OSDN Git Service
Takashi Iwai [Wed, 26 Oct 2011 21:51:43 +0000 (23:51 +0200)]
Merge branch 'topic/misc' into for-linus
Takashi Iwai [Wed, 26 Oct 2011 21:04:08 +0000 (23:04 +0200)]
ALSA: hda - Fix pin-config for ASUS W90V
The association numbers of surround/CLFE speaker pins aren't correctly
mapped by the auto-parser. This patch fixes the CLFE speaker pin to the
right assoc value (from 3 to 1).
Tested-by: Nika Topolchanskaya <nanodesuu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 26 Oct 2011 14:06:27 +0000 (16:06 +0200)]
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
When 5.1 or more headphone or speaker pins are provided, the parser still
takes as is without fixing the order of channel mapping, which leads in
the unexpected strange channel order by surround outputs.
This patch fixes the issue by applying the same fix-up not only to
line_out_pins[] but also hp_pins[] and speaker_pins[].
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alexander Stein [Wed, 26 Oct 2011 07:58:45 +0000 (09:58 +0200)]
ALSA: hda - Fix typo
Signed-off-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 25 Oct 2011 08:00:22 +0000 (10:00 +0200)]
ALSA: Update the sound git tree URL
Now back to kernel.org but without -2.6 suffix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Tue, 18 Oct 2011 12:07:51 +0000 (14:07 +0200)]
ALSA: HDA: Add new revision for ALC662
The revision 0x100300 was found for ALC662. It seems to work well
with patch_alc662.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/877373
Tested-by: Shengyao Xue <Shengyao.xue@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 21 Oct 2011 13:07:42 +0000 (15:07 +0200)]
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
When a device has multiple speakers and still has the auto-mute support,
the driver copies line_outs[] to speaker_outs[]. And then it tries to
assign DACs for both. This ended up with the assignment only to the
primary DAC to all speakers.
This patch fixes the situation by checking the duplicated LO/SPK case
appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 19 Oct 2011 15:20:08 +0000 (17:20 +0200)]
Merge branch 'fix/hda' into topic/hda
Daniel Suchy [Tue, 18 Oct 2011 09:09:44 +0000 (11:09 +0200)]
ALSA: HDA: conexant support for Lenovo T520/W520
This is patch for Conexant codec of Intel HDA driver, adding new quirk
for Lenovo Thinkpad T520 and W520. Conexant autodetection works fine for
T520 (similar subsystem ID is used also in W520 model) and detects more
mixer features compared to generic (fallback) Lenovo quirk with
hardcoded options in Conexant codec.
Patch was activelly tested with Linux 3.0.4, 3.0.6 and 3.0.7 without any
problems.
Signed-off-by: Daniel Suchy <danny@danysek.cz>
Cc: <stable@kernel.org> [3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 18 Oct 2011 08:44:05 +0000 (10:44 +0200)]
ALSA: hda - Add position_fix quirk for Dell Inspiron 1010
The previous fix for the position-buffer check gives yet another
regression on a Dell laptop. The safest fix right now is to add a
static quirk for this device (and better to apply it for stable
kernels too).
Reported-by: Éric Piel <Eric.Piel@tremplin-utc.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 17 Oct 2011 14:50:59 +0000 (16:50 +0200)]
ALSA: hda/realtek - Cache COEF 0 value
The COEF #0 value represents a sort of device id, so it's supposedly
constant while operation. Better to use the cached value instead of
reading it at each time from the performance POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 17 Oct 2011 14:39:09 +0000 (16:39 +0200)]
ALSA: hda/realtek - Clean up codec renames
Use a static table for detecting the codec renames.
Also clean up the error paths in each patch_*() function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 17 Oct 2011 14:07:43 +0000 (16:07 +0200)]
ALSA: hda/realtek - Use alc_codec_rename()
Replaced with alc_codec_rename() in all possible places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kailang Yang [Mon, 17 Oct 2011 14:02:42 +0000 (16:02 +0200)]
ALSA: hda - ALC888S-VC remark to ALC886
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 17 Oct 2011 14:00:35 +0000 (16:00 +0200)]
ALSA: hda/realtek - Check the error from alc_codec_rename()
Should be a rare case, but...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 13 Oct 2011 06:19:09 +0000 (08:19 +0200)]
ALSA: usb-audio - Fix possible access over audio_feature_info[] array
The audio_feature_info[] array should contain all entries for UAC2_FU_*,
but currently a few last entries are missing. Even though, the driver
tries to probe these entries in parse_audio_feature_unit() and may
access the range over the array. This patch fixes the bug by limiting
the loop size properly using ARRAY_SIZE() instead of a hard-coded
magic number.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
William Light [Mon, 10 Oct 2011 15:54:23 +0000 (15:54 +0000)]
ALSA: snd-usb-caiaq: Add support for Maschine
This adds partial support for the Maschine controller by Native Instruments.
Supported now are the 1x1 MIDI interface and the 41 buttons, 11 endless
rotary encoders, and 16 pressure-sensitive drum pads. Still to work on are the
dimmable LEDs and the two monochrome screens.
Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
William Light [Mon, 10 Oct 2011 15:54:22 +0000 (15:54 +0000)]
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>
David Henningsson [Wed, 12 Oct 2011 17:26:03 +0000 (19:26 +0200)]
ALSA: HDA: Fixup Realtek headphone pin initialization
This typo caused headphone pins not to be initialized correctly.
BugLink: https://bugs.launchpad.net/bugs/871582
Reported-by: Effenberg
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Charles Chin [Thu, 13 Oct 2011 05:54:09 +0000 (07:54 +0200)]
ALSA: hda - Remove bad code for IDT 92HD83 family patch
The purpose of this patch is to remove a section of "bad" code that
assigns the last DAC to ports E or F in order to support notebooks
with docking in earlier days, around ALSA 1.0.19 - 21. This is not
necessary now and actually breaks some configurations that use these
ports as other devices. This have been tested on several different
configurations to make sure that it is working for different combinations.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Feng Tang [Mon, 10 Oct 2011 02:31:48 +0000 (10:31 +0800)]
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>
Clemens Ladisch [Fri, 7 Oct 2011 20:38:59 +0000 (22:38 +0200)]
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>
Takashi Iwai [Thu, 6 Oct 2011 08:07:58 +0000 (10:07 +0200)]
ALSA: hda - Distinguish each substream for better sticky assignment
The commit
ef18beded8ddbaafdf4914bab209f77e60ae3a18 introduced a
mechanism to assign the previously used slot for the next reopen of a
PCM stream. But the PCM device number isn't always unique (it may
have multiple substreams), and also the code doesn't check the stream
direction, thus both playback and capture streams share the same
device number.
For avoiding this conflict, make a unique key for each substream and
store/check this value at reopening.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 6 Oct 2011 08:04:30 +0000 (10:04 +0200)]
Merge branch 'fix/hda' into topic/hda
Takashi Iwai [Thu, 6 Oct 2011 06:27:19 +0000 (08:27 +0200)]
ALSA: hda/realtek - Choose more cleverly the primary outputs
When the speaker outputs are more than the headphone outputs, it implies
that the system has surround speakers while the headphones are only for
monitoring the front. In such a case, it's better to put speakers as
the primary outputs so that the driver can build up and keep the
surround setup. Otherwise the system will pick up the headphone as
primary, and offers less channels than the speakers do support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 6 Oct 2011 06:16:29 +0000 (08:16 +0200)]
ALSA: hda - Moved snd_print_pcm_rates() back into hda_proc.c
Since hda_proc.c is now the only user of snd_print_pcm_rates(), better to
put it back locally to hda_proc.c and revert to the old style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pierre-Louis Bossart [Wed, 5 Oct 2011 20:14:20 +0000 (15:14 -0500)]
ALSA: hdmi: fix printout of SAD sampling rates
SAD sampling rate information reported in
/proc/asound/cardX/eldX is incorrect due to a mismatch
between HDA and HDMI frequencies. Add new routine to provide
relevant values.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Wed, 5 Oct 2011 13:53:25 +0000 (15:53 +0200)]
ALSA: jack - Add "Line In" input jack constants
Similar to Line Out, these constants form the base for future
patches enabling input jack reporting for Line in jacks.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Wed, 5 Oct 2011 07:49:05 +0000 (09:49 +0200)]
ALSA: HDA: Fix DAC assignment for secondary headphone on Sigmatel/IDT
If we run out of DACs when trying to assign a DAC to a secondary
headphone, prefer the DAC of the first headphone to the primary
(usually line out) DAC.
BugLink: http://bugs.launchpad.net/bugs/845275
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Carpenter [Tue, 4 Oct 2011 06:29:39 +0000 (09:29 +0300)]
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>
Raymond Yau [Tue, 4 Oct 2011 01:46:44 +0000 (09:46 +0800)]
ALSA: hda - Add documentation for codec specific mixer controls of Analog codecs
* Channel Mode
This is an enum control to change the surround-channel setup,
appears only when the surround channels are available.
It gives the number of channels to be used, "2ch", "4ch" abd "6ch".
According to the configuration, this also controls the
jack-retasking of multi-I/O jacks.
* Independent HP
When this enum control is enabled, the headphone output is routed
from an individual stream (the third PCM such as hw:0,2) instead of
the primary stream.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stefan Richter [Sat, 27 Aug 2011 14:45:28 +0000 (16:45 +0200)]
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
f3f7c1837f6bcae3601fc535b339426868bf1549
"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>
David Henningsson [Mon, 3 Oct 2011 14:25:42 +0000 (16:25 +0200)]
ALSA: HDA: Fix naming of input jacks for IDT parser
The Sigmatel/IDT parser should have the same naming convention
for input jacks as the other codecs have.
BugLink: http://bugs.launchpad.net/bugs/859704
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pierre-Louis Bossart [Fri, 30 Sep 2011 21:35:41 +0000 (16:35 -0500)]
ALSA: hda/hdmi: expose ELD control
Applications may want to read ELD information to
understand what codecs are supported on the HDMI
receiver and handle the a-v delay for better lip-sync.
ELD information is exposed in a device-specific
IFACE_PCM kcontrol. Tested both with amixer and
PulseAudio; with a corresponding patch passthrough modes
are enabled automagically.
ELD control size is set to zero in case of errors or
wrong configurations. No notifications are implemented
for now, it is expected that jack detection is used to
reconfigure the audio outputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 30 Sep 2011 06:52:26 +0000 (08:52 +0200)]
ALSA: hda - Fix a regression of the position-buffer check
The commit
a810364a0424c297242c6c66071a42f7675a5568
ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.
This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().
Reported-and-tested-by: Rocko Requin <rockorequin@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Carpenter [Thu, 29 Sep 2011 06:10:48 +0000 (09:10 +0300)]
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>
Takashi Iwai [Wed, 28 Sep 2011 18:12:08 +0000 (20:12 +0200)]
ALSA: hda - Allow patching with any vendor/subsystem ids
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 28 Sep 2011 15:16:09 +0000 (17:16 +0200)]
ALSA: hda - Add snoop option
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 28 Sep 2011 15:12:59 +0000 (17:12 +0200)]
ALSA: pcm - Export snd_pcm_lib_default_mmap() helper
Export the default mmap function, snd_pcm_lib_default_mmap().
The upcoming non-snooping support in HD-audio driver will use this
to override the mmap method.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 28 Sep 2011 14:43:36 +0000 (16:43 +0200)]
ALSA: hda:via - Skip creations of empty PCM streams
If no analog I/O is defined, skip creating the corresponding PCM stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 27 Sep 2011 16:21:41 +0000 (18:21 +0200)]
Merge branch 'fix/asoc' into for-linus
Takashi Iwai [Tue, 27 Sep 2011 15:33:45 +0000 (17:33 +0200)]
ALSA: hda - Avoid unnecessary verbs to clear PCM formats
Since really_cleanup_stream() is called from both purity_inactive_streams()
and hda_cleanup_all_streams(), the verbs to clear the PCM channel and
format may be called multiple times unnecessarily.
This patch adds checks to skip these unneeded verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lars-Peter Clausen [Tue, 27 Sep 2011 09:08:46 +0000 (11:08 +0200)]
ASoC: ssm2602: Re-enable oscillator after suspend
Currently the the internal oscillator is powered down when entering BIAS_OFF
state, but not re-enabled when going back to BIAS_STANDBY. As a result the
CODEC will stop working after suspend if the internal oscillator is used to
generate the sysclock signal. This patch fixes it by clearing the appropriate
bit in the power down register when the CODEC is re-enabled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Clemens Ladisch [Mon, 26 Sep 2011 19:15:27 +0000 (21:15 +0200)]
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>
Thomas Pfaff [Mon, 26 Sep 2011 13:43:59 +0000 (15:43 +0200)]
ALSA: usb-audio: Check for possible chip NULL pointer before clearing probing flag
Before clearing the probing flag in the error exit path, check that the
chip pointer is not NULL.
Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 26 Sep 2011 13:27:10 +0000 (15:27 +0200)]
Merge branch 'fix/hda' into topic/hda
Takashi Iwai [Mon, 26 Sep 2011 13:19:55 +0000 (15:19 +0200)]
ALSA: hda/realtek - Don't detect LO jack when identical with HP
The spec->autocfg.line_out_pins[] may contain the same pins as hp_pins[]
depending on the configuration. When they are identical, detecting the
line_jack_present flag screws up the auto-mute because alc_line_automute()
is called unconditionally at initialization while it won't be triggered
by unsol events, thus the old line_jack_present flag is kept for the
whole run.
For fixing this buggy behavior, the driver needs to check whether the
line-outs are really individual, and skip if same as headphone jacks.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 26 Sep 2011 08:41:21 +0000 (10:41 +0200)]
ALSA: hda/realtek - Avoid bogus HP-pin assignment
When the headphone pin is assigned as primary output to line_out_pins[],
the automatic HP-pin assignment by ASSID must be suppressed. Otherwise
a wrong pin might be assigned to the headphone and breaks the auto-mute.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Takashi Iwai [Sat, 24 Sep 2011 10:16:29 +0000 (12:16 +0200)]
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>
David Henningsson [Sat, 24 Sep 2011 06:30:44 +0000 (08:30 +0200)]
ALSA: HDA: No power nids on 92HD93
This patch is necessary to make internal speakers work on this chip.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Tested-by: Alex Wolfson <alex.wolfson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 23 Sep 2011 13:26:37 +0000 (15:26 +0200)]
Merge branch 'fix/asoc' into for-linus
Thomas Pfaff [Thu, 22 Sep 2011 16:26:06 +0000 (18:26 +0200)]
ALSA: usb-audio - clear chip->probing on error exit
The Terratec Aureon 5.1 USB sound card support is broken since kernel
2.6.39.
2.6.39 introduced power management support for USB sound cards that added
a probing flag in struct snd_usb_audio.
During the probe of the card it gives following error message :
usb 7-2: new full speed USB device number 2 using uhci_hcd
cannot find UAC_HEADER
snd-usb-audio: probe of 7-2:1.3 failed with error -5
input: USB Audio as
/devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6
generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio]
on usb-0000:00:1d.1-2/input3
I can not comment about that "cannot find UAC_HEADER" error, but until
2.6.38 the card worked anyway.
With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl
stops in snd_usb_autoresume with -ENODEV.
Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Raymond Yau [Fri, 23 Sep 2011 11:03:25 +0000 (19:03 +0800)]
ALSA: HDA - Add Independent Headphone for all models of ad1988/ad1989
- Add "AD198x Headphone" playback device for independent headphone playback
while playing 7.1 surround using rear panel audio jacks.
- Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using
DAC0 instead of DAC1 (HDA_FRONT) was already added to all models.
- Add "Independent HP" switch to enable/disable this playback device.
When the switch is OFF, headphone use "copy front" mode to get the front
channel as the green jack.
When the switch is ON, you can play stereo sound through "AD198x Headphone"
device to headphone while playing 7.1 surround sound through "AD198x Analog"
device.
The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog"
is open.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andy Shevchenko [Fri, 23 Sep 2011 11:32:11 +0000 (14:32 +0300)]
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>
Jarkko Nikula [Fri, 23 Sep 2011 08:19:13 +0000 (11:19 +0300)]
ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
Attempt to change McBSP CLKS source while another stream is active is not
safe after commit
d135865 ("OMAP: McBSP: implement functional clock
switching via clock framework") in 2.6.37.
CLKS parent clock switching using clock framework have to idle the McBSP
before switching and then activate it again. This short break can cause a
DMA transaction error to already running stream which halts and recovers
only by closing and restarting the stream.
This goes more fatal after commit
e2fa61d ("OMAP3: l3: Introduce
l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a
severe timeout error and does BUG_ON().
Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk
if the McBSP is already active. This test should have been here just from
the beginning anyway.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Dan Carpenter [Fri, 23 Sep 2011 06:25:05 +0000 (09:25 +0300)]
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>
Dan Carpenter [Fri, 23 Sep 2011 06:24:21 +0000 (09:24 +0300)]
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>
Takashi Iwai [Thu, 22 Sep 2011 14:54:23 +0000 (16:54 +0200)]
ALSA: fm801 - Clean up redundant reference to snd_fm801_tea575x_gpios[]
Use macro to improve readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 22 Sep 2011 14:41:52 +0000 (16:41 +0200)]
Merge branch 'fix/misc' into topic/misc
Ben Hutchings [Thu, 22 Sep 2011 13:39:52 +0000 (14:39 +0100)]
ALSA: fm801: Gracefully handle failure of tuner auto-detect
Commit
9676001559fce06e37c7dc230ab275f605556176
("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>
Ben Hutchings [Thu, 22 Sep 2011 13:38:58 +0000 (14:38 +0100)]
ALSA: fm801: Fix double free in case of error in tuner detection
Commit
9676001559fce06e37c7dc230ab275f605556176
("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>
Mark Brown [Tue, 20 Sep 2011 10:41:54 +0000 (11:41 +0100)]
ASoC: Ensure we generate a driver name
Commit
873bd4c (ASoC: Don't set invalid name string to snd_card->driver
field) broke generation of a driver name for all ASoC cards relying on the
automatic generation of one. Fix this by using the old default with spaces
replaced by underscores.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Tue, 23 Aug 2011 14:56:03 +0000 (16:56 +0200)]
ALSA: hda: hdmi: Hint matching between input devices and pcm devices
Since modern HDMI cards often have more than one output pin and thus
input device, we need to know which one has actually been plugged in.
This patch adds a name hint that indicates which PCM device is connected
to which pin.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Tue, 20 Sep 2011 10:04:56 +0000 (12:04 +0200)]
ALSA: HDA: Refactor Realtek's automute
Increase readability and understandability in the automute code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Mon, 19 Sep 2011 22:33:35 +0000 (23:33 +0100)]
ASoC: Remove bitrotted wm8962_resume()
This functionality is now subsumed within the bias management, using the
standard cache management functionality, without assuming the cache type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Takashi Iwai [Tue, 20 Sep 2011 07:14:04 +0000 (09:14 +0200)]
Merge branch 'fix/hda' into topic/hda
David Henningsson [Tue, 20 Sep 2011 07:02:22 +0000 (09:02 +0200)]
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>
Clemens Ladisch [Fri, 16 Sep 2011 21:16:05 +0000 (23:16 +0200)]
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>
Clemens Ladisch [Fri, 16 Sep 2011 21:13:38 +0000 (23:13 +0200)]
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>
Clemens Ladisch [Fri, 16 Sep 2011 21:08:28 +0000 (23:08 +0200)]
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>
Clemens Ladisch [Fri, 16 Sep 2011 21:03:02 +0000 (23:03 +0200)]
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>
Clemens Ladisch [Fri, 16 Sep 2011 20:52:48 +0000 (22:52 +0200)]
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>
Takashi Iwai [Mon, 19 Sep 2011 09:31:34 +0000 (11:31 +0200)]
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>
Daniele Guerrieri [Fri, 16 Sep 2011 06:31:45 +0000 (08:31 +0200)]
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>
Takashi Iwai [Fri, 16 Sep 2011 06:29:04 +0000 (08:29 +0200)]
Merge branch 'fix/misc' into topic/misc
Axel Lin [Thu, 15 Sep 2011 03:04:56 +0000 (11:04 +0800)]
ASoC: bf5xx-
ad73311: Fix prototype for bf5xx_probe
Fix below build warning:
sound/soc/blackfin/bf5xx-
ad73311.c: warning: initialization from incompatible pointer type
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Arjan van de Ven [Thu, 15 Sep 2011 06:49:25 +0000 (08:49 +0200)]
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>
Takashi Iwai [Wed, 14 Sep 2011 17:11:13 +0000 (19:11 +0200)]
Merge branch 'fix/asoc' into for-linus
Daniel Mack [Wed, 14 Sep 2011 10:46:57 +0000 (12:46 +0200)]
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>
Daniel Mack [Mon, 12 Sep 2011 16:54:12 +0000 (18:54 +0200)]
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>
Daniel Mack [Mon, 12 Sep 2011 16:54:11 +0000 (18:54 +0200)]
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>
Daniel Mack [Mon, 12 Sep 2011 16:54:10 +0000 (18:54 +0200)]
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>
David Henningsson [Wed, 14 Sep 2011 11:22:54 +0000 (13:22 +0200)]
ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit
b4dabfc452a10 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>
Clemens Ladisch [Tue, 13 Sep 2011 09:24:41 +0000 (11:24 +0200)]
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>
Takashi Iwai [Tue, 13 Sep 2011 08:33:16 +0000 (10:33 +0200)]
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>
Arnd Bergmann [Sun, 11 Sep 2011 18:07:30 +0000 (20:07 +0200)]
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>
Axel Lin [Fri, 9 Sep 2011 09:50:52 +0000 (17:50 +0800)]
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>
Axel Lin [Fri, 9 Sep 2011 11:04:45 +0000 (19:04 +0800)]
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>
Raymond Yau [Wed, 31 Aug 2011 02:30:59 +0000 (10:30 +0800)]
ALSA: hda - Add Headphone Playback Volume control for ad1988/ad1989
- use DAC0 instead of DAC1 for Port-A Headphone
- assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp"
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Raymond Yau [Fri, 9 Sep 2011 11:15:01 +0000 (19:15 +0800)]
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>
Pierre-Louis Bossart [Wed, 7 Sep 2011 00:15:34 +0000 (19:15 -0500)]
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>
Lars-Peter Clausen [Mon, 5 Sep 2011 11:49:57 +0000 (13:49 +0200)]
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
Mark Brown [Sun, 4 Sep 2011 15:18:18 +0000 (08:18 -0700)]
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
Fabio Estevam [Tue, 30 Aug 2011 03:28:42 +0000 (00:28 -0300)]
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>
Lars-Peter Clausen [Sat, 27 Aug 2011 16:24:13 +0000 (18:24 +0200)]
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>
Lars-Peter Clausen [Sat, 27 Aug 2011 16:24:12 +0000 (18:24 +0200)]
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>
Peter Ujfalusi [Fri, 26 Aug 2011 13:33:52 +0000 (16:33 +0300)]
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>
Mark Brown [Fri, 19 Aug 2011 08:53:12 +0000 (17:53 +0900)]
MAINTAINERS: Add some missed Wolfson files
Mostly input related.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kristian Amlie [Fri, 26 Aug 2011 11:19:49 +0000 (13:19 +0200)]
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>
Takashi Iwai [Fri, 26 Aug 2011 07:29:43 +0000 (09:29 +0200)]
Merge branch 'fix/asoc' into for-linus