Commit Graph

360799 Commits

Author SHA1 Message Date
Takashi Iwai 8bc0a8469c ALSA: hda - Make the resume of digital beep setup proper
The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2
should be executed at resume as well.  Use the cached write for it
being performed automatically at resume.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 12:58:48 +01:00
Takashi Iwai e914b25e37 ALSA: hda - Fix power-saving during playing beep sound
While playing the digital beep tone, the codec shouldn't be turned
off.  This patch adds proper snd_hda_power_up()/down() calls at each
time when the beep is played or off.

Also, this fixes automatically an unnecessary codec power-up at
detaching the beep device when the beep isn't being played.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 12:58:47 +01:00
Takashi Iwai 7504b6cd22 ALSA: hda - Move beep attach/detach calls in hda_generic.c
Instead of calling snd_hda_attach_beep_device() and
snd_hda_detach_beep_device() in each codec driver, move them to the
generic parser.  The codec driver just needs to set spec->beep_nid for
activating the digital beep.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 12:58:42 +01:00
Takashi Iwai cf30f46acd Merge branch 'for-linus' into for-next
Back-merged for refactoring beep stuff.
2013-03-18 11:04:42 +01:00
Takashi Iwai a86b1a2cd2 ALSA: hda/cirrus - Fix the digital beep registration
The argument passed to snd_hda_attach_beep_device() is a widget NID
while spec->beep_amp holds the composed value for amp controls.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 11:00:44 +01:00
Takashi Iwai 31b6945a89 ALSA: hda - Fix missing beep detach in patch_conexant.c
This leaks the beep input device after module unload, which leads to
Oops.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 10:06:41 +01:00
Daniel Mack 0959f22ee6 ALSA: snd-usb: add delay quirk for "Playback Design" products
"Playback Design" products need a 50ms delay after setting the USB
interface.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:47:21 +01:00
Daniel Mack 717bfb5f46 ALSA: snd-usb: handle raw data format of UAC2 devices
UAC2 compliant audio devices may announce the capability to transport
raw audio data on their endpoints. Catch this and handle it as
'special' stream on the ALSA side.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:47:13 +01:00
Daniel Mack 2fcdb06d49 ALSA: snd-usb: handle the bmFormats field as unsigned int
This field may use up to 32 bits, so it should be handled as unsigned
int.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:47:04 +01:00
Mark Hills 59ea586f54 ALSA: usb-audio: Trust fields given in the quirk
The maxpacksize field is given in some quirks, but it gets ignored (in
favour of wMaxPacketSize from the first endpoint.) This patch favours
the one in the quirk.

Digidesign Mbox and Mbox 2 are the only affected quirks and the devices
are assumed to be working without this patch. So for safety against the
values in the quirk being incorrect, remove them.

The datainterval is also ignored but there are not currently any quirks
which choose to override this.

Cc: Damien Zammit <damien@zamaudio.com>
Cc: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:46:37 +01:00
Mark Hills 5e212332cc ALSA: usb-audio: Playback and MIDI support for Novation Twitch DJ controller
The hardware also has a PCM capture device which is not implemented in
this patch.

It may be possible to generalise this to Saffire 6 USB support and some
of the other Focusrite interfaces, but as I don't have access to these
devices we should wait until capture support is working first.

Capture support is not implemented because the code assumes the endpoint
to have its own interface (instead, it shares the interface with playback)
and some thought will be needed to lift this limitation.

Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:46:18 +01:00
Masanari Iida 9ad477a145 ALSA: documentation: Fix typo in Documentation/sound
Correct spelling typos in Documentation/sound/alsa

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-17 10:12:13 +01:00
Takashi Iwai 6d3073e124 ALSA: hda - Fix missing EAPD/GPIO setup for Cirrus codecs
During the transition to the generic parser, the hook to the codec
specific automute function was forgotten.  This resulted in the silent
output on some MacBooks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15 14:24:45 +01:00
Dan Carpenter 57220bc1f5 sound: sequencer: cap array index in seq_chn_common_event()
"chn" here is a number between 0 and 255, but ->chn_info[] only has
16 elements so there is a potential write beyond the end of the
array.

If the seq_mode isn't SEQ_2 then we let the individual drivers
(either opl3.c or midi_synth.c) handle it.  Those functions all
do a bounds check on "chn" so I haven't changed anything here.
The opl3.c driver has up to 18 channels and not 16.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15 07:45:20 +01:00
Dylan Reid b714a7106b ALSA: hda/ca0132 - Remove extra setting of dsp_state.
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset
and check it in ca0132_download_dsp().

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15 07:41:12 +01:00
Dylan Reid e8f1bd5d77 ALSA: hda/ca0132 - Check download state of DSP.
Instead of using the dspload_is_loaded() function, check the dsp_state
that is kept in the spec.  The dspload_is_loaded() function returns
true if the DSP transfer was never started.  This false-positive leads
to multiple second delays when ca0132_setup_efaults() times out on
each write.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15 07:40:39 +01:00
Dylan Reid d1d28500cc ALSA: hda/ca0132 - Check if dspload_image succeeded.
If dspload_image() fails, it was ignored and dspload_wait_loaded() was
still called.  dsp_loaded should never be set to true in this case,
skip it.  The check in dspload_wait_loaded() return true if the DSP is
loaded or if it never started.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15 07:40:11 +01:00
David Henningsson 303985f810 ALSA: hda - Disable IDT eapd_switch if there are no internal speakers
If there are no internal speakers, we should not turn the eapd switch
off, because it might be necessary to keep high for Headphone.

BugLink: https://bugs.launchpad.net/bugs/1155016
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-14 15:31:45 +01:00
Takashi Iwai ba615b86d6 ALSA: hda - Don't apply EAPD power filter as default
So far, the driver doesn't power down the widget at going down to D3
when the widget node has an EAPD capability and EAPD is actually set
on all codecs unless codec->power_filter is set explicitly.
This caused a problem on some Conexant codecs, leading to click
noises, and we set it as NULL there.  But it is very unlikely that the
problem hits only these codecs.

Looking back at the development history, this workaround for EAPD was
introduced just for some laptops with STAC9200 codec, then we applied
it blindly for all.  Now, since it's revealed to have an ill effect,
we should disable this workaround per default and apply only for the
known requiring systems.

The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(),
and this has to be set explicitly by the codec driver when needed.
As of now, only patch_stac9200() sets this one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-13 18:07:05 +01:00
Takashi Iwai bce0d2a80e ALSA: hda - Allow unlimited pins and converters in patch_hdmi.c
Use the dynamic array allocations for pins, converters and PCM arrays
instead of the fixed size arrays.  The modern HDMI codecs get more and
more pins, and we don't know the sensitive limit.

Most of the patch are spent for the straight conversions from the
fixed array access to snd_array helpers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-13 18:07:04 +01:00
Takashi Iwai 5265fd9a9f ALSA: hda - Drop explicit memset() by reallocation with __GFP_ZERO
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-13 18:06:59 +01:00
Takashi Iwai 0bc0ec903c ALSA: info: Small refactoring and a sanity check in snd_info_get_line()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-13 12:11:13 +01:00
Takashi Iwai 0d861ac238 ALSA: info: Avoid leaking kernel memory
Make sure that the allocated buffer for reading the proc file won't
expose the uncleared kernel memory.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-13 12:03:33 +01:00
Takashi Iwai b5f82b1044 ALSA: hda - Fix snd_hda_get_num_raw_conns() to return a correct value
In the connection list expansion in hda_codec.c and hda_proc.c, the
value returned from snd_hda_get_num_raw_conns() is used as the array
size to store the connection list.  However, the function returns
simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the
widget list with ranges isn't considered there.  Thus it may return a
smaller size than the actual list, which results in -ENOSPC in
snd_hda_get_raw_conections().

This patch fixes the bug by parsing the connection list correctly also
for snd_hda_get_num_raw_conns().

Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-12 16:47:30 +01:00
Clemens Ladisch 281a6ac0f5 ALSA: usb-audio: add a workaround for the NuForce UDH-100
The NuForce UDH-100 numbers its interfaces incorrectly, which makes the
interface associations come out wrong, which results in the driver
erroring out with the message "Audio class v2 interfaces need an
interface association".

Work around this by searching for the interface association descriptor
also in some other place where it might have ended up.

Reported-and-tested-by: Dave Helstroom <helstroom@google.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-12 08:35:30 +01:00
Wei Yongjun 2e9b9a3c24 ALSA: asihpi - fix potential NULL pointer dereference
The dereference should be moved below the NULL test.

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-12 08:34:36 +01:00
Yacine Belkadi eb7c06e8e9 ALSA: add/change some comments describing function return values
script/kernel-doc reports the following type of warnings (when run in verbose
mode):

Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'

To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values

Along the way:
- complete some descriptions
- fix some typos

Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-12 08:32:53 +01:00
Adrian Knoth a817650ebb ALSA: hdspm - Enable new TCO ALSA controls
Expose the newly added TCO LTC and sync check functions to userspace.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 10:11:21 +01:00
Adrian Knoth f99c78812f ALSA: hdspm - Add ALSA controls to read the TCO LTC state
This patch adds new ALSA controls to query the LTC state from userspace.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 10:11:20 +01:00
Adrian Knoth 345422133a ALSA: hdspm - Also check for TCO sync states
This patch prepares snd_hdspm_get_sync_check() to also check the TCO
sync state. The added feature will be exposed to the user in a later
commit.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 10:11:19 +01:00
Adrian Knoth e5b7b1fe3b ALSA: hdspm - Remove duplicate code from ALSA controls
Considerably shorten the code by using a macro. Though this won't lower
the binary size, it makes the source more readable.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 10:11:18 +01:00
Adrian Knoth 696be0fbe2 ALSA: hdspm - Provide ALSA control to disable 96K frames
For 96kHz, MADI allows to multiplex the samples (SMUX) or to use a
dedicated 96K mode. The RME cards default to 96K mode, but since not all
external MADI equipment supports this, provide a switch to users that
changes the on-wire protocol to SMUX.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 10:11:17 +01:00
Adrian Knoth fcdc4ba1d8 ALSA: hdspm - Allow the TCO and SYNC-IN to be used in slave mode
When using the additional Time Code Option module in slave mode or the
SYNC-In wordclock connector, the sample rate needs to be returned by
hdspm_external_sample_rate().

Since this sample rate may contain any value with 1Hz granularity, we
need to round it to a common rate as done by the OSX driver.

[Fixed missing function declarations by tiwai]

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 10:10:53 +01:00
Adrian Knoth 3f7bf918bf ALSA: hdspm - Refactor sample rate acquisition
This commit introduces hdspm_get_pll_freq() to avoid code duplication.
Reading the sample rate from the DDS register will be required by
upcoming code.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 09:57:22 +01:00
Takashi Iwai 93c9d8ae0b ALSA: hda - Don't re-initialize shared hp/mic pinctl
When a headphone pin is set up as a shared hp/mic pin, we rather want
to keep it as a headphone primarily as default, but the driver
overrides it always as a mic pin, just because the input controls are
created after outputs.  Add a check of pin NID and skip the
re-initialization of pinctl for such a shared hp/mic pin.

Reported-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 09:57:21 +01:00
Takashi Iwai 66efdc71d9 ALSA: seq: Fix missing error handling in snd_seq_timer_open()
snd_seq_timer_open() didn't catch the whole error path but let through
if the timer id is a slave.  This may lead to Oops by accessing the
uninitialized pointer.

 BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae
 IP: [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130
 PGD 785cd067 PUD 76964067 PMD 0
 Oops: 0002 [#4] SMP
 CPU 0
 Pid: 4288, comm: trinity-child7 Tainted: G      D W 3.9.0-rc1+ #100 Bochs Bochs
 RIP: 0010:[<ffffffff819b3477>]  [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130
 RSP: 0018:ffff88006ece7d38  EFLAGS: 00010246
 RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000
 RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38
 RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe
 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000
 R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007
 FS:  00007f7aaa650700(0000) GS:ffff88007f800000(0000) GS:0000000000000000
 CS:  0010 DS: 0000 ES: 0000 CR0: 0000000080050033
 CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0
 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000
 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400
 Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290)
 Stack:
  0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d
  65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000
  ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520
 Call Trace:
  [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10
  [<ffffffff819b17e9>] snd_seq_queue_timer_open+0x29/0x70
  [<ffffffff819ae01a>] snd_seq_ioctl_set_queue_timer+0xda/0x120
  [<ffffffff819acb9b>] snd_seq_do_ioctl+0x9b/0xd0
  [<ffffffff819acbe0>] snd_seq_ioctl+0x10/0x20
  [<ffffffff811b9542>] do_vfs_ioctl+0x522/0x570
  [<ffffffff8130a4b3>] ? file_has_perm+0x83/0xa0
  [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10
  [<ffffffff811b95ed>] sys_ioctl+0x5d/0xa0
  [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f
  [<ffffffff81faed69>] system_call_fastpath+0x16/0x1b

Reported-and-tested-by: Tommi Rantala <tt.rantala@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 09:40:36 +01:00
Christine Spang d5702162f8 ALSA: Make snd_BUG_ON() always evaluate and return the conditional expression
Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG
is set leads to frequent bugs, since other similar macros in the kernel
have different behavior. Let's make snd_BUG_ON() act like those macros
so it will stop being accidentally misused.

Signed-off-by: Christine Spang <christine.spang@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-11 09:33:34 +01:00
Takashi Iwai 8ba955cef3 ALSA: hda - Avoid automatic pin-ctl update for hp/mic when jack ctl exists
When the headphone mic jack enum control is created (via explicitly
specification by user), it doesn't make much sense to change the I/O
direction dynamically per capture source change, since the I/O
direction is rather controlled over the enum ctl.

This also reduces the implicit dependency between the capture source
and the hp mic jack enum ctls, which might confuse a program accessing
the whole control elements at once like alsactl.

In addition, this patch introduces update_hp_automute_hook() function
to call the proper hook function.  It's just to remove the open codes
in multiple places in hda_generic.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 18:43:27 +01:00
Takashi Iwai f811c3cf8f ALSA: hda - Consolidate add_in_jack_modes and add_out_jack_modes hints
There is no big merit to distinguish these two hints.  Instead, just
have a single flag, add_jack_modes, for creating the jack mode enum
ctls for both I/O directions.

The hint string parser code is left and translated as add_jack_modes
just for keeping compatibility.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 18:32:59 +01:00
Takashi Iwai 3f550e3232 ALSA: hda - Allow to change I/O direction in hp/mic jack mode ctl
The previous commits added the capability to change the pin control of
hp/mic shared jack, but it actually didn't work as expected when the
value is changed from the output to the input, since I forgot to reset
the pin I/O bit in that case.  This patch fixes the problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 18:30:27 +01:00
Takashi Iwai 5ebd3bbdcc ALSA: hda - Add some model name strings for ALC260
In order to let user test the known workaround more easily, give a few
known fixups for ALC260 to the model strings so that it can be passed
via the module option.

Also, move the unusual setups found in FSC S7020 fixup into a special
model, fujitsu-jwse, Jonathan Woithe Special Edition.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 18:30:07 +01:00
Takashi Iwai 5f171baaa5 ALSA: hda - Handle shared hp/mic jack mode
When a headphone jack is configured as a shared hp/mic jack, the jack
mode enum needs to handle both input and output directions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 18:30:01 +01:00
Takashi Iwai 967303dabc ALSA: hda - Add the generic Headphone Mic feature
This patch improves the generic parser code to allow to set up the
headphone jack as a mic input.  User can enable this feature by giving
hp_mic hint string.

The former shared hp/mic feature for the single built-in mic is still
retained.  This detection can be disabled now via hp_mic_detect hint
string, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 18:29:52 +01:00
Sean Connor 69a4cfdd44 ALSA: ice1712: Initialize card->private_data properly
Set card->private_data in snd_ice1712_create for fixing NULL
dereference in snd_ice1712_remove().

Signed-off-by: Sean Connor <sconnor004@allyinics.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 15:38:58 +01:00
Daniel Mack 2dad940219 ALSA: snd-usb-caiaq: fix smatch warnings
Fix three smatch warnings recently introduced:

sound/usb/caiaq/device.c:166 usb_ep1_command_reply_dispatch() warn:
  variable dereferenced before check 'cdev' (see line 163)
sound/usb/caiaq/device.c:517 snd_disconnect() warn: variable
  dereferenced before check 'card' (see line 514)
sound/usb/caiaq/input.c:510 snd_usb_caiaq_ep4_reply_dispatch() warn:
  variable dereferenced before check 'cdev' (see line 506)

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 09:24:12 +01:00
Kailang Yang 84dfd0ac23 ALSA: hda - Add support of new codec ALC233
It's compatible with ALC282.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 09:21:01 +01:00
Xi Wang 3bc085a12d ALSA: hda/ca0132 - Avoid division by zero in dspxfr_one_seg()
Move the zero check `hda_frame_size_words == 0' before the modulus
`buffer_size_words % hda_frame_size_words'.

Also remove the redundant null check `buffer_addx == NULL'.

Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 09:18:00 +01:00
Mengdong Lin 4c7a548a70 ALSA: hda - check NULL pointer when creating SPDIF PCM switch
If the new control cannot be created, this function will return to avoid
snd_hda_ctl_add dereferencing a NULL control pointer.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 09:14:03 +01:00
Mengdong Lin 25336e8ae2 ALSA: hda - check NULL pointer when creating SPDIF controls
If the SPDIF control array cannot be reallocated, the function
will return to avoid dereferencing a NULL pointer.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 09:12:14 +01:00
Takashi Iwai 9fedcc44f1 ASoC: Updates for v3.9
A few driver fixes, none of them terribly dramatic.
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Merge tag 'asoc-v3.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v3.9

A few driver fixes, none of them terribly dramatic.
2013-03-07 09:11:22 +01:00