openwrt/staging/blogic.git
13 years agoALSA: hda - Don't refer ELD when unplugged
Takashi Iwai [Tue, 11 Jan 2011 17:07:14 +0000 (18:07 +0100)]
ALSA: hda - Don't refer ELD when unplugged

When unplugged, we shouldn't refer to ELD information for PCM open
any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
13 years agoALSA: hda - Add support for multiple headphone/speaker controls for Realtek
Takashi Iwai [Mon, 10 Jan 2011 14:45:23 +0000 (15:45 +0100)]
ALSA: hda - Add support for multiple headphone/speaker controls for Realtek

So far, Realtek auto-parser assumed that the multiple pins are only for
line-outs, and assigned the channel names like Front, Surround, etc for
the multiple outputs.  But, there are devices that have multiple
headphones, and these can be better controlled with the corresponding
control-name like "Headphone" with indicies.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agoALSA: hda - Fix multi-headphone handling for Realtek codecs
Takashi Iwai [Mon, 10 Jan 2011 13:47:35 +0000 (14:47 +0100)]
ALSA: hda - Fix multi-headphone handling for Realtek codecs

When multiple headphone pins are defined without line-out pins, the
driver takes them as primary outputs.  But it forgot to set line_out_type
to HP by assuming there is some rest of HP pins.  This results in some
mis-handling of these pins for Realtek codec parser.  It takes as if
these are pure line-out jacks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
13 years agoALSA: hda: Use vostro model quirk for Dell Vostro 1014
Daniel T Chen [Sat, 8 Jan 2011 23:25:27 +0000 (18:25 -0500)]
ALSA: hda: Use vostro model quirk for Dell Vostro 1014

BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5184
A user reported on the alsa-devel mailing list that he needs to use
the vostro model quirk to have audible playback, so apply it for his
PCI SSID.

Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agoALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx
David Henningsson [Fri, 7 Jan 2011 06:53:39 +0000 (07:53 +0100)]
ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx

BugLink: http://bugs.launchpad.net/bugs/689036
Many new Lenovos need the ideapad quirk. Also, since the
auto parser for this chip is far from optimal, the regression
risk is low (although not zero).

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agoALSA: HDA: Fix volume control indices for Mics (Realtek)
David Henningsson [Wed, 5 Jan 2011 10:03:56 +0000 (11:03 +0100)]
ALSA: HDA: Fix volume control indices for Mics (Realtek)

If more than one mic is present with different locations,
e g "Front Mic" and "Rear Mic", they can use the same index (0),
since their names are different.

Previous behavior was to have "Front Mic" as index 1, causing it
to be ignored by e g PulseAudio.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agoALSA: HDA: Rename "Mic Boost" to "Mic Boost Volume"
David Henningsson [Tue, 4 Jan 2011 14:24:24 +0000 (15:24 +0100)]
ALSA: HDA: Rename "Mic Boost" to "Mic Boost Volume"

BugLink: http://bugs.launchpad.net/bugs/697240
If the "Volume" suffix is not given, alsa-lib gets confused and
loses the dB information at the simple element level.

Boosts generally affects both playback and capture, as they are
applied early in the chain. Hence no "Playback" or "Capture" in
the suffix.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agoALSA: HDA: Add internal mic for IDT 92HD88B
David Henningsson [Tue, 4 Jan 2011 05:30:53 +0000 (06:30 +0100)]
ALSA: HDA: Add internal mic for IDT 92HD88B

BugLink: http://bugs.launchpad.net/bugs/696493
According to datasheet (and real-world testing), IDT 92HD88B can
have internal mics at NID 0x11 and 0x20, so enable them accordingly.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Mon, 10 Jan 2011 09:49:13 +0000 (10:49 +0100)]
Merge branch 'fix/hda' into topic/hda

13 years agoALSA: hda: Use LPIB quirk for Dell Inspiron m101z/1120
Daniel T Chen [Tue, 28 Dec 2010 22:20:02 +0000 (17:20 -0500)]
ALSA: hda: Use LPIB quirk for Dell Inspiron m101z/1120

Sjoerd Simons reports that, without using position_fix=1, recording
experiences overruns. Work around that by applying the LPIB quirk
for his hardware.

Reported-and-tested-by: Sjoerd Simons <sjoerd@debian.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
13 years agosound: Prevent buffer overflow in OSS load_mixer_volumes
Dan Rosenberg [Sat, 25 Dec 2010 21:23:40 +0000 (16:23 -0500)]
sound: Prevent buffer overflow in OSS load_mixer_volumes

The load_mixer_volumes() function, which can be triggered by
unprivileged users via the SOUND_MIXER_SETLEVELS ioctl, is vulnerable to
a buffer overflow.  Because the provided "name" argument isn't
guaranteed to be NULL terminated at the expected 32 bytes, it's possible
to overflow past the end of the last element in the mixer_vols array.
Further exploitation can result in an arbitrary kernel write (via
subsequent calls to load_mixer_volumes()) leading to privilege
escalation, or arbitrary kernel reads via get_mixer_levels().  In
addition, the strcmp() may leak bytes beyond the mixer_vols array.

Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into for-linus
Takashi Iwai [Thu, 23 Dec 2010 15:37:31 +0000 (16:37 +0100)]
Merge branch 'fix/hda' into for-linus

14 years agoALSA: hda - Fix GPIO2-fixup for Sony laptops
Takashi Iwai [Thu, 23 Dec 2010 15:35:34 +0000 (16:35 +0100)]
ALSA: hda - Fix GPIO2-fixup for Sony laptops

The fix-up entries by the commit 2785591a9760c677a7ee6f541e751c23086f5bfd
     ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs
weren't applied in the right position.  They had to be before the quirk
entry matching to all Sony devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/misc' into for-linus
Takashi Iwai [Thu, 23 Dec 2010 09:28:26 +0000 (10:28 +0100)]
Merge branch 'fix/misc' into for-linus

14 years agoALSA: hda - Try to find an empty control index when it's occupied
Takashi Iwai [Thu, 23 Dec 2010 09:17:52 +0000 (10:17 +0100)]
ALSA: hda - Try to find an empty control index when it's occupied

When a mixer control element was already created with the given name,
try to find another index for avoiding conflicts, instead of breaking
with an error.  This makes the driver more robust.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Fix conflict of d-mic capture volume controls
Takashi Iwai [Thu, 23 Dec 2010 09:16:05 +0000 (10:16 +0100)]
ALSA: hda - Fix conflict of d-mic capture volume controls

When the d-mics are assigned to the same purpose of another analog mic
pins, the driver doesn't compute the index properly, resulting in an
error with "existing control".  This patch fixes it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Don't apply ALC269-specific initialization to ALC275
Kailang Yang [Tue, 21 Dec 2010 08:14:13 +0000 (09:14 +0100)]
ALSA: hda - Don't apply ALC269-specific initialization to ALC275

ALC275 doesn't require the ALC269 (and its variants) specific init
sequences.  Add the check of codec id.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs
Kailang Yang [Tue, 21 Dec 2010 08:09:53 +0000 (09:09 +0100)]
ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs

Set GPIO2 for some Sony VAIO with ALC275 to fix speaker output.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: pcm: remember to always call va_end() on stuff that we va_start()
Jesper Juhl [Mon, 20 Dec 2010 23:03:17 +0000 (00:03 +0100)]
ALSA: pcm: remember to always call va_end() on stuff that we va_start()

The Coverity checker spotted that we do not always remember to call
va_end() on 'args' in failure paths in snd_pcm_hw_rule_add().
Here's a patch to fix that up (compile tested only) - it also removes
some annoying trailing whitespace that caught my eye while I was in the
area..

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Rename "e-Mic" and "i-Mic" to "Mic" and "Internal Mic"
David Henningsson [Mon, 20 Dec 2010 13:50:59 +0000 (14:50 +0100)]
ALSA: HDA: Rename "e-Mic" and "i-Mic" to "Mic" and "Internal Mic"

Change non-standard mic control names to standard control names
to clean up the namespace.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Rename "Ext Mic" and "External Mic" to "Mic"
David Henningsson [Mon, 20 Dec 2010 13:43:54 +0000 (14:43 +0100)]
ALSA: HDA: Rename "Ext Mic" and "External Mic" to "Mic"

Usually external microphones are just labelled "Mic", so rename
"Ext Mic" and "External Mic" to "Mic" to clear up the namespace.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Rename "Int Mic" to "Internal Mic"
David Henningsson [Mon, 20 Dec 2010 13:24:29 +0000 (14:24 +0100)]
ALSA: HDA: Rename "Int Mic" to "Internal Mic"

"Int Mic" and "Internal Mic" both mean the same thing, so rename
the former to the latter in order to clean up the namespace a little.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Mon, 20 Dec 2010 09:28:51 +0000 (10:28 +0100)]
Merge branch 'fix/hda' into topic/hda

14 years agoALSA: HDA: Add auto-mute for Thinkpad SL410/SL510
David Henningsson [Fri, 17 Dec 2010 19:43:04 +0000 (20:43 +0100)]
ALSA: HDA: Add auto-mute for Thinkpad SL410/SL510

BugLink: http://launchpad.net/bugs/580006
SKU turns off auto-mute for these machines, so ignore the SKU.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/asoc' into for-linus
Takashi Iwai [Fri, 17 Dec 2010 14:28:37 +0000 (15:28 +0100)]
Merge branch 'fix/asoc' into for-linus

14 years agoMerge branch 'fix/hda' into for-linus
Takashi Iwai [Fri, 17 Dec 2010 14:28:33 +0000 (15:28 +0100)]
Merge branch 'fix/hda' into for-linus

14 years agoALSA: hda - Fix conflict of Mic Boot controls
Takashi Iwai [Fri, 17 Dec 2010 14:23:41 +0000 (15:23 +0100)]
ALSA: hda - Fix conflict of Mic Boot controls

Due to the recent change for multiple mics assignment, we need to handle
the index of each Mic Boost control respectively.  Otherwise the driver
gets the control element conflicts, and gives the unsable state.

Reference: kernel bug 25002
https://bugzilla.kernel.org/show_bug.cgi?id=25002

Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Clean up dead code in patch_realtek.c
Takashi Iwai [Thu, 16 Dec 2010 16:55:42 +0000 (17:55 +0100)]
ALSA: hda - Clean up dead code in patch_realtek.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - factorize an automute_mic realtek quirk function
Anisse Astier [Thu, 16 Dec 2010 11:19:47 +0000 (12:19 +0100)]
ALSA: hda - factorize an automute_mic realtek quirk function

Multiple quirk functions were using the exact same code to verify if the Mic
jack was plugged and mute the Mic accordingly

Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Enable subwoofer on Asus G73Jw
David Henningsson [Wed, 15 Dec 2010 08:18:18 +0000 (09:18 +0100)]
ALSA: HDA: Enable subwoofer on Asus G73Jw

Set default association/sequence right on pin 0x17 in order for
the automatic parser to recognize the subwoofer correctly.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Fix auto-mute on Lenovo Edge 14
David Henningsson [Wed, 15 Dec 2010 07:01:46 +0000 (08:01 +0100)]
ALSA: HDA: Fix auto-mute on Lenovo Edge 14

BugLink: http://launchpad.net/bugs/690530
The SKU value of this machine dictates that auto-mute should be
disabled. Since the SKU value is similar to the PCI SSID, the most
likely conclusion is that the SKU value should be ignored.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: Fix bias power down of non-DAPM codec
Jarkko Nikula [Fri, 10 Dec 2010 18:53:55 +0000 (20:53 +0200)]
ASoC: Fix bias power down of non-DAPM codec

Currently bias of non-DAPM codec will be powered down (standby/off) whenever
there is a stream stop. This is wrong in simultaneous playback/capture since
the bias is put down immediately after stopping the first stream.

Fix this by using the codec->active count when figuring out the needed bias
level after stream stop.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoALSA: hda - Mute speakers when line-out jack is plugged with Conexant auto mode
Takashi Iwai [Mon, 13 Dec 2010 11:48:35 +0000 (12:48 +0100)]
ALSA: hda - Mute speakers when line-out jack is plugged with Conexant auto mode

Mute speakers when a line-out jack is plugged as well as headphone jacks
with the new Conexant codec parser in the auto mode.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: WM8580: Fix R8 initial value
Seungwhan Youn [Thu, 9 Dec 2010 09:07:52 +0000 (18:07 +0900)]
ASoC: WM8580: Fix R8 initial value

Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c.

Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
14 years agoASoC: fix deemphasis control in wm8904/55/60 codecs
Dmitry Artamonow [Wed, 8 Dec 2010 20:36:17 +0000 (23:36 +0300)]
ASoC: fix deemphasis control in wm8904/55/60 codecs

Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.

Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
14 years agoMerge branch 'fix/asoc' into for-linus
Takashi Iwai [Thu, 9 Dec 2010 07:24:32 +0000 (08:24 +0100)]
Merge branch 'fix/asoc' into for-linus

14 years agoMerge branch 'fix/hda' into for-linus
Takashi Iwai [Thu, 9 Dec 2010 07:24:25 +0000 (08:24 +0100)]
Merge branch 'fix/hda' into for-linus

14 years agoALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
David Henningsson [Thu, 9 Dec 2010 06:17:27 +0000 (07:17 +0100)]
ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work

BugLink: http://launchpad.net/497546
Confirmed that the ideapad model works better than the current
quirk for Dell Vostro 320.

Cc: stable@kernel.org (2.6.35+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda: Add fixup for mario system
Todd Broch [Wed, 8 Dec 2010 00:51:05 +0000 (16:51 -0800)]
ALSA: hda: Add fixup for mario system

create fixup function for the mario model and override amp capabilities
for NID 0x2

Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda: Add modelname lookup and fixup for realtek codecs
Todd Broch [Mon, 6 Dec 2010 19:19:51 +0000 (11:19 -0800)]
ALSA: hda: Add modelname lookup and fixup for realtek codecs

Facilitate fixup for realtek codecs via modelname lookup of fixup
data.  Fallback to quirk based lookup in absence of model definition.

Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Remove unconnected PCM devices for Intel HDMI
David Henningsson [Tue, 23 Nov 2010 09:23:40 +0000 (10:23 +0100)]
ALSA: HDA: Remove unconnected PCM devices for Intel HDMI

Some newer chips have more than one HDMI output, but usually not
all of them are exposed as physical jacks. Removing the unused
PCM devices (as indicated by BIOS in the pin config default) will
reduce user confusion as they currently have to choose between
several HDMI devices, some of them not working anyway.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Wed, 8 Dec 2010 08:07:38 +0000 (09:07 +0100)]
Merge branch 'fix/hda' into topic/hda

14 years agoALSA: hda - Reset sample sizes and max bitrates when reading ELD
Anssi Hannula [Tue, 7 Dec 2010 19:19:23 +0000 (21:19 +0200)]
ALSA: hda - Reset sample sizes and max bitrates when reading ELD

When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.

The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.

The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.

Fix that by always clearing sample_bits and max_bitrate when reading
SADs.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Always allow basic audio irrespective of ELD info
Anssi Hannula [Tue, 7 Dec 2010 18:56:19 +0000 (20:56 +0200)]
ALSA: hda - Always allow basic audio irrespective of ELD info

Commit bbbe33900d1f3c added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.

However, according to CEA-861-D no SAD is needed for basic audio
(32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a
basic audio flag in the CEA EDID Extension.

The flag is not present in ELD. However, as all audio capable sinks are
required to support basic audio, we can assume it to be always
available.

Fix allowed audio formats with sinks that have SADs (Short Audio
Descriptors) which do not completely overlap with the basic audio
formats (there are no reports of affected devices so far) by always
assuming that basic audio is supported.

Reported-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Do not wrongly restrict min_channels based on ELD
Anssi Hannula [Tue, 7 Dec 2010 16:41:35 +0000 (18:41 +0200)]
ALSA: hda - Do not wrongly restrict min_channels based on ELD

Commit bbbe33900d1f3c added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.

However, it wrongly assumes that the bits 0-2 of the first byte of
CEA Short Audio Descriptors mean a supported number of channels. In
reality, they mean the maximum number of channels (as per CEA-861-D
7.5.2). This means that the channel count can only be used to restrict
max_channels, not min_channels.

Restricting min_channels causes us to deny opening the device in stereo
mode if the sink only has SADs that declare larger numbers of channels
(like Primare SP32 AV Processor does).

Fix that by not restricting min_channels based on ELD information.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Jean-Yves Avenard <jyavenard@gmail.com>
Tested-by: Jean-Yves Avenard <jyavenard@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: Correct WM8962 interrupt mask register read
Mark Brown [Tue, 7 Dec 2010 15:32:38 +0000 (15:32 +0000)]
ASoC: Correct WM8962 interrupt mask register read

Fix mismerge from the out of tree BSP where this support was developed.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: WM8580: Debug BCLK and sample size
Jassi Brar [Tue, 7 Dec 2010 10:23:07 +0000 (19:23 +0900)]
ASoC: WM8580: Debug BCLK and sample size

In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32,
rather than WM8580_AIF_LENGTH_24.
Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Fix resource leak if soc_register_ac97_dai_link failed
Axel Lin [Tue, 7 Dec 2010 12:56:30 +0000 (20:56 +0800)]
ASoC: Fix resource leak if soc_register_ac97_dai_link failed

Properly free the resources in the case of soc_register_ac97_dai_link failure.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Hold client_mutex while calling snd_soc_instantiate_cards()
Axel Lin [Mon, 6 Dec 2010 08:48:03 +0000 (16:48 +0800)]
ASoC: Hold client_mutex while calling snd_soc_instantiate_cards()

As the comments of snd_soc_instantiate_cards() said,
snd_soc_instantiate_cards() must be called with client_mutex.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain
Uk Kim [Sun, 5 Dec 2010 08:26:07 +0000 (17:26 +0900)]
ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain

SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit.

Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
14 years agoASoC: Fix off by one error in WM8994 EQ register bank size
Uk Kim [Sun, 5 Dec 2010 08:32:16 +0000 (17:32 +0900)]
ASoC: Fix off by one error in WM8994 EQ register bank size

Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
14 years agoALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic
Daniel T Chen [Sun, 5 Dec 2010 13:43:14 +0000 (08:43 -0500)]
ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic

BugLink: https://launchpad.net/bugs/685161
The reporter of the bug states that he must use position_fix=1 to enable
capture for the internal microphone, so set it for his machine's PCI
SSID.  Verified using 2.6.35 and the 2010-12-04 alsa-driver build.

Reported-and-tested-by: Ralph Wabel <rwabel@gmx.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - use generic hdmi parser for ATI R6xx codec
Anssi Hannula [Sun, 5 Dec 2010 00:34:15 +0000 (02:34 +0200)]
ALSA: hda - use generic hdmi parser for ATI R6xx codec

Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx
HDMI), as the codec appears to work fine with it.

Note that the codec is still limited to stereo output only, despite it
reportedly being multichannel capable. Some as of yet unknown quirks
will be needed to get that working.

Testing was done on 2.6.36 by John Ettedgui.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: John Ettedgui <john.ettedgui@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Enable jack sense for Thinkpad Edge 13
Manoj Iyer [Sat, 4 Dec 2010 00:43:55 +0000 (18:43 -0600)]
ALSA: hda - Enable jack sense for Thinkpad Edge 13

Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13.

Reference: http://launchpad.net/bugs/685015

Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Fix beep-tone on IDT 92HD87/88 codecs
Takashi Iwai [Fri, 3 Dec 2010 14:19:14 +0000 (15:19 +0100)]
ALSA: hda - Fix beep-tone on IDT 92HD87/88 codecs

It sounds like a non-linear beep tone on my test machines...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Enable beep for IDT92HD87 / 88 codecs
Takashi Iwai [Fri, 3 Dec 2010 13:58:37 +0000 (14:58 +0100)]
ALSA: hda - Enable beep for IDT92HD87 / 88 codecs

These codecs have the digital beep widget in NID 0x21.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Clean up cxt5066 port-D handling & co
Takashi Iwai [Fri, 3 Dec 2010 11:29:14 +0000 (12:29 +0100)]
ALSA: hda - Clean up cxt5066 port-D handling & co

Instead of hard-coded magic numbers, properly define and use macros
for improve the readability.  Also, dell_automute is handled samely
as thinkpad, since it also sets port_d_mode, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Fri, 3 Dec 2010 11:27:47 +0000 (12:27 +0100)]
Merge branch 'fix/hda' into topic/hda

14 years agoALSA: hda - Fix ThinkPad T410[s] docking station line-out
John Baboval [Thu, 2 Dec 2010 16:21:31 +0000 (11:21 -0500)]
ALSA: hda - Fix ThinkPad T410[s] docking station line-out

On the docking station for the Lenovo T410 and T410s, the line-out
doesn't work. The trouble seems to be that it generates a plug event,
but then doesn't report that the jack is connected. So automute mutes
the jack when you plug something into it. The following patch (next
message) fixes it.

Signed-off-by: John Baboval <john.baboval at virtualcomputer.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture
Daniel T Chen [Fri, 3 Dec 2010 03:45:45 +0000 (22:45 -0500)]
ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture

BugLink: https://launchpad.net/bugs/595482
The original reporter states that audible playback from the internal
speaker is inaudible despite the hardware being properly detected.  To
work around this symptom, he uses the model=lg quirk to properly enable
both playback, capture, and jack sense.  Another user corroborates this
workaround on separate hardware.  Add this PCI SSID to the quirk table
to enable it for further LG P1 Expresses.

Reported-and-tested-by: Philip Peitsch <philip.peitsch@gmail.com>
Tested-by: nikhov
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/asoc' into for-linus
Takashi Iwai [Thu, 2 Dec 2010 16:33:53 +0000 (17:33 +0100)]
Merge branch 'fix/asoc' into for-linus

14 years agoMerge branch 'for-2.6.37' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc...
Takashi Iwai [Thu, 2 Dec 2010 16:31:18 +0000 (17:31 +0100)]
Merge branch 'for-2.6.37' of git://git./linux/kernel/git/lrg/asoc-2.6 into fix/asoc

14 years agoMerge branch 'fix/hda' into for-linus
Takashi Iwai [Thu, 2 Dec 2010 16:14:50 +0000 (17:14 +0100)]
Merge branch 'fix/hda' into for-linus

14 years agoASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
Jarkko Nikula [Wed, 1 Dec 2010 09:01:20 +0000 (11:01 +0200)]
ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency

Not all omap boards use kernel based pin multiplexing so
CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make
harm to other boards in multi-board kernels.

Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver.

Thanks to Tony Lindgren <tony@atomide.com> for noticing.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoALSA: hda: Use "alienware" model quirk for another SSID
Daniel T Chen [Thu, 2 Dec 2010 00:16:07 +0000 (19:16 -0500)]
ALSA: hda: Use "alienware" model quirk for another SSID

BugLink: https://launchpad.net/bugs/683695
The original reporter states that headphone jacks do not appear to
work.  Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.

Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: WM8731: Fix incorrect mask for bypass path disable
Dimitris Papastamos [Wed, 1 Dec 2010 09:38:55 +0000 (09:38 +0000)]
ASoC: WM8731: Fix incorrect mask for bypass path disable

According to the datasheet the bypass path enable/disable is
bit 3 therefore we need 0x8 and not 0x4.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agos6105-ipcam: fix compilation
Daniel Glöckner [Tue, 30 Nov 2010 00:00:18 +0000 (01:00 +0100)]
s6105-ipcam: fix compilation

When the s6105-ipcam ASoC driver had been converted to the
multi-component API, a single reference to a former structure
element remained, blocking successful compilation.

Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agos6000-pcm: fix compilation
Daniel Glöckner [Tue, 30 Nov 2010 00:00:17 +0000 (01:00 +0100)]
s6000-pcm: fix compilation

s6000_soc_platform has lost its forward declaration and there no
longer is a name element in it, so use a string constant when
calling request_irq.

Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agos6000-i2s: fix compilation
Daniel Glöckner [Tue, 30 Nov 2010 00:00:16 +0000 (01:00 +0100)]
s6000-i2s: fix compilation

A semicolon was missing.

Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Fix missing spin_unlock_irqrestore
Axel Lin [Mon, 29 Nov 2010 09:42:47 +0000 (17:42 +0800)]
ASoC: Fix missing spin_unlock_irqrestore

In nuc900_dma_hw_params(), if snd_pcm_lib_malloc_pages failed
it returns without calling spin_unlock_irqrestore().

Since snd_pcm_lib_malloc_pages() does not touch struct nuc900_audio,
we don't need to hold the lock while calling snd_pcm_lib_malloc_pages().
Fix it by moving spin_lock_irqsave() down to after snd_pcm_lib_malloc_pages().

In nuc900_dma_prepare(), spin_unlock_irqrestore() is missing in the error path.
Fix it by removing the return in default case.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation
Takashi Iwai [Tue, 30 Nov 2010 07:14:21 +0000 (08:14 +0100)]
ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation

In OSS emulation, SNDCTL_DSP_RESET ioctl needs the reset of the internal
buffer state in addition to drop of the running streams.  Otherwise the
succeeding access becomes inconsistent.

Tested-by: Amit Nagal <helloin.amit@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: Add missing dev_set_drvdata in p1022_ds_probe
Axel Lin [Mon, 29 Nov 2010 06:55:58 +0000 (14:55 +0800)]
ASoC: Add missing dev_set_drvdata in p1022_ds_probe

Otherwise, calling dev_get_drvdata in p1022_ds_remove returns NULL.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
Axel Lin [Mon, 29 Nov 2010 06:54:58 +0000 (14:54 +0800)]
ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe

Otherwise, calling dev_get_drvdata in mpc8610_hpcd_remove returns NULL.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Remove unneeded !! operations while checking return value of nuc900_checkready
Axel Lin [Mon, 29 Nov 2010 09:43:39 +0000 (17:43 +0800)]
ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready

I think this unneededd !! operations just reduce the readability.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Fix compile error for nuc900-pcm.c
Axel Lin [Mon, 29 Nov 2010 09:40:53 +0000 (17:40 +0800)]
ASoC: Fix compile error for nuc900-pcm.c

This patch fixes below error:

  CC      sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_open':
sound/soc/nuc900/nuc900-pcm.c:267: error: 'nuc900_ac97_data' undeclared (first use in this function)
sound/soc/nuc900/nuc900-pcm.c:267: error: (Each undeclared identifier is reported only once
sound/soc/nuc900/nuc900-pcm.c:267: error: for each function it appears in.)
sound/soc/nuc900/nuc900-pcm.c: At top level:
sound/soc/nuc900/nuc900-pcm.c:337: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-pcm.c:354: error: 'nuc900_soc_platform_probe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-pcm.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove
Axel Lin [Mon, 29 Nov 2010 09:40:05 +0000 (17:40 +0800)]
ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove

This patch fixes below compile warning:

  CC      sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Fix compile error for nuc900-ac97.c
Axel Lin [Mon, 29 Nov 2010 09:39:10 +0000 (17:39 +0800)]
ASoC: Fix compile error for nuc900-ac97.c

Fix below compile error by add a missing ';'.

  CC      sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:318: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-ac97.c:405: error: 'nuc900_ac97_drvprobe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-ac97.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoALSA: hda - Remove dead md2 quirk code
Takashi Iwai [Mon, 29 Nov 2010 06:42:59 +0000 (07:42 +0100)]
ALSA: hda - Remove dead md2 quirk code

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Mon, 29 Nov 2010 06:44:01 +0000 (07:44 +0100)]
Merge branch 'fix/hda' into topic/hda

14 years agoALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2
Daniel T Chen [Sat, 27 Nov 2010 18:58:04 +0000 (13:58 -0500)]
ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2

BugLink: https://launchpad.net/bugs/682199
A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c04 of sound-2.6.git, a new model was added with this
machine's PCI SSID.  Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.

Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.

Reported-and-tested-by: burningphantom1
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/asoc' into for-linus
Takashi Iwai [Fri, 26 Nov 2010 16:17:42 +0000 (17:17 +0100)]
Merge branch 'fix/asoc' into for-linus

14 years agoALSA: hda - Use ALC_INIT_DEFAULT for really default initialization
Takashi Iwai [Fri, 26 Nov 2010 16:11:18 +0000 (17:11 +0100)]
ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization

When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far.  This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.

Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.

This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.

Reference:
https://bugzilla.redhat.com/show_bug.cgi?id=657388

Reported-and-tested-by: Kyle McMartin <kyle@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: Fix resource reclaim for osk5912
Axel Lin [Wed, 24 Nov 2010 14:24:01 +0000 (22:24 +0800)]
ASoC: Fix resource reclaim for osk5912

In current implementation, there are resources leak in the error path.
This patch properly reclaims the allocated resources in the error path.

Also adds a missing clk_put in osk_soc_exit.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: tlv320aic3x - fix variable may be used uninitialized warning
Axel Lin [Wed, 24 Nov 2010 14:40:59 +0000 (22:40 +0800)]
ASoC: tlv320aic3x - fix variable may be used uninitialized warning

If aic3x_read failed , val is used uninitialized.
Fix it by initializing val to 0.

This patch fixes below compile warning:
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_get_gpio':
sound/soc/codecs/tlv320aic3x.c:1183: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_headset_detected':
sound/soc/codecs/tlv320aic3x.c:1211: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_button_pressed':
sound/soc/codecs/tlv320aic3x.c:1219: warning: 'val' may be used uninitialized in this function

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: davinci-vcif - fix a memory leak
Axel Lin [Thu, 25 Nov 2010 03:33:14 +0000 (11:33 +0800)]
ASoC: davinci-vcif - fix a memory leak

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: phycore-ac97: fix resource leak
Axel Lin [Thu, 25 Nov 2010 07:14:03 +0000 (15:14 +0800)]
ASoC: phycore-ac97: fix resource leak

Fix imx_phycore_init() error path and imx_phycore_exit() to properly free
allocated resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: imx-ssi: fix resource leak
Axel Lin [Thu, 25 Nov 2010 07:13:09 +0000 (15:13 +0800)]
ASoC: imx-ssi: fix resource leak

Fix imx_ssi_probe() error path and imx_ssi_remove() to properly free
allocated resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: simone: fix resource leak in simone_init error path
Axel Lin [Thu, 25 Nov 2010 07:12:30 +0000 (15:12 +0800)]
ASoC: simone: fix resource leak in simone_init error path

Fix the error path to properly free allocated resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path
Axel Lin [Thu, 25 Nov 2010 07:11:03 +0000 (15:11 +0800)]
ASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path

Fix the error path to properly free allocated resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path
Axel Lin [Thu, 25 Nov 2010 02:44:59 +0000 (10:44 +0800)]
ASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path
Axel Lin [Thu, 25 Nov 2010 07:08:31 +0000 (15:08 +0800)]
ASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path

Add missing platform_device_put() if platform_device_add() failed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path
Axel Lin [Thu, 25 Nov 2010 07:07:25 +0000 (15:07 +0800)]
ASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path

Add missing platform_device_put() if platform_device_add() failed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_r...
Axel Lin [Thu, 25 Nov 2010 09:23:55 +0000 (17:23 +0800)]
ASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_remove

We call snd_soc_register_dais() in sh4_soc_dai_probe(),
thus we should call snd_soc_unregister_dais() in sh4_soc_dai_remove().

Otherwise, we got "too many arguments to function 'snd_soc_unregister_dai'"
error message.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: fix SND_PXA2XX_LIB Kconfig warning
Dmitry Artamonow [Wed, 24 Nov 2010 21:46:15 +0000 (00:46 +0300)]
ASoC: fix SND_PXA2XX_LIB Kconfig warning

Fix following warning observed when SND_PXA2XX_SOC is set and SND_ARM isn't:

warning: (SND_PXA2XX_AC97 && SOUND && !M68K && SND && SND_ARM && ARCH_PXA ||
SND_PXA2XX_SOC && SOUND && !M68K && SND && SND_SOC && ARCH_PXA) selects
SND_PXA2XX_LIB which has unmet direct dependencies (SOUND && !M68K && SND &&
SND_ARM)

Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoALSA: hda - Fix ALC660-VD/ALC861-VD capture/playback mixers
Herton Ronaldo Krzesinski [Thu, 25 Nov 2010 02:08:01 +0000 (00:08 -0200)]
ALSA: hda - Fix ALC660-VD/ALC861-VD capture/playback mixers

The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).

Reference: https://qa.mandriva.com/show_bug.cgi?id=61159

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: HDA: Add an extra DAC for Realtek ALC887-VD
David Henningsson [Wed, 24 Nov 2010 13:17:47 +0000 (14:17 +0100)]
ALSA: HDA: Add an extra DAC for Realtek ALC887-VD

The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: nuc900-ac97: fix a memory leak
Axel Lin [Wed, 24 Nov 2010 08:44:23 +0000 (16:44 +0800)]
ASoC: nuc900-ac97: fix a memory leak

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoMerge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound...
Mark Brown [Wed, 24 Nov 2010 11:22:55 +0000 (11:22 +0000)]
Merge branch 'fix/asoc' of git://git./linux/kernel/git/tiwai/sound-2.6 into for-2.6.37

14 years agoASoC: Return proper error for omap3pandora_soc_init
Axel Lin [Wed, 24 Nov 2010 07:20:48 +0000 (15:20 +0800)]
ASoC: Return proper error for omap3pandora_soc_init

Return PTR_ERR(omap3pandora_dac_reg) instead of 0 if regulator_get failed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: wm8961 - clear WM8961_MCLKDIV bit for freq <= 16500000
Axel Lin [Wed, 24 Nov 2010 02:21:54 +0000 (10:21 +0800)]
ASoC: wm8961 - clear WM8961_MCLKDIV bit for freq <= 16500000

MCLKDIV bit of Register 04h Clocking1:
0 : Divide by 1
1 : Divide by 2

Thus in the case of freq <= 16500000, we should clear MCLKDIV bit.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org