Commit Graph

13738 Commits

Author SHA1 Message Date
Mark Brown 50fcfe45d7 ASoC: arizona: Record FLL setting when disabling
Otherwise we skip reenables.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-28 13:42:39 +00:00
Mark Brown 1cbe4bcae3 ASoC: arizona: Suppress noop FLL updates
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-27 19:30:56 +00:00
Mark Brown ba6b047ab9 ASoC: wm5102: Add missing routes for ASRC inputs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-27 19:30:25 +00:00
Mark Brown 38113360f0 ASoC: arizona: Support higher clock rates
Some devices support higher clock rates, allow users to select these.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-27 10:36:34 +00:00
Dimitris Papastamos 4cbc365509 ASoC: wm5102: Register DSP1 Aux widgets
It seems WM_ADSP2("DSP1", 0) is added twice to the widgets list, remove
that and in place use ARIZONA_DSP_WIDGETS(DSP1, "DSP1").

We need to make sure that the DSP1 Aux widgets are provided otherwise
we'll see errors such as "Failed to add route DSP1 Aux 1 -> DSP1" etc.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-21 10:32:00 +09:00
Mark Brown 9ce6565faa ASoC: wm5102: Remove output OSR and PGA volume control
These are managed automatically in current revisions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-20 17:34:31 +09:00
Mark Brown 24a118de11 Linux 3.7-rc6
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Merge tag 'v3.7-rc6' into asoc-arizona

Linux 3.7-rc6

Conflicts:
	sound/soc/codecs/wm5102.c
2012-11-19 15:43:50 +09:00
Takashi Iwai 10e44239f6 ALSA: usb-audio: Fix mutex deadlock at disconnection
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again.  There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:

A. snd_usb_audio_disconnect() ->
     card.c::register_mutex ->
       chip->shutdown_rwsem (write) ->
         snd_card_disconnect() ->
           pcm.c::register_mutex ->
             pcm->open_mutex

B. snd_pcm_open() ->
     pcm->open_mutex ->
       snd_usb_pcm_open() ->
         chip->shutdown_rwsem (read)

Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().

Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-14 15:29:09 +01:00
Dan Carpenter effded75e2 ALSA: fm801: precedence bug in snd_fm801_tea575x_get_pins()
There is a precedence bug because | has higher precedence than ?:.  This
code was cut and pasted and I fixed a similar bug a few days ago.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-14 09:34:28 +01:00
Charles Keepax 5886c74379 ASoC: wm5110: Remove mixer widgets on the ASRC
There is no mixer attached to the ASRC on the wm5110 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5110 CODEC driver.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-14 10:32:29 +09:00
Charles Keepax 0f3ec6a935 ASoC: wm5102: Remove mixer widgets on the ASRC
There is no mixer attached to the ASRC on the wm5102 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5102 CODEC driver.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-14 10:32:29 +09:00
Charles Keepax 17bd09e545 ASoC: arizona: Add support for multiplexer with no associated mixer
The Asynchronous Sample Rate Converters on the wm5102/wm5110 have no
mixer attached to their input, but they do allow the input to be
selected from a number of sources via a multiplexer. Currently the
platform assumes the presence of 4 multiplexers and a mixer for each
block.

This patch adds support multiplexed single input blocks into the Arizona
platform.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-14 10:32:29 +09:00
Dan Carpenter d2153a1595 ALSA: es1968: precedence bug in snd_es1968_tea575x_get_pins()
I don't think this works as intended.  '|' higher precedence than ?: so
the bitwize OR "0 | (val & STR_MOST)" is a no-op.

I have re-written it to be more clear.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-13 08:51:47 +01:00
Takashi Iwai 6214b54cbf ASoC: Fixes for v3.7
A few small fixes plus a large but simple change for WM5102 which writes
 out a bunch of register updates to the device when we enable the clock
 as recommended following chip evaluation.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.7

A few small fixes plus a large but simple change for WM5102 which writes
out a bunch of register updates to the device when we enable the clock
as recommended following chip evaluation.
2012-11-13 07:48:07 +01:00
Mark Brown ba027da8eb Merge branches 'fix/arizona', 'fix/core', 'fix/cs42l52', 'fix/mxs', 'fix/samsung' and 'fix/wm8978' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into tmp 2012-11-13 15:13:29 +09:00
Takashi Iwai 05193639ca ALSA: hda - Add a missing quirk entry for iMac 9,1
This is another variant of iMac 9,1 with a different codec SSID.

Reported-and-tested-by: Everaldo Canuto <everaldo.canuto@gmail.com>
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-12 10:07:36 +01:00
Mukund Navada d055852ee8 ASoC: core: Double control update err for snd_soc_put_volsw_sx
snd_soc_put_volsw_sx function fails to update second control
if first control is updated by snd_soc_update_bits_locked.

Signed-off-by: Mukund Navada <navada@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-11-09 16:32:05 +00:00
Misael Lopez Cruz 445632ad6d ASoC: dapm: Use card_list during DAPM shutdown
DAPM shutdown incorrectly uses "list" field of codec struct while
iterating over probed components (codec_dev_list). "list" field
refers to codecs registered in the system, "card_list" field is
used for probed components.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-11-09 16:31:59 +00:00
Takashi Iwai 8bb4d9ce08 ALSA: Fix card refcount unbalance
There are uncovered cases whether the card refcount introduced by the
commit a0830dbd isn't properly increased or decreased:
- OSS PCM and mixer success paths
- When lookup function gets NULL

This patch fixes these places.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08 14:36:18 +01:00
Kailang Yang 19a62823ea ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150)
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08 10:29:22 +01:00
Kailang Yang 1387e2d127 ALSA: hda - Improve HP depop when system enter to S3
alc269_toggle_power_output() was only use in ALC269VB.  I rename it to
alc269vb_toggle_power_output().

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08 10:29:20 +01:00
Takashi Iwai f58161ba1b ALSA: usb-audio: Fix crash at re-preparing the PCM stream
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback.  It turned out that the problem is that we don't
wait until all URBs are killed.

This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181

Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08 08:56:44 +01:00
Adrian Knoth d1a3c98d50 ALSA: hdspm - Fix sync check reporting on RME RayDAT
The RayDAT reports the sync status of its inputs in consecutive bit
positions, so all we do in hdspm_s1_sync_check is to iterate over idx:

    status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);

    lock = (status & (0x1<<idx)) ? 1 : 0;
    sync = (status & (0x100<<idx)) ? 1 : 0;

The index is given in kcontrol->private_value:

    HDSPM_SYNC_CHECK("WC SyncCheck", 0),
    HDSPM_SYNC_CHECK("AES SyncCheck", 1),
    HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2),
    HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3),
    HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4),
    HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5),
    HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6),
    HDSPM_SYNC_CHECK("TCO SyncCheck", 7),
    HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8),

The patch corrects the indicated sync flags by passing the proper index
value to hdspm_s1_sync_check().

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-07 19:55:22 +01:00
Wei Yongjun 5c855c8e2b ASoC: cs42l52: fix the return value of cs42l52_set_fmt()
Fix the return value of cs42l52_set_fmt() when clock inversion is
not allowed and also remove the useless variable ret.

dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)

[We had been assigning to ret but then ignoring the value we assgined
-- broonie]

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-11-07 15:50:06 +01:00
Charles Keepax 6268f74990 ASoC: bells: Correct type in sub speaker DAI name for WM5102
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-07 15:46:11 +01:00
Takashi Iwai d5266125fb ALSA: hda - Add pin fixups for ASUS G75
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.

Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-07 14:42:05 +01:00
Takashi Iwai ef4da45828 ALSA: hda - Fix invalid connections in VT1802 codec
VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e.  This confuses
the auto-parser.  Fix it up in the driver by overriding these
connections.

Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-07 14:42:04 +01:00
Takashi Iwai 5b3761954d ALSA: hda - Fix empty DAC filling in patch_via.c
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i).  When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[].  This confuses is_empty_dac() and trims the detected DAC
in later reference.

This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().

Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-07 14:42:00 +01:00
Eric Millbrandt 55c6f4cb6e ASoC: wm8978: pll incorrectly configured when codec is master
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission.  On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used.  Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.

Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-11-06 09:37:35 +01:00
Takashi Iwai ae24c3191b ALSA: hda - Force to reset IEC958 status bits for AD codecs
Several bug reports suggest that the forcibly resetting IEC958 status
bits is required for AD codecs to get the SPDIF output working
properly after changing streams.

Original fix credit to Javeed Shaikh.

BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361

Reported-by: Robin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-05 12:36:32 +01:00
Ondrej Zary 5c0ee9497b ALSA: es1968: Add ESS vendor ID to pm_whitelist
Add generic ESS vendor ID to pm_whitelist. This should fix suspend on
all Maestro-2 and Maestro-2E based PCI cards.
Tested on Terratec DMX and SF64-PCE2.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-05 12:32:35 +01:00
Daniel J Blueman 00e17f767e ALSA: HDA: Mark CS260x immutable structures const
Mark structures that won't change const.

Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-05 12:29:00 +01:00
Daniel J Blueman 16337e028a ALSA: HDA: Fix digital microphone on CS420x
Correctly enable the digital microphones with the right bits in the
right coeffecient registers on Cirrus CS4206/7 codecs. It also
prevents misconfiguring ADC1/2.

This fixes the digital mic on the Macbook Pro 10,1/Retina.

Based-on-patch-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-05 12:28:30 +01:00
Alexander Stein 5a83b4b5a3 ALSA: hda: Cirrus: Fix coefficient index for beep configuration
Signed-off-by: Alexander Stein <alexander.stein@systec-electronic.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-05 12:27:38 +01:00
Lars R. Damerow f0b3da9843 ALSA: hda - support Teradici 2200 host card audio
The audio chipset used in Teradici's Tera2 host cards is the same as that in
the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards.

Signed-off-by: Lars R. Damerow <lars@pixar.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-04 09:24:08 +01:00
Masanari Iida ec8f53fb69 ALSA: Fix typo in drivers sound
Correct spelling typo in debug messages within drivers/sound

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-04 09:20:58 +01:00
Fabio Estevam f55f14752e ASoC: mxs-saif: Fix channel swap for 24-bit format
Playing 24-bit format file leads to channel swap on mx28 and the reason is that
the current driver performs one write/read to/from the SAIF_DATA register to
trigger the transfer.

This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register
and thus is capable of storing the 16-bit left and right channels, but for the
S24_LE case it can only store one channel, so in order to not lose the FIFO sync
an extra read/write is needed.

Reported-by: Dan Winner <DWinner@tc-helicon.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Dan Winner <DWinner@tc-helicon.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-02 15:03:06 +00:00
Dimitris Papastamos 4868ce57bf ASoC: bells: Select WM1250-EV1 Springbank audio I/O module
Ensure we select the WM1250-EV1 as the current software system
configuration demands it.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-02 14:20:03 +00:00
Dimitris Papastamos 213a796564 ASoC: bells: Add missing select of WM0010
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-02 14:20:01 +00:00
Fabio Estevam 9f4c3f1cde ASoC: mxs-saif: Add MODULE_ALIAS
Add MODULE_ALIAS information.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-11-01 14:49:15 +00:00
Takashi Iwai 16c2e1fae8 ALSA: ice1724: Fix rate setup after resume
The rate isn't restored properly after resume since it's only set up
in hw_params, and not in prepare callback.  For fixing it, put the
corresponding call to resume callback as well.

Reported-and-tested-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-31 07:41:42 +01:00
Mark Brown fe81ad1c2d ASoC: wm5102: Write register value corrections after SYSCLK is enabled
Evalation of the WM5102 has identified a number of register values which
should be written after SYSCLK is enabled on revision A in order to
improve performance.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-10-30 11:51:46 +00:00
Takashi Iwai 0914f7961b ALSA: Avoid endless sleep after disconnect
When disconnect callback is called, each component should wake up
sleepers and check card->shutdown flag for avoiding the endless sleep
blocking the proper resource release.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:15 +01:00
Takashi Iwai a0830dbd4e ALSA: Add a reference counter to card instance
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.

The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:10 +01:00
Takashi Iwai 888ea7d5ac ALSA: usb-audio: Fix races at disconnection in mixer_quirks.c
Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:05 +01:00
Takashi Iwai 34f3c89fda ALSA: usb-audio: Use rwsem for disconnect protection
Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.

Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:00 +01:00
Takashi Iwai 978520b75f ALSA: usb-audio: Fix races at disconnection
Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.

The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
 mixer.c and others; the device speed is now cached in subs->speed
 instead of accessing to chip->dev

The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.

The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks.  They'll be covered by the
upcoming change to rwsem.

Also the mixer codes are untouched, too.  These will be fixed in
another patch, too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:06:54 +01:00
Takashi Iwai 9b0573c07f ALSA: PCM: Fix some races at disconnection
Fix races at PCM disconnection:
- while a PCM device is being opened or closed
- while the PCM state is being changed without lock in prepare,
  hw_params, hw_free ops

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:06:48 +01:00
Mark Brown 804f5ba7e8 ASoC: wm5102: Hook up DSP1
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-10-28 17:47:41 +00:00
Mark Brown 0b09df6652 ASoC: arizona: Define standard hookup for ADSP2
Many Arizona class devices contain ADSP2 cores with a standard method for
hooking them into the audio map. Define standard helpers for this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-10-28 17:47:40 +00:00