:53 +01:00
Liam Girdwood
e48b46ba16 ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
The DL1 PDM interface adds a little gain depending on the output device.
Add a method to retrieve the gain value for machine driver usage.

Signed-off-by: Liam Girdwood <lrg@ti.com>
2012-01-11 12:43:24 +00:00
Takashi Iwai
f2cbba7602 ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
When multiple headphone or other detectable output pins are present,
the power-map has to be updated after resume appropriately, but the
current driver doesn't check all pins but only the first pin (since
it's enough to check it for the mute-behavior).  This resulted in the
silent output from the secondary outputs after PM resume.

This patch fixes the problem by checking all pins at (re-)init time.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740347

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 12:34:11 +01:00
Clemens Ladisch
8c3f5d8a9b ALSA: usb-audio: add Yamaha MOX6/MOX8 support
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 09:24:53 +01:00
Clemens Ladisch
56225e4cc8 ALSA: virtuoso: add S/PDIF input support for all Xonars
All Xonar cards support S/PDIF input, but the cards without optical or
coaxial plugs have only undocumented pin connectors.  Support for the
ST/STX was already added in a previous patch; this adds support for the
D1/DX (JP2), DG (J5), DS (J5), and HDAV Slim (J12).

Many thanks to Zoltan Miklos for testing the DS and DX.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 09:24:50 +01:00
Pavel Hofman
52cd0a76fd ALSA: ice1724 - Support for ooAoo SQ210a
This card shares PCI ids with Chaintec AV710. Therefore, it will not be
detected automatically, it can only be activated by the module parameter
model=sq210a.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 08:57:54 +01:00
Pavel Hofman
2b151ef734 ALSA: ice1724 - Allow card info based on model only
When two different cards share the same PCI vendor/subvendor
identification, allow card info based on model only.
Do not require subvendor ID.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 08:57:40 +01:00
Pavel Hofman
ffd364ddd3 ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
Add the capture pcm only if there is at least one ADC configured in
the SYSCONF register.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 08:56:59 +01:00
Adrian Knoth
f7de8ba3fc ALSA: hdspm - Provide unique driver id based on card serial
Before, /proc/asound looked like this:

 2 [Default        ]: HDSPM - RME RayDAT_f1cd85
                      RME RayDAT S/N 0xf1cd85 at 0xf7300000, irq 18

In case of a second HDSPM card, its name would be Default_1. This is
cumbersome, because the order of the cards isn't stable across reboots.

To help userspace tools referring to the correct card, this commit
provides a unique id for each card:

 2 [HDSPMxf1cd85   ]: HDSPM - RME RayDAT_f1cd85
                      RME RayDAT S/N 0xf1cd85 at 0xf7300000, irq 18

In this example, userspace (configuration files) would then use
hw:HDSPMxf1cd85 to choose the right card.

The serial is masked to 24bits, so this string is always shorter than
sixteen chars.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 08:55:58 +01:00
Mark Brown
36ae1a96c4 ASoC: Dynamically allocate the rtd device for a non-empty release()
The device model needs a release() function so it can free devices when
they become dereferenced.  Do that for rtds.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-10 14:53:56 -08:00
Axel Lin
e4e9e05409 ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
commit 739be96 "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC"
introduces below build warnings:

drivers/misc/Kconfig:212:error: recursive dependency detected!
drivers/misc/Kconfig:212:       symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC
sound/soc/atmel/Kconfig:9:      symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731
sound/soc/atmel/Kconfig:18:     symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC

SND_ATMEL_SOC_SSC needs ATMEL_SSC to pass compilation.
This patch remove the "select ATMEL_SSC" from SND_ATMEL_SOC_SSC to avoid above
warnings. And then ensures all the machine drivers that select SND_ATMEL_SOC_SSC
need to depend on ATMEL_SSC.

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-10 14:52:20 -08:00
Takashi Iwai
4808d12d1d ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
Currently the driver checks only the out_mix_path[] for the primary
output route for judging whether to create the loopback-mixing control
or not.  But, there are cases where aamix-routing is available only on
headphone or speaker paths but not on the primary output path.  So, the
driver ignores such cases inappropriately.

This patch fixes the check of the loopback-mixing control by testing
all mix-routing paths.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-10 15:16:02 +01:00
Takashi Iwai
3a90274de3 ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
When an invalid NID is given, get_wcaps() returns zero as the error,
but get_wcaps_type() takes it as the normal value and returns a bogus
AC_WID_AUD_OUT value.  This confuses the parser.

With this patch, get_wcaps_type() returns -1 when value 0 is given,
i.e. an invalid NID is passed to get_wcaps().

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740118

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-10 12:41:22 +01:00
Takashi Iwai
de4da59e48 ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
These laptops can work well with the auto-parser and their BIOS setups,
and in addition, the auto-parser fixes the problem with S3/S4 where
the unsol event handling is killed after resume due to fallback to the
single-cmd mode.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740115

Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-10 09:04:26 +01:00
Takashi Iwai
74eeb141d3 ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
Dan Carpenter reported that setting 0 to uinfo->value.enumerated.items
in snd_asihpi_cmode_info() may lead to Oops.  This function should
return an error immediately in such a case instead.

Cc: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 18:27:54 +01:00
Takashi Iwai
9badda0a0a ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
Dan Carpenter reported that setting 0 to uinfo->value.enumerated.items
in snd_hdsp_info_pref_sync_ref() may lead to Oops.  This function should
return an error immediately in such a case instead.

Cc: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 18:27:44 +01:00
Jérémy Lal
7e5bea19ae ALSA: hda/cirrus - support for iMac12,2 model
This early 2011 model just need to have headphones on GPI02
instead of GPI01, and use BIOS pincfgs.
It is detected by codec SSID.
The iMac12,1 model is known to work the same way, although maybe
not with the same codec SSID.

Signed-off-by: Jérémy Lal <kapouer@melix.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 17:26:25 +01:00
Janusz Krzysztofik
f75a8ff67d ASoC: cx20442: add bias control over a platform provided regulator
Now that a regulator device for controlling the codec chip reset state
over a platform agnostic regulator API is available on the only board
using this driver so far, extend the driver with a bias control function
which will request virtual power to the codec chip from that virtual
regulator, and will supersede the present implementation existing at the
sound card level.

Thanks to the regulator sharing mechanism, both the old (the sound card)
and the new (the codec) implementations should coexist smoothly until
the sound card file is updated. For this to work as expected, update the
sound card .set_bias_level callback to not touch codec->dapm.bias_level.

While extending the cx20442 structure, drop unused control_type member.

Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1:
ams-delta: set up a regulator over the modem reset GPIO pin".

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2012-01-09 11:20:16 +00:00
Takashi Iwai
80c8a2a372 ALSA: usb-audio - Avoid flood of frame-active debug messages
With some buggy devices, the usb-audio driver may give "frame xxx active"
kernel messages too often.  Better to keep it as debug-only using
snd_printdd(), and also add the rate-limit for avoiding floods.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 11:40:46 +01:00
Karsten Wiese
d0f3a2eb90 ALSA: snd-usb-us122l: Delete calls to preempt_disable
They are not needed here.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 11:31:30 +01:00
Mark Brown
b2ed1b0bc6 ASoC: Fix idma build after update for channel count check
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-08 23:04:28 -08:00
Adrian Knoth
7d53a631ed ALSA: hdspm - Refactor serial number to avoid code duplication
The serial number is used multiple times in hdspm.c. Since it belongs
to the card, let's store it in struct hdspm and refer to it whenever
necessary.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 16:05:40 +01:00
Xi Wang
4fa0e81b83 ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()
A malicious USB device may feed in carefully crafted min/max/res values,
so that the inner loop in parse_uac2_sample_rate_range() could run for
a long time or even never terminate, e.g., given max = INT_MAX.

Also nr_rates could be a large integer, which causes an integer overflow
in the subsequent call to kmalloc() in parse_audio_format_rates_v2().
Thus, kmalloc() would allocate a smaller buffer than expected, leading
to a memory corruption.

To exploit the two vulnerabilities, an attacker needs physical access
to the machine to plug in a malicious USB device.

This patch makes two changes.

1) The type of "rate" is changed to unsigned int, so that the loop could
   stop once "rate" is larger than INT_MAX.

2) Limit nr_rates to 1024.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 16:03:12 +01:00
Raymond Yau
fb65c2dfe6 ALSA: Au88x0 - Fix channels swapping of 4 channels playback
Fix channels swapping of 4 channels playback by
using vortex_adbdma_stopfifo instead of vortex_adbdma_pausefifo
for SNDRV_PCM_TRIGGER_STOP event

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 14:39:26 +01:00
Raymond Yau
3ae4e1f7a0 ALSA: Au88x0 - Fix IRQ fifo error and channels swap of 4 channels playback
Fix IRQ fifo error when playing stereo by set stereo flag of fifo control.
This also fix the swap of front and rear channels on au8830.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 14:38:42 +01:00
Raymond Yau
76474da05f ALSA: Au88x0 - Fix Xtalk's constants
- Fix XtalkGainsDefault, XtalkGains1Chn
- Fix XtalkWideCoefsLeftEQ, XtalkWideCoefsRightEQ
- Fix XtlakWideCoefsLeftXT, XtalkWideCoefsRightXT

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 14:35:40 +01:00
Raymond Yau
9489f2c63f ALSA: Au88x0 - Xtalk - fix write/read of eq and xt instates
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 14:35:29 +01:00
Pavel Hofman
219e2cd41b ALSA: ice1724 - External clock item only for cards with SPDIF_IN
Append the external clock item to the clock list only if
the SPDIF_IN capability is defined in the SPDIF register.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 14:28:32 +01:00
Pavel Hofman
e7848163aa ALSA: ice1724 - Check for ac97 to avoid kernel oops
Cards with identical PCI ids but no AC97 config in EEPROM do not have
the ac97 field initialized. We must check for this case to avoid kernel oops.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 14:26:01 +01:00
David Henningsson
f16c2cc3c4 ALSA: HDA: Remove Poulsbo position fix quirks
Now that we have changed the poulsbo chip to use LPIB position fix,
we can remove the individual machine quirks that do the same thing.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 10:00:10 +01:00
David Henningsson
2267ea9762 ALSA: HDA: Fix typo for ALC269VB_FIXUP_DMIC
This fixup is not actually used, so in practice this is just a
cosmetic fix.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 09:59:30 +01:00
David Henningsson
5660ffd069 ALSA: HDA: Add support for Cirrus Logic 4213
The CS4213 chip is similar to the CS4210, but it does not have
SPDIF capabilities. Also, it has fewer pins, and the vendor specific
nid is different. With this patch, we have working inputs and outputs
(and automute/autoswitch). However, we don't know anything about
the vendor specific processing coefficients, so we don't read or write
to that node in this patch.

BugLink: https://bugs.launchpad.net/bugs/910792
Tested-by: Hsin-Yi Chen <hychen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 09:57:37 +01:00
David Henningsson
78e2a928e3 ALSA: HDA: Fix automute for Cirrus Logic 421x
There was a bug in the automute logic causing speakers not to
mute when headphones were plugged in.

Cc: stable@kernel.org
Tested-by: Hsin-Yi Chen <hychen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 09:57:18 +01:00
David Henningsson
40d03e63e9 ALSA: HDA: Fix master control for Cirrus Logic 421X
The control name "HP/Speakers" is non-standard, and since there is
only one DAC on this chip there is no need for a virtual master
anyway.

Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 09:55:13 +01:00
David Henningsson
716e5db488 ALSA: HDA: Use LPIB position fix for Oaktrail
According to the thread on alsa-devel, the LPIB method is to prefer
for Oaktrail controller chip.

Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-January/047800.html

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 09:54:37 +01:00
Joachim Eastwood
25e9e7565f ASoC: check for substream not channels_min in pcm engines
This is a follow up on 53dea36c70 which fixes the other affected
pcm engines.

Description from 53dea36c70c1857:
 Don't rely on the codec's channels_min information to decide wheter or
 not allocate a substream's DMA buffer. Rather check if the substream
 itself was allocated previously.

Without this patch I was seeing null-pointer dereferenc in atmel-pcm.

Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-07 11:18:05 -08:00
Axel Lin
739be96ab8 ASoC: Fix build dependency for SND_ATMEL_SOC_SSC
Make SND_ATMEL_SOC_SSC select ATMEL_SSC to fix below build errors:

  LD      .tmp_vmlinux1
sound/built-in.o: In function `atmel_ssc_remove':
sound/soc/atmel/atmel_ssc_dai.c:713: undefined reference to `ssc_free'
sound/built-in.o: In function `atmel_ssc_probe':
sound/soc/atmel/atmel_ssc_dai.c:700: undefined reference to `ssc_request'
sound/built-in.o: In function `atmel_ssc_set_audio':
sound/soc/atmel/atmel_ssc_dai.c:845: undefined reference to `ssc_request'
sound/soc/atmel/atmel_ssc_dai.c:851: undefined reference to `ssc_free'
make: *** [.tmp_vmlinux1] Error 1

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-05 23:00:12 -08:00
Axel Lin
7a748e4318 ASoC: sta32x: Optimize the array work to find rate_min and rate_max
For a given ir and fs, there is at most one possible match for the case
mclk_ratios[ir][j].ratio * fs == freq.
Thus we can break from the inner loop once a match is found.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-03 21:18:32 +00:00
Sangsu Park
a500231da4 ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs
The original code does not cover the case that two DAIs(CPU) have different
ASoC core PCM operations(like mmap, pointer...). Currently we have only one
global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different
pointer functions, second DAI's pointer function is set for both first DAI
and second DAI in case of original code.

This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So
each DAIs can have different ASoC core PCM operations. This is needed to
support multiple DAIs.

Signed-off-by: Sangsu Park <sangsu4u.park@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-03 20:28:51 +00:00
Axel Lin
34be9244c7 ASoC: pxa: Convert corgi to use snd_soc_register_card()
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Haojian Zhuang <haojian.zhuang@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02 13:08:40 +00:00
Axel Lin
748b217827 ASoC: Fix return value of wm8580_set_sysclk()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02 13:08:40 +00:00
Axel Lin
c49c7f0cf9 ASoC: Use dai_fmt in tavorevb3 machine driver
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Haojian Zhuang <haojian.zhuang@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02 13:08:39 +00:00
Axel Lin
385bd9379b ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out()
We can't just pass back the return value of snd_soc_update_bits() as it
will be 1 if a bit changed rather than zero.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02 13:08:39 +00:00
Axel Lin
3c3f51f6a3 ASoC: Convert z2 to table based DAPM init
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02 13:08:38 +00:00
Axel Lin
1a2dbcbe04 ASoC: Convert tavorevb3 to table based DAPM init
Also remove a unsued ret variable to silence the build warning.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-02 13:08:37 +00:00