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Commit Graph

262804 Commits

Author SHA1 Message Date
William Light
3d37fbe441 ALSA: snd-usb-caiaq: Fix NULL dereference in input.c
There was a case where a newly-registered input device could be opened before
a necessary variable in the device structure was set. When code tried to use
the variable in the URB reply callback, it would cause an Oops.

This fix sets the aforementioned variable before calling input_register_device.

Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-13 08:16:42 +02:00
Feng Tang
ffd3d5c6c7 ALSA: pcm - remove the dead code from snd_pcm_open_file()
The rpcm_file parameter is never used in current ALSA code, so remove
it to make it cleaner.

Signed-off-by: Feng Tang <feng.tang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-10 12:53:53 +02:00
Clemens Ladisch
8d448162bd ALSA: control: add support for ENUMERATED user space controls
Handling of user control elements was implemented for all types except
ENUMERATED.  This type will be needed for the device-specific mixers of
upcoming FireWire drivers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-09 09:09:11 +02:00
Dan Carpenter
f92766bc89 ALSA: oss-mixer - use strlcpy() instead strcpy()
This is mostly a static checker fix more than anything else.  We're
copying from a 64 char buffer into a 44 char buffer.

The 64 character buffer is str[] in snd_mixer_oss_build_test_all().
The call tree is:
	snd_mixer_oss_build_test_all()
	-> snd_mixer_oss_build_test()
	   -> snd_mixer_oss_build_test().

We never actually do fill str[] buffer all the way to 64 characters.
The longest string is:
	sprintf(str, "%s Playback Switch", ptr->name);
ptr->name is a 32 character buffer so 32 plus 16 characters for
" Playback Switch" still puts us over the 44 limit from "id.name".

Most likely ptr->name never gets filled to the limit, but we can't
really change the size of that buffer so lets just use strlcpy() here
and be safe.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-04 08:47:52 +02:00
Stefan Richter
a0978e8039 ALSA: firewire-speakers: fix locking
There is a lock inversion between fwspk->mutex and pcm->open_mutex
reported by lockdep when fwspk_hw_free is called.

Fixed by copying the fix from the same former issue in the isight
sound driver (commit f3f7c1837f
"ALSA: isight: fix locking").

Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-04 07:16:31 +02:00
Dan Carpenter
bb690c9e27 sound: oss: use strlcpy() in sound_timer_init()
sound_timer.info.name is a 32 character buffer.  This function only
has one caller (in sound/oss/ad1848.c) and it passes as 128 character
buffer as "name".  I don't know if this is a problem in real life,
and I doubt we're going to add more OSS drivers so it's unlikely to
become an issue.  But we may as well take care of it.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-29 08:12:33 +02:00
Clemens Ladisch
17d900c4a1 ALSA: usb-audio: increase control transfer timeout
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers.  Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.

The timeout parameter of snd_usb_ctl_msg() is now constant, so we can
drop it.

Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-27 09:21:48 +02:00
Takashi Iwai
6b69a0e520 ALSA: aloop - Use vmalloc buffer
snd-aloop driver is virtual and has no need for allocating contiguous
pages.  It'll be more system-friendly to use vmalloc buffers.

Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-24 12:16:29 +02:00
Andy Shevchenko
49957f3966 ALSA: 6fire: don't use custom hex_to_bin()
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 15:18:52 +02:00
Dan Carpenter
2ca595ab7a ALSA: hdspm - cleanup __user tags in ioctl()
This makes the code cleaner and silences a Sparse complaint:
sound/pci/rme9652/hdspm.c:6341:23: warning: incorrect type in assignment (incompatible argument 4 (different address spaces))
sound/pci/rme9652/hdspm.c:6341:23:    expected int ( *ioctl )( ... )
sound/pci/rme9652/hdspm.c:6341:23:    got int ( static [toplevel] *<noident> )( ... )
sound/pci/rme9652/hdspm.c:6102:44: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6225:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6264:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6283:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6289:59: warning: dereference of noderef expression

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 08:29:08 +02:00
Dan Carpenter
643d6bbb96 ALSA: hdspm - potential info leak in snd_hdspm_hwdep_ioctl()
Smatch has a new check for Rosenberg type information leaks where
structs are copied to the user with uninitialized stack data in them.

The status struct has a hole in it, and on some paths not all the
members were initialized.

struct hdspm_status {
        unsigned char              card_type;            /*     0     1 */
        /* XXX 3 bytes hole, try to pack */
        enum hdspm_syncsource      autosync_source;      /*     4     4 */
        long long unsigned int     card_clock;           /*     8     8 */

The hdspm_version struct had holes in it as well.

struct hdspm_version {
        unsigned char              card_type;            /*     0     1 */
        char                       cardname[20];         /*     1    20 */
        /* XXX 3 bytes hole, try to pack */
        unsigned int               serial;               /*    24     4 */
        short unsigned int         firmware_rev;         /*    28     2 */
        /* XXX 2 bytes hole, try to pack */
        int                        addons;               /*    32     4 */

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 08:28:56 +02:00
Takashi Iwai
8e699d2cc2 ALSA: fm801 - Clean up redundant reference to snd_fm801_tea575x_gpios[]
Use macro to improve readability.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 07:18:22 +02:00
Takashi Iwai
272a487056 Merge branch 'fix/misc' into topic/misc 2011-09-22 16:41:52 +02:00
Ben Hutchings
c37279b92a ALSA: fm801: Gracefully handle failure of tuner auto-detect
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.

As a bonus, this should fix init and cleanup for the case where the
tuner is explicitly disabled.

Reported-and-tested-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-22 15:52:52 +02:00
Ben Hutchings
2ba34e43ba ALSA: fm801: Fix double free in case of error in tuner detection
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") added
incorrect error handling.

Once we have successfully called snd_device_new(), the cleanup
function fm801_free() will automatically be called by snd_card_free()
and we must *not* also call fm801_free() directly.

Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-22 15:51:46 +02:00
David Henningsson
46724c2e02 ALSA: HDA: Add support for IDT 92HD93
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 09:13:18 +02:00
Clemens Ladisch
5495ffbd7b ALSA: via82xx: allow to disable the SRC
Add the PCM rule to allow disabling the PCM playback SRC.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:57:00 +02:00
Clemens Ladisch
57e5c63007 ALSA: emu10k1: allow to disable the SRC
Add the PCM rule to allow disabling the PCM playback SRC.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:56:58 +02:00
Clemens Ladisch
5b0416a3c2 ALSA: ymfpci: allow to disable the SRC
Add the PCM rules to allow disabling the PCM playback and capture SRCs.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:56:56 +02:00
Clemens Ladisch
d5b702a64b ALSA: pcm: add snd_pcm_hw_rule_noresample()
Add a helper function to allow drivers to disable hardware resampling
when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE
flag.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:56:45 +02:00
Clemens Ladisch
84f9df159d ALSA: ymfpci: fix PCM open error handling
The installation of the minimum period size constraint in the PCM open
callbacks was not checked for errors.  Add this check, and move the call
to the beginning of the function to avoid having to do any cleanups in
the error case.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:56:34 +02:00
Takashi Iwai
8974bd51a7 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration
When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work.  It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.

The patch fixes the problem and add a comment to indicate the
relationship briefly.

BugLink: http://bugs.launchpad.net/bugs/851697

Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Cc: stable@kernel.org (3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-19 11:31:34 +02:00
Daniele Guerrieri
14515a0829 ALSA: usb-audio: Added support for Roland UM-ONE midi-usb interface
Roland UM-ONE midi usb interface differs from Roland UM-1.

Signed-off-by: Daniele Guerrieri <d.guerrieri@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-16 08:31:45 +02:00
Takashi Iwai
0308110615 Merge branch 'fix/misc' into topic/misc 2011-09-16 08:29:04 +02:00
Arjan van de Ven
763437a9e7 ALSA: pcm - fix race condition in wait_for_avail()
wait_for_avail() in pcm_lib.c has a race in it (observed in practice by an
Intel validation group).

The function is supposed to return once space in the buffer has become
available, or if some timeout happens.  The entity that creates space (irq
handler of sound driver and some such) will do a wake up on a waitqueue
that this function registers for.

However there are two races in the existing code

1) If space became available between the caller noticing there was no
   space and this function actually sleeping, the wakeup is missed and the
   timeout condition will happen instead

2) If a wakeup happened but not sufficient space became available, the
   code will loop again and wait for more space.  However, if the second
   wake comes in prior to hitting the schedule_timeout_interruptible(), it
   will be missed, and potentially you'll wait out until the timeout
   happens.

The fix consists of using more careful setting of the current state (so
that if a wakeup happens in the main loop window, the schedule_timeout()
falls through) and by checking for available space prior to going into the
schedule_timeout() loop, but after being on the waitqueue and having the
state set to interruptible.

[tiwai: the following changes have been added to Arjan's original patch:
 - merged akpm's fix for waitqueue adding order into a single patch
 - reduction of duplicated code of avail check
]

Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-15 09:03:16 +02:00
Takashi Iwai
4038a12e74 Merge branch 'fix/asoc' into for-linus 2011-09-14 19:11:13 +02:00
Daniel Mack
c731bc96ad ALSA: snd-usb: move code from urb.c to endpoint.c
No code altered at this point, simply preparing for upcoming
refactorizations.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:03 +02:00
Daniel Mack
e8e8babf56 ALSA: snd-usb: re-order code
Move code from endpoint.c into a new file called stream.c and rename
functions so that their names actually reflect what they're doing.

This way, endpoint.c will be available to functions that hold all the
endpoint logic.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:02 +02:00
Daniel Mack
358e2bd4a9 ALSA: snd-usb: re-order the Makefile
Sort its entries in alphabetical order.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:01 +02:00
Daniel Mack
00137425fe USB: Add endpoint usage definitions to ch9.h
The endpoint usage field is described in the USB 2.0 specification,
chapter 9.6.6.

Also, move the sync type fields block down by some lines to reflect the
fact that these are also stuffed in bmAttributes.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:06:47 +02:00
David Henningsson
2e1210bc3d ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 13:45:12 +02:00
Clemens Ladisch
dba8b46992 ALSA: mpu401: clean up interrupt specification
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive:  To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero.  At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller.  This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.

With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.

This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter.  As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 11:00:51 +02:00
Takashi Iwai
99e14c9d41 ALSA: hda - Terminate the recursive connection search properly
The recursive search of widget connections in snd_hda_get_conn_index()
must be terminated at the pin and the audio-out widgets.  Otherwise
you'll get "too deep connection" warnings unnecessarily.

Reported-by: Francis Moreau <francis.moro@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-13 10:33:16 +02:00
Arnd Bergmann
5013951be8 ASoC: Fix trivial build regression in Kirkwood I2S
A fix merged in 3.1-rc2 introduced a small regression, this should get it
to build again.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-12 11:48:12 +01:00
Axel Lin
47124373b5 ALSA: keywest: Remove obsolete cleanup for clientdata
The i2c core will clear the clientdata pointer automatically.
We don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:36:12 +02:00
Axel Lin
5758960353 ALSA: aoa: Remove obsolete cleanup for clientdata
The i2c core will clear the clientdata pointer automatically.
We don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.

Also remove a unneeded NULL checking for kfree.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:35:47 +02:00
Raymond Yau
89f3325a6e ALSA: ymfpci: add "Playback" to FM Legacy Volume control
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:30:28 +02:00
Pierre-Louis Bossart
294c4fb8ab ALSA: usb: refine delay information with USB frame counter
Existing code only updates the audio delay when URBs were
submitted/retired. This can introduce an uncertainty of 8ms
on the number of samples played out with the default settings,
and a lot more when URBs convey more packets to reduce the
interrupt rate and power consumption.

This patch relies on the USB frame counter to reduce the
uncertainty to less than 2ms worst-case. The delay information
essentially becomes independent of the URB size and number of
packets. This should help applications like PulseAudio which
require accurate audio timing. Clemens Ladisch reported
a decrease of mplayer's A-V difference from nrpacks down to at
most 1ms.

Thanks to Clemens for also pointing out that the implementation
of frame counters varies between different HCDs. Only the
8 lowest-bits are used to estimate the delay.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:30:20 +02:00
Lars-Peter Clausen
c5d2e650bd ASoC: Blackfin: bf5xx-ad193x: Fix codec device name
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-09-05 18:11:29 -07:00
Mark Brown
747da0f80e ASoC: Fix reporting of partial jack updates
We need to report the entire jack state to the core jack code, not just
the bits that were being updated by the caller, otherwise the status
reported by other detection methods will be omitted from the state seen
by userspace.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-09-05 18:10:52 -07:00
Fabio Estevam
117ef9570b ASoC: imx: Fix build warning of unused 'card' variable
Fixes the following warning:

  CC      sound/soc/imx/imx-pcm-fiq.o
sound/soc/imx/imx-pcm-fiq.c: In function 'imx_pcm_fiq_new':
sound/soc/imx/imx-pcm-fiq.c:243: warning: unused variable 'card'
  CC      sound/soc/imx/imx-pcm-dma-mx2.o

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:48:13 +01:00
Lars-Peter Clausen
6c5b756aaa ASoC: Fix register cache sync register_writable WARN_ONs
Currently the condition for these WARN_ONs is reversed and they are placed
before the actual check whether we are going to write to that register. So if
the codec implements the register_writable callback we'll get a warning for each
writable register when syncing the register cache.

While we are at it change the check to use snd_soc_codec_writable_register
instead of open-coding it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:46:16 +01:00
Lars-Peter Clausen
63fa0a288c ASoC: snd_soc_codec_{readable,writable}_register change default to true
Change the default return value of snd_soc_codec_{readable,writable}_register to
true when no codec specific callback for this function is given. Otherwise all
registers of that codec will neither be readable nor writable, which is most
certainly not what we want.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:46:01 +01:00
Peter Ujfalusi
728a522224 ASoC: soc-dapm: Fix parameter comment for snd_soc_dapm_free
We have dapm_context instead of codec parameter.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:45:33 +01:00
Mark Brown
59ec6da2e3 MAINTAINERS: Add some missed Wolfson files
Mostly input related.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-29 14:00:26 +01:00
Kristian Amlie
1ef0e0a053 ALSA: usb-audio: add Starr Labs USB MIDI support
Add support for Starr Labs USB MIDI devices such as the Z7S, which are
based on an FTDI serial UART chip.

Based on a patch by Daniel Mack.

Signed-off-by: Kristian Amlie <kristian@amlie.name>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-26 14:12:34 +02:00
Takashi Iwai
26b9b559ed Merge branch 'fix/asoc' into for-linus 2011-08-26 09:29:43 +02:00
David Henningsson
468c545885 ALSA: hda: Conexant: Allow different output types to share DAC
Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.

BugLink: http://bugs.launchpad.net/bugs/817943
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-25 15:08:03 +02:00
Timur Tabi
3bdf28feaf ASoC: MPC5200: replace of_device with platform_device
'struct of_device' no longer exists, and its functionality has been merged
into platform_device.  Update the MPC5200 audio DMA driver (mpc5200_dma)
accordingly.  This fixes a build break.

Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-08-24 20:22:05 +01:00
Mark Brown
18036b5866 ASoC: Correct element count for WM8996 sidetone HPF
I can count. Honest.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-08-24 17:36:12 +01:00