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Commit Graph

6084 Commits

Author SHA1 Message Date
Clemens Ladisch
18dd0aa5af sound: snd_ctl_remove_user_ctl: prevent removal of kernel controls
Ensure that userspace can remove only user controls.  Controls created
by kernel drivers must not be removed because they might be referenced
in calls to snd_ctl_notify().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:21 +02:00
Clemens Ladisch
f217ac59b6 sound: snd_ctl_remove_unlocked_id: simplify user control counting
Move the decrementing of the user controls counter from
snd_ctl_elem_remove to snd_ctl_remove_unlocked_id; this saves the
separate locking of the controls semaphore, and therefore removes
a harmless race.

Since the purpose of the function is to operate on user controls (the
control being unlocked is just a prerequisite), rename it to
snd_ctl_remove_user_ctl.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:15 +02:00
Clemens Ladisch
317b80817f sound: snd_ctl_remove_unlocked_id: simplify error paths
Use a common exit path to release the mutex and to return a possible
error.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:06 +02:00
Clemens Ladisch
2a031aedf7 sound: snd_ctl_elem_add: fix value count check
Make sure that no user element that has no values can be added.

The check for count>1024 is not needed because the count is checked
later for the individual control types.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:00 +02:00
Janusz Krzysztofik
471e3dec3a ASoC: OMAP: Enhance OMAP1510 DMA progress software counter
Enhance period_index accuracy, particularly just before buffer rewind, by
making use of DMA interrupt status flags in addition to simply counting up
interrupts.

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 11:00:34 +01:00
Janusz Krzysztofik
64844a6ac8 ASoC: OMAP: Make use of DMA channel self linking on OMAP1510
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP
models. Remove unnecessary DMA transfer restart from interrupt handler
routine.

The interrupt routine used to maintain a period index, originally needed for
counting up periods up to a full buffer in order to restart the DMA transfer.
For some time, this counter is also used as a replacement for hardware DMA
progress counter that has been found unusable on OMAP1510 in case of playback.
Thus, the period index calculation cannot be omitted completely. However, the
accuracy of this counter can still suffer from missing DMA interrupts.

In order to work correctly, it requires patch 1 from this series also applied:
[RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 10:59:59 +01:00
Mark Brown
1e97f50b70 ASoC: Factor out cache I/O from WM8974
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 12:15:10 +01:00
Mark Brown
37cfa1950e Merge branch 'wm8974-upstream' into for-2.6.32 2009-08-15 11:52:43 +01:00
Mark Brown
29e02cb3ff ASoC: Hook i.MX into build
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:37:30 +01:00
Mark Brown
d555a552ae ASoC: Staticise unexported variables
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:36:49 +01:00
Mark Brown
a2d512a978 ASoC: Remove unneeded i.MX dependency on SND
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:36:20 +01:00
Mark Brown
08229de4b4 Merge branch 'for-2.6.32' into mxc
Conflicts:
	sound/soc/Makefile
2009-08-15 11:20:44 +01:00
Takashi Iwai
7570ef1834 ALSA: hda - Add missing num_adc_nids definition for IDT92HD8xxx
The previous fix removed the definition of num_adc_nids wrongly, and
this resulted in the missing input-source control.  Now readded again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-15 11:57:53 +02:00
Barry Song
2a708137fd ASoC: delete -spi suffix in ad1938 and free private data while registers fail
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:53:02 +01:00
Peter Ujfalusi
9028935d75 ASoC: TWL4030: Fix for capture mixer strings
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:52:59 +01:00
Mark Brown
d91e9a7ab9 ARM: S3C24XX: Add platform device for AC97 controller
Move the definition of the "generic" IRQ in the process.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
2009-08-14 01:13:29 +01:00
Marek Vasut
4ac0478f2a ALSA: Allow passing platform_data for pxa2xx-ac97
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:37 +01:00
Chaithrika U S
30230f4cd7 ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:36 +01:00
Chaithrika U S
517ee6cf69 ASoC: DaVinci: Add a DAI format to McASP driver
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:35 +01:00
Chaithrika U S
6a99fb5fb8 ASoC: DaVinci: McASP driver enhacements
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf

This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:35 +01:00
Mark Brown
a2342ae325 ASoC: Factor out shared code from WM8993
The WM8993 analogue control is shared with other devices in the same
product line.  Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:31 +01:00
Takashi Iwai
667067d898 ALSA: hda - Fix / clean up IDT92HD83xxx codec parser
A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
  MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
  capture volumes

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-13 18:14:42 +02:00
Takashi Iwai
a6cd7a71fd Merge branch 'topic/hda-dmic-fix' into topic/hda 2009-08-13 18:14:02 +02:00
Mark Brown
e9ade7f933 ASoC: Minor cleanups to AD1938 driver
- Build in SND_SOC_ALL_CODECS.
- Remove null suspend/resume stuff.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 15:19:42 +01:00
Barry Song
7eaae41ea5 new ad1836 codec driver based on asoc
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 15:18:53 +01:00
Peter Ujfalusi
9008adf9a9 ASoC: TWL4030: Introduce PGAs for outputs
Dynamically control and control only the needed output amplifier
muting/un-muting.

The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.

Move these as separate PGA so only the needed amplifier will be touched.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 14:56:13 +01:00
Barry Song
c4ff357ada ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM work
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 10:47:22 +01:00
Takashi Iwai
1c4bdf9be0 ALSA: hda - Enable line-out detection only with speakers
Enable line-out detection for IDT/STAC codecs only when speaker pins
exist.  In some cases, the speaker itself is identified as line-out,
and this confuses the situation.  Only the extra line-outs should do
auto-muting.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-13 08:27:38 +02:00
Tim Blechmann
c18bc9b927 ALSA: hdsp - allow proc reporting with disconnected io box
the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-12 18:21:30 +02:00
Mark Brown
d2a382143b ASoC: Update AD1938 for new TDM slot API
It's only actually paying attention to the slot count anyway.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-12 14:30:33 +01:00
Takashi Iwai
8884be98bc Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Don't override ADC definitions for ALC codecs
  ALSA: hda - Add missing vmaster initialization for ALC269
2009-08-12 08:05:20 +02:00
Takashi Iwai
909a2607a5 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Add missing DRV_NAME definitions for fsl/* drivers
2009-08-12 08:05:19 +02:00
Herton Ronaldo Krzesinski
5908589f31 ALSA: hda - fix noise issue when recording from digital mic with alc268
With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).

Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-12 07:35:02 +02:00
Takashi Iwai
4f5d170620 ALSA: hda - Clean up init and setup hooks for Realtek codecs
Move static codes to setup from init_hook for each model.

Also, use the common auto-mic selection helper for devices that support
auto-mic selection.  They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 18:17:46 +02:00
Mark Brown
e0026beac0 ASoC: Update WM9081 for tdm_slot() API change
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.

Also allow 0 TDM slots to be configure (for use when disabling TDM).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-11 16:29:21 +01:00
Takashi Iwai
e9c364c04f ALSA: hda - Add setup hook to ALC preset struct
Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 17:16:13 +02:00
Takashi Iwai
4d8e22e0f6 ALSA: hda - Add a white-list for MSI option
Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems.  Simply using a quirk list.
As the first case, take HP Compaq CQ40.

Reference: Novell bnc#529971
	https://bugzilla.novell.com/show_bug.cgi?id=529971

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 14:25:46 +02:00
Mark Brown
1921bab217 Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 2009-08-11 13:09:27 +01:00
Randy Dunlap
17244c24f9 ASoC: fix I2C build errors
Fix soc build errors when I2C is built as a loadable module:

(.text+0x5d26b): undefined reference to `i2c_master_send'
soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer'

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-11 10:47:25 +01:00
Takashi Iwai
b59bdf3b0c ALSA: hda - Check connectivity for auto-mic of Realtek codecs
Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC.  Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.

This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.

Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 09:47:30 +02:00
Takashi Iwai
52b5deefbb Merge branch 'fix/hda' into topic/hda 2009-08-11 08:47:38 +02:00
Takashi Iwai
dd704698f5 ALSA: hda - Don't override ADC definitions for ALC codecs
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition.  This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).

This patch avoids the overriding by adding the proper checks.

Reference: Novell bnc#529467
	https://bugzilla.novell.com/show_bug.cgi?id=529467

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 08:45:11 +02:00
Takashi Iwai
f1e6d3c5cf ALSA: usb-audio - Fix types taken in min()
Fix the compile warning due to different integer types used in min():
  sound/usb/usbaudio.c: In function 'init_substream_urbs':
  sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 08:16:15 +02:00
Takashi Iwai
2a22d3f812 ALSA: hda - Use only one capture stream for auto-mic
When the auto-mic feature is enabled, we should support only one
capture stream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 18:56:05 +02:00
Takashi Iwai
6c81949227 ALSA: hda - Add auto-mic support for Realtek codecs
Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode).  The auto-mic mode is enabled
only when one internal mic and one external mic are present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 18:47:44 +02:00
Tejun Heo
93fe4483e6 sound: make OSS device number claiming optional and schedule its removal
If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases.  This feature is redundant as chrdev already has such
mechanism.  This preemptive claiming prevents alternative OSS
implementation.

The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.

This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel.  The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.

As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.

For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .

Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:59:36 +02:00
Mark Brown
e0c48a18f7 ASoC: Drop unneeded declaration of removed wm8731 SPI write function
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-10 12:59:29 +01:00
Tejun Heo
0a848680a8 sound: request char-major-* module aliases for missing OSS devices
Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded.  The custom module aliases don't
have any inherent benefit than backward compatibility.

sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.

Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:59:26 +02:00
Clemens Ladisch
5e8e7c3853 sound: fix OSS MIDI output data loss
In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call.  Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:15:43 +02:00
Clemens Ladisch
6e2efaacb3 sound: ymfpci: increase timer resolution to 96 kHz
Allow the interval timer to be programmed with its full 96 kHz
precision.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:14:46 +02:00
Clemens Ladisch
765e8db078 sound: usb-audio: do not make URBs longer than sync packet interval
Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:13:56 +02:00
Takashi Iwai
d5c9c8912a Merge branch 'fix/hda' into topic/hda 2009-08-10 11:58:09 +02:00
Takashi Iwai
100d5eb36b ALSA: hda - Add missing vmaster initialization for ALC269
Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.

Reference: Novell bnc#527361
	https://bugzilla.novell.com/show_bug.cgi?id=527361

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-08-10 11:57:05 +02:00
Takashi Iwai
da2a2aaa8e ALSA: hda - Fix Oops due to STAC/IDT auto-mic changes
The previous auto-mic patch for STAC/IDT codecs causes the Oops on
machines without digital mic pins.  This patch fixes the problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 07:44:09 +02:00
Mark Brown
35b1207b34 ASoC: Convert WM8776 to use factored out register cache code
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-08 10:37:33 +01:00
Chaithrika U S
7ae5945f0c ASoC: DaVinci: Support Audio on DA830 EVM
Add support for audio on DA830 EVM- here McASP1 is interfaced to
TLV320AIC3106 codec.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-08 09:12:54 +01:00
Uwe Kleine-König
dbe9ea6e79 ASoC: s3c2443-ac97: convert semaphore to mutex
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig):

	  CC [M]  sound/soc/s3c24xx/s3c2443-ac97.o
	sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX'
	sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration
	sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read':
	sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function)
	sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once
	sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.)
	sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write':
	sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function)

Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-08 08:50:13 +01:00
Takashi Iwai
afc5e65245 ASoC: Add missing DRV_NAME definitions for fsl/* drivers
Module builds are broken due to missing DRV_NAME for
efika-audio-fabric and pcm030-audio-fabric.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-07 16:33:53 +02:00
Janusz Krzysztofik
b7b8f9bf0c TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253
The patch changes the line discipline name registered in include/linux/tty.h
and updates the ams-delta machine driver to use it.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 11:48:02 +01:00
Mark Brown
06cddefc1f Merge branch 'reg-cache' into for-2.6.32 2009-08-07 11:43:58 +01:00
Mark Brown
b9b5cc26d0 Merge branch 'for-2.6.31' into for-2.6.32 2009-08-07 11:42:01 +01:00
Troy Kisky
6a90d536fe ASoC: DaVinci: pcm, constrain buffer size to multiple of period
The dma setup code assumes that the buffer size is a multiple
of the period size.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 11:38:29 +01:00
Troy Kisky
9bb7415056 ASoC: DaVinci: i2s: don't bounce through rtd to get dai
dai is a parameter to the functions, so use it instead of
looking it up.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 11:38:29 +01:00
Jarkko Nikula
c12abc012e ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop
Simultaneous audio playback and capture on OMAP1510 can cause that second
stream is stalled if there is enough delay between startup of the audio
streams.

Current implementation of the omap_mcbsp_start is starting both transmitter
and receiver at the same time and it is called only for firstly started
audio stream from the OMAP McBSP based ASoC DAI driver.

Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is
missed if there is no DMA transfer set up at that time when the first word
after McBSP startup is transmitted. The problem hasn't noted before since
later OMAPs are using level sensitive DMA request lines.

Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by
allowing to start and stop individually McBSP transmitter and receiver
logics. Then call those functions individually for both audio playback
and capture streams. This ensures that DMA transfer is setup before
transmitter or receiver is started.

Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem
analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA
request line behavior differences between the OMAP generations.

Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 10:57:42 +01:00
Daniel Ribeiro
a5479e389e ASoC: change set_tdm_slot api to allow slot_width override.
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.

Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.

While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).

(this series is meant for Mark's for-2.6.32 branch)

Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 15:52:24 +01:00
Janusz Krzysztofik
9029bb316b ASoC: CX20442: simplify codec controller usage
This patch is a workaround for the problem of several subsequent control
statements not being applied correctly to the codec controller (modem).

In order to follow the hook switch state change from handset to handsfree
while
in full duplex mode, two consecutive +VLS control commands were sent to the
modem. The first one was M1 (microphone only), the seconds one was M1S1 (both
microphone and speaker). As there was no real modem handshaking procedure
implemented, neither in the codec nor in the machine driver part of the line
discipline, the modem was having the second command missed.

Since a possibility to switch to microphone only mode (and speaker only mode
as well) seams of no value, I have modified the code to issue single M1S1
command only for any of those cases.

Tested on my Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 11:36:46 +01:00
Janusz Krzysztofik
4977b03e3d ASoC: CX20442: add some debugging
This patch adds debugging statement that can help in tracing
how the driver is trying to control the codec device.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 11:36:45 +01:00
Mark Brown
924914ee95 ASoC: Add WM8776 CODEC driver
The WM8776 is a high performance, stereo audio CODEC with five channel
input selector. The WM8776 is ideal for surround sound processing
applications for home hi-fi, DVD-RW and other audio visual equipment.

This driver implements support for most WM8776 features - currently the
ADC automatic level control/limiter functionality is omitted.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 11:36:45 +01:00
Takashi Iwai
1972d02504 ALSA: hda - Add quirks for some HP laptops
The new HP laptops have PCI SSID 103c:701x and requires model=hp-dv5.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-06 08:44:43 +02:00
javier Martin
fbb474deda ASoC: Fix review issues in i.MX2x PCM driver
Signed-off-by: javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:55 +01:00
javier Martin
2ccafed43a ASoC: add machine driver for i.mx27_visstrim_m10 board
This adds support for i.mx27_visstrim_sm10 board machine driver which
uses an i.mx27 processor plus a wm8974 codec.

It has been tested on a visstrim_sm10 board.

Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:54 +01:00
javier Martin
9d8bc2968c ASoC: add DAI platform ssi driver for MXC
This adds support for DAI platform for the SSI present in MXC platforms.

It currently does not support i.MX3, the only thing necessary to do
this is to export DMA data for i.MX3 interface which I haven't done
because I don't have a i.MX3 based board available.

It has been tested on i.MX27 board.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:54 +01:00
javier Martin
fd6a6394d7 ASoC: add DMA platform driver for MX1x and MX2x
This adds support for DMA platform valid for i.MX1 and i.MX2 platforms.

This is not valid for i.MX3 since it doesn't share the same DMA
interface than i.MX1 and i.MX2.

It has been tested on i.MX27 board.

Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:54 +01:00
Daniel Mack
15b5bdaeeb ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
Power management for the cs4270 codec is currently implemented as part
of the i2c_driver struct. The disadvantage of doing it this way is that
the callbacks registered in the snd_soc_card struct are called _before_
the codec's callbacks.

That doesn't work, because the snd_soc_card callbacks will most likely
switch down the codec's power domains or pull the reset GPIOs, and
hence make the i2c communication bail out.

Fix this by binding the suspend and resume code to the
snd_soc_codec_device driver model and let the I2C functions only call
the SoC core function for resume and suspend, which do nothing currently
but will do later.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:13:49 +01:00
John Bonesio
b0a2712ffd ASoC: MPC5200: Support for buffer wrap around
The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that
s->runtime->control->appl_ptr can wrap around to the beginning of the
buffer. This change fixes this problem.

Signed-off-by: John Bonesio <bones@secretlab.ca>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:10:14 +01:00
Mark Brown
4bc4c9a5f5 ASoC: Existing S3C24xx AC97 drivers should depend on S3C24xx
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 17:15:04 +01:00
Linus Torvalds
6ce90c430b Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Read buffer overflow
  ALSA: hda: Correct EAPD for Dell Inspiron 1525
  ALSA: hda: warn on spurious response
  ALSA: hda: remember last command for each codec
  ALSA: hda: read CORBWP inside reg_lock
  ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
  ALSA: hda: take cmd_mutex in probe_codec()
  ALSA: hda: track CIRB/CORB command/response states for each codec
  ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
2009-08-04 15:39:55 -07:00
Takashi Iwai
16ffe32c77 ALSA: hda - Fix line-out jack handling with STAC/IDT codec
When the line-out jack is plugged/unplugged, the driver needs to check
the headphone plug, not only the line-out jack itself.  Otherwise the
headphone or the speaker may be wrongly muted/unmuted.

As a result, both STAC_HP_EVENT and STAC_LO_EVENT need to call the
same function, stac92xx_hp_detect().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-04 13:40:54 +02:00
Mark Brown
27ded041f0 ASoC: Factor out 7 bit register 9 bit data SPI write
This converts all the Wolfson drivers using this format (the only devices
that do) except WM8753 to use it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:16 +01:00
Mark Brown
8d50e447d1 ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:16 +01:00
Mark Brown
afa2f1066e ASoC: Factor out I2C 8 bit address 16 bit data I/O
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:15 +01:00
Mark Brown
7084a42b96 ASoC: Add I/O control bus information to factored out cache setup
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.

Initially just use this to factor out hw_write_t for I2C.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:09 +01:00
Takashi Iwai
15cfa2b3db ALSA: hda - Fix line-out jack detection
The commit fefd67f31e
    ALSA: hda - Add line-out jack detection on IDT/STAC codecs
enabled wrong pins for jack detections.  Fixed to the correct ones.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 14:23:33 +02:00
Lubomir Rintel
51840409b6 ALSA: pcm - Tell user that stream to be rewound is suspended
Return STRPIPE instead of EBADF when userspace attempts to rewind
of forward a stream that was suspended in meanwhile, so that it
can be recovered by snd_pcm_recover().

This was causing Pulseaudio to unload the ALSA sink module under a race
condition when it attempted to rewind the stream right after resume from
suspend, before writing to the stream which would cause it to revive the
stream otherwise. Tested to work with Pulseaudio patched to attempt to
snd_pcm_recover() upon receiving an error from snd_pcm_rewind().

Signed-off-by: Lubomir Rintel <lkundrak@v3.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:57:00 +02:00
Takashi Iwai
e44d4e4cee Merge branch 'fix/hda' into topic/hda 2009-08-03 08:37:17 +02:00
Wu Fengguang
559059b27f ALSA: hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin number
The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes.
Here we assume only the first ones will be used.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:35:05 +02:00
Roel Kluin
4b35d2ca23 ALSA: hda - Read buffer overflow
Check whether index is within bounds before testing the element.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:34:06 +02:00
Chengu Wang
84d3dc200f ALSA: hda: Correct EAPD for Dell Inspiron 1525
The commit 24918b61b5 statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.

The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.

Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:30:56 +02:00
Wu Fengguang
e310bb0646 ALSA: hda: warn on spurious response
To help disclose hardware bugs.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:27:53 +02:00
Wu Fengguang
feb273404f ALSA: hda: remember last command for each codec
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:27:39 +02:00
Wu Fengguang
c32649feb4 ALSA: hda: read CORBWP inside reg_lock
This converts the last CORBWP access outside of reg_lock.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:55 +02:00
Wu Fengguang
cdb1fbf231 ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
Just for safety.  azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:42 +02:00
Wu Fengguang
a678cdee25 ALSA: hda: take cmd_mutex in probe_codec()
Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.

So cmd_mutex would be a safe addition to probe_codec().

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:23 +02:00
Wu Fengguang
deadff1665 ALSA: hda: track CIRB/CORB command/response states for each codec
Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.

Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:13 +02:00
Takashi Iwai
ce577e8cf5 ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.

Reference: Novell bnc#526325
	https://bugzilla.novell.com/show_bug.cgi?id=526325

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:23:52 +02:00
Janusz Krzysztofik
6d7f68a1ea ASoC: add support for Amstrad E3 (Delta) machine
This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC.

Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.

Depends on:
1) latest version of the CX20442 codec driver that exposes v253_ops
   structure[1],
2) patch 2/3 form this series: TTY: Add definition of a new line
   discipline required by Amstrad E3 (Delta) ASoC driver[2].

CPU DAI parameters best matching the codec DAI has been selected out
empirically for best user experience.

Board specific audio function control (with related DAPM widgets) has been
modeled after empirically discovered codec capabilities.

Unlike other ASoC machine drivers, this one makes use of a codec provided line
discipline that is required for talking to a modem chip that can control the
codec behavoiur. As the line discipline operations must call board specific
bits as well, the machine driver registers its own line discipline ops, not
the codec provided, and then calls those codec provided from inside its own
callbacks.
If some kind of a glue, like a bus over a tty, exsited that could help in
runtime detection of a modem (bus adapter) over a more generic line discipline
(bus driver)[3], the line discipline code could be probably designed in a
more generic way.

In order to work at all, this driver requires a working McBSP1. On OMAP1510
based machines (not sure if other OMAP1 variants as well), where McBSP1 is a
DSP public peripheral, that means the kernel must provide basic DSP support,
ie. omap_dsp_init(), in order to power up the DSP. This used to be included in
linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c.
Without that, the driver would not work, ie. not shift in/out any bits over
the CPU DAI[4]. This limitation is not board, but CPU specific, and may apply
to other code that makes use of McBSP1/McBSP3 on affected machines. I provide
an extra patch (4/3) as a temporary solution.

To work correctly in playback mode, this driver requires my prevoiusly
submitted patch that corrects pcm pointer calculation for OMAP1510 based
machines[5] (already included in linux-2.6.31-rc3).

To support codec controls, this driver requires my previously submitted patch
that adds support for modem found on Amstrad Delta[6].

[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.html
[2] http://www.spinics.net/lists/linux-serial/msg01862.html
[3] http://www.spinics.net/lists/linux-serial/msg01856.html
[4] http://www.spinics.net/lists/linux-omap/msg15114.html
[5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html
[6] http://www.spinics.net/lists/linux-omap/msg15432.html

Credits to:
Mark Underwood - for his initial, omap-alsa based sound driver for
this machine,
Mark Brown - for his help, patience and excellent subsytem maintainer support.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 22:38:44 +01:00
Janusz Krzysztofik
ad120dae12 ASoC: CX20442: push down machine independent line discipline bits
This corrected patch adds machine independent line discipline code, prevoiusly
exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442
codec driver. The code can be used as a standalone line discipline, or as a
set of codec specific functions called from machine's line discipline
callbacks. Anyway, the line discipline itself must be registered by a machine
driver.

Applies on top of the followup to my initial driver version:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html

Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 22:38:44 +01:00
Lars-Peter Clausen
b8e22c1fe3 ASoC: jack: Fix race in snd_soc_jack_add_gpios
The irq can fire as soon as it has been requested, thus all fields accessed
from within the irq handler must be initialized prior to requesting the irq.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 22:38:43 +01:00
Mark Brown
77ee09c67e ASoC: Allow CODECs to flag invalid registers
This helps CODECs with sparse register maps work better with the
register cache display interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 18:54:48 +01:00
Takashi Iwai
ec86fe5209 Merge branch 'fix/oss' into for-linus
* fix/oss:
  sound: mpu401.c: Buffer overflow
  sound: aedsp16: Buffer overflow
2009-07-31 10:17:45 +02:00
Takashi Iwai
d62e345f14 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: sound/aoa: Add kmalloc NULL tests
2009-07-31 10:17:44 +02:00
Takashi Iwai
6280b61af5 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Increase PCM stream name buf in patch_realtek.c
  ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
  ALSA: hda - Add quirk for Dell Studio 1555
2009-07-31 10:17:42 +02:00
Julia Lawall
f065fabc86 ALSA: sound/aoa: Add kmalloc NULL tests
Check that the result of kzalloc is not NULL before a dereference.

The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@
expression *x;
identifier f;
constant char *C;
@@

x = \(kmalloc\|kcalloc\|kzalloc\)(...);
... when != x == NULL
    when != x != NULL
    when != (x || ...)
(
kfree(x)
|
f(...,C,...,x,...)
|
*f(...,x,...)
|
*x->f
)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-31 10:14:58 +02:00
Takashi Iwai
aa563af763 ALSA: hda - Increase PCM stream name buf in patch_realtek.c
The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo".  Now the size is doubled.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-31 10:06:34 +02:00
Takashi Iwai
03cb2dafcb Merge branch 'topic/hda-cirrus' into topic/hda 2009-07-30 18:09:04 +02:00
Takashi Iwai
d195658bd7 Merge branch 'fix/hda' into topic/hda 2009-07-30 18:08:54 +02:00
Takashi Iwai
fefd67f31e ALSA: hda - Add line-out jack detection on IDT/STAC codecs
Add the automatic mute of speakers via line-out jack plugging on
STAC/IDT codecs.  The feature is enabled when the HP detect is present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 18:08:46 +02:00
Stelian Pop
3a38516750 ALSA: hda - Enable HP output with Macbook Pro 5, 5
The patch below, to be applied on the latest sound-unstable-2.6.git,
enables headphones output on my MacBookPro 5,5, together with the
automuting feature.

Here is the exact soundcard id:
	Vendor Id: 0x10134206
	Subsystem Id: 0x106b4d00
	Revision Id: 0x100301

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 14:57:25 +02:00
Mark Brown
a1daf67d72 Merge branch 'gta02-audio' into for-2.6.32 2009-07-30 13:21:38 +01:00
Takashi Iwai
5207e10ed4 ALSA: hda - Integrate Digital Input Source to Input Source
STAC/IDT codecs provide both "Input Source" and "Digital Input Source"
controls to choose the analog input source and the digital input source.
But this is far user-unfriendly.

This patch merges the input source selections into one "Input Source"
control.  To have separate digital and analog input source controls,
you can pass "separate_dmux = 1 " hint string.

At the same time, this patch gets rid of analog mixer stuff that was
already disabled in previous patches.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 13:09:08 +02:00
Takashi Iwai
bf677bd8fb ALSA: hda - Fix typos of Capture controls.
The commit 6479c63188
    ALSA: hda - Create Capture controls dynamically
introduced typos of "Capture".  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 09:24:29 +02:00
Wu Fengguang
6732bd0d15 ALSA: hda: add HP automute support to Intel ALC889/ALC889A models
It auto mutes all 8-channel outputs at rear panel when
the front panel headphone is connected.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 09:19:14 +02:00
Wu Fengguang
dd7714c922 ALSA: hda: add 2-channel mode to Intel ALC889/ALC889A models
This 2-channel mode is useful in that it will broadcast
a 2-channel audio stream to all front/side/... ports.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 08:53:56 +02:00
Barry Song
3a39f832a5 ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm dai
1. fix "line over 80 characters" checkpatch warnings
2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead
3. fix typos

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-29 21:31:53 +01:00
Lars-Peter Clausen
82c4362ee3 ASoC: neo1973_gta02_wm8753: Replace deprecated s3c_gpio calls with gpiolib
With the s3c platform has implementing gpiolib support the s3c_gpio api has been
deprecated.
This patch gets rid of all s3c_gpio calls and replaces them by using gpiolib.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-29 21:29:33 +01:00
Lars-Peter Clausen
69331fbdee ASoC: neo1973_gta02_wm8753: Replace snd_soc_cnew with snd_soc_add_controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-29 21:29:33 +01:00
Takashi Iwai
71443b0b74 ALSA: hda - No analog mix input source as default for IDT92HD71bxx
The analog mix is disabled now as default (unless "analog_mixer" hint
is given), so it shoudn't appear in the digital input source as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 18:41:29 +02:00
Takashi Iwai
15b4f296fc ALSA: hda - Add missing DMUX initialization for auto-mic with STAC/IDT
Added the missing initialization of DMUX connection (to analog input)
for auto-mic mode with STAC/IDT codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 16:32:55 +02:00
Takashi Iwai
26a2798053 ALSA: hda - Remove static connection in IDT 92HD71bxx
We don't need any more static connection to the port F (which is often
used for docking stations) since its connection is done dynamically via
DAC assignment now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 16:30:14 +02:00
Roel Kluin
a987004fbc sound: mpu401.c: Buffer overflow
mpu_synth_info[m].name is a char[30], and the minimum length of the data
written by sprintf is 31 bytes including terminating null.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:41:24 +02:00
Roel Kluin
c45ec06c74 sound: aedsp16: Buffer overflow
DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes
including terminating null.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:37:12 +02:00
Roel Kluin
78735cffc2 ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:35:20 +02:00
Takashi Iwai
3d21d3f7e7 ALSA: hda - Support auto-mic switching with IDT/STAC codec
Support the automatic mic-switching with some devices with IDT/STAC
codecs.  The condition is that the device has only two inputs, one
for an external mic and one for an internal mic.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:32:56 +02:00
Takashi Iwai
62558ce157 ALSA: hda - Avoid overwrite of jack events with STAC/IDT
Since only one event can be associated to a (pin) widget, it's safer
to avoid the multiple mapping.  This patch fixes the behavior of the
STAC/IDT codec driver.

Now stac_get_event() doesn't take the type argument but simply returns
the first hit element.  Then enable_pin_detect() checks the validity
of the type, and returns non-zero only if a valid entry.  The caller
can call stac_issue_unsol_event() after checking the return value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:28:37 +02:00
Barry Song
c8489c3ed3 ASoC: board driver to connect bf5xx with ad1938
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-28 22:16:02 +01:00
Barry Song
01e2ab207c ASoC: blackfin I2S(TDM mode) CPU DAI driver
The I2S DAI driver for blackfin SPORT, but works in TDM mode.
I2S is not a special case of TDM with only left and right two slots for
SPORT interface. I2S coordinates with TDM in SPORT, but not a part of
TDM. TDM require different hardware configuration with I2S, not only
different slot number.  One is "Stereo Serial Operation" mode of SPORT,
the other one is "Multichannel Operation" mode. They are incompatible
at the same time.
Hardware and DMA description and data transfer flow are much different
for I2S and TDM. Merging them as a whole will be very ugly and difficult
to maintain.
So we don't define a new DAI type, but give two DAI instances for standard
I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the
I2S-family DAI type.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-28 22:15:27 +01:00
Janusz Krzysztofik
b84eab08a6 ASoC: CX20442: fix issues pointed out by subsystem maintainer
The patch fixes some checkpatch identified issues and adds a comment about
line discipline interaction to my driver code, as requested by Mark on my
inital submission (thank you Mark for applying my imperfect patch anyway).
It also fixes MODULE_ALIAS mismatch as used in my machine driver.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-28 22:15:23 +01:00
Takashi Iwai
50c62f068e ALSA: hda - Don't create analog mixer for IDT92HD71bxx
The analog mixer unit on IDT 92HD71Bxx codecs is almost useless
since we use only the direct connections from DAC to pin.

Remove the controls to avoid unneeded confusion as default now.
This can be still back via "analog_mixer = 1" hint.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 18:28:27 +02:00
Takashi Iwai
6479c63188 ALSA: hda - Create Capture controls dynamically
Instead of static snd_kcontrol_new arrays, create "Capture Volume"
and "Capture Switch" controls dynamically based on the mixer attr
values (made via HDA_COMPOSE_AMP_VAL()).
This reduces the code size and gives more flexibility to change
the number of controls later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 18:25:03 +02:00
Takashi Iwai
4417932315 ALSA: hda - Don't create unneeded digital input source for IDT 92HD71x
The current driver creates always the digital input source mixer
elements for IDT 92HD71x codecs no matter whether digital mics are
present.  This patch adds the proper check to avoid the creation of
these controls if unnecessary.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 17:03:49 +02:00
Takashi Iwai
9a11f1aa8e ALSA: hda - Reword information messages for BIOS auto-probing mode
The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.

Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 16:06:56 +02:00
Takashi Iwai
626f5cefc6 ALSA: hda - Add quirk for Dell Studio 1555
Added a quirk entry for Dell Studio 1555.

Reference: Novell bnc#525244
	https://bugzilla.novell.com/show_bug.cgi?id=525244

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 00:54:39 +02:00
Takashi Iwai
1ba7a7c650 ALSA: hda - Add exception for volume-knob in snd_hda_get_connections()
Volume-knob widgets may have connections even if they have no CONN_LIST
cap bit.  Allow the query exceptionally in snd_hda_get_connections().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-27 12:56:26 +02:00
Takashi Iwai
a22d543a95 ALSA: hda - Introduce get_wcaps_type() macro
Add a helper macro to retrieve the widget type from wiget cap bits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-27 12:54:26 +02:00
Takashi Iwai
57e4a5c4f8 Merge branch 'fix/usb-audio' into for-linus
* fix/usb-audio:
  ALSA: usb-audio - Volume control quirk for QuickCam E 3500
2009-07-26 11:07:08 +02:00
Takashi Iwai
b88158846f Merge branch 'fix/pcm-hwptr' into for-linus
* fix/pcm-hwptr:
  ALSA: pcm - Fix hwptr buffer-size overlap bug
  ALSA: pcm - Fix warnings in debug loggings
  ALSA: pcm - Add logging of hwptr updates and interrupt updates
  ALSA: pcm - Fix regressions with VMware
2009-07-26 11:07:07 +02:00
Takashi Iwai
de5d674c02 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Fix mute control with some ALC262 models
  ALSA: hda - Restore GPIO1 properly at resume with AD1984A
  ALSA: hda - Use snprintf() to be safer
2009-07-26 11:07:06 +02:00
Takashi Iwai
f35e2965b2 Merge branch 'fix/ctxfi' into for-linus
* fix/ctxfi:
  ALSA: ctxfi - Fix uninitialized error checks
2009-07-26 11:07:05 +02:00
Takashi Iwai
29769d533b Merge branch 'fix/caiaq' into for-linus
* fix/caiaq:
  ALSA: snd_usb_caiaq: add support for Audio2DJ
2009-07-26 11:07:04 +02:00
Takashi Iwai
7679d5c65b Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: tlv320aic3x: Enable PLL when not bypassed
2009-07-26 11:07:03 +02:00
Takashi Iwai
a3daf68931 Merge branch 'fix/hda' into topic/hda 2009-07-24 16:54:59 +02:00
Takashi Iwai
8de56b7deb ALSA: hda - Fix mute control with some ALC262 models
The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted.  This patch fixes
the issue.

Reference: Novell bnc#404873
	https://bugzilla.novell.com/show_bug.cgi?id=404873

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-07-24 16:54:31 +02:00
Marek Vasut
4ce2f2fe61 ASoC: Switch palm27x-asoc to jack detection api
This patch removes the old method of jack detection from palm27x-asoc
driver and adds jack detection api. It also removes some other (now)
useless stuff from the driver and corrects pin configuration for the
codec.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-24 11:29:03 +01:00
Janusz Krzysztofik
178b699c25 ASoC: Jack handling enhancements as suggested by subsystem maintainer
The patch adds a few small enhancements to the ASoC jack handling, as
suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes
for related bugs found while learning Mark's code and testing results.

Enhancements:
1. Update status of an ASoC jack while associating it with new gpios.
2. Really update DAPM pins while associating them with an ASoC jack.
3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes.

Fixes:
1. Apply mask on jack status report before using it, just for case.
2. While updating jack associated DAPM pins, use full resulting jack status,
   not the status report passed as an argument.

Created and tested on linux-2.6.31-rc3

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-24 11:28:59 +01:00
Daniel Mack
b30c494773 ALSA: snd_usb_caiaq: add support for Audio2DJ
This adds support for Native Instrument's freshly announced Audio2DJ
sound device hardware. Version number bumped to 1.3.19.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 16:31:58 +02:00
Takashi Iwai
947ca210f1 ALSA: pcm - Fix hwptr buffer-size overlap bug
The fix 79452f0a28 introduced another
bug due to the missing offset for the overlapped hwptr.
When the hwptr goes back to zero, the delta value has to be corrected
with the buffer size.  Otherwise this causes looping sounds.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 16:21:08 +02:00
Takashi Iwai
8935064043 ALSA: pcm - Fix warnings in debug loggings
Add proper cast.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 14:28:37 +02:00
Marek Vasut
474828a40f ALSA: Allow passing platform_data to devices attached to AC97 bus
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:30:56 +01:00
Joonyoung Shim
a7569afa8b ASoC: MAX9877: fix write operation for register
The MAX9877 needs an address of start register when we write values to
registers through i2c_master_send(), but the code for this was missed in
max9877_write_regs().

If the value of control is 0 in the max9877_set_out_mode(), the value is
not increased to 1, but actually the value to write to the register
should be 1.
And the register bits for out_mode and osc_mode should be cleared before
writing.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:08:30 +01:00
Janusz Krzysztofik
459dc35233 ASoC: Add support for Conexant CX20442-11 voice modem codec
This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.

This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.

The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.

Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.

Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.

[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html

Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:08:29 +01:00
Jaroslav Kysela
87a8c3702e ALSA: hda - Add better Intel IbexPeak platform support
Here are the new sound enabling patches for IbexPeak.

Summary of tested features:

  - playback
    - Front Headphone: OK
    - 8 channel audio: Front/Rear/CLFE/Side all OK

  - recording
    - Front Mic/Rear Mic: both OK
      (front/rear/line mics are selectable in the "Input source" alsamixer control)
    - Line In: not working
      (in 6ch mode, its amp/mute, direction and route all looks fine,
       so I'm a little puzzled)
      (hopefully no one will care this feature)

  - digital SPDIF input/output: not tested (no equipment)

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 11:25:42 +02:00
Takashi Iwai
cedb8118e8 ALSA: pcm - Add logging of hwptr updates and interrupt updates
Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.

For example,
	# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 11:09:03 +02:00
Mark Brown
c30853df98 Merge branch 'for-2.6.31' into for-2.6.32 2009-07-23 08:22:58 +01:00
Lopez Cruz, Misael
d756b27748 ASoC: OMAP: Staticise pcm creation function of omap-pcm
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 08:22:16 +01:00
Chaithrika U S
06c71282a9 ASoC: tlv320aic3x: Enable PLL when not bypassed
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.

Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.

Tested on DM6467 EVM, playback tested on DM355 EVM.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 08:14:29 +01:00
Takashi Iwai
4012ade933 ALSA: hda - Restore GPIO1 properly at resume with AD1984A
The commit 099db17e66 introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.

The fix is simple, use the cached write for storing GPIO data.

Reference: Novell bnc#522764
	https://bugzilla.novell.com/show_bug.cgi?id=522764

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 18:15:10 +02:00
Takashi Iwai
35ebf6e721 ALSA: ctxfi - Simple code clean up
- replace NULL == xxx with !xxx
- replace NULL != xxx with xxx
- similar trivial cleanups

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 17:12:34 +02:00
Takashi Iwai
89e1b9511f Merge branch 'fix/ctxfi' into topic/ctxfi 2009-07-22 17:06:53 +02:00
Takashi Iwai
68110661e8 ALSA: ctxfi - Fix uninitialized error checks
Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
  sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
  sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
  sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
  sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
  sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
  sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 17:05:15 +02:00
Jaroslav Kysela
16a433d8b3 ALSA: hda-intel: Cleanups for widget connection list handling
This patch adds a check to snd_hda_get_connections() routine for
presence of AC_WCAP_CONN_LIST. Also, make sure that negative error
codes from noted route are handled on all places as errors.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 16:30:03 +02:00
Takashi Iwai
86de741660 ALSA: hda - Use snprintf() to be safer
Use snprint() for creating the jack name string instead of sprintf()
in patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 16:02:46 +02:00
Alexey Fisher
2cf313ee75 ALSA: usb-audio - Volume control quirk for QuickCam E 3500
- E3500 report cval->max more than it actually can handel, so if you
set 95% capture level it will be silently muted.
- Betwen cval->min and cval-max(real) is 2940 control units,
but real are only 7 with cval->res = 384.
- Alsa can't handel less than 10 controls, so make it more
and set cval->res = 192.

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:52:15 +02:00
Jaroslav Kysela
254da007f9 ALSA: hda_generic: use AC_WCAP_CONN_LIST check for widget connections
Previous patch used widget type, but the presence flag of the connection
list is in the widget capabilities.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:39:56 +02:00
Jaroslav Kysela
0529604838 ALSA: hda_generic: do not read connections for widged with an unknown type
Reading node connections for an unknown widget can confuse HDA codec bus.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:39:44 +02:00
Takashi Iwai
1c20930a41 ALSA: hda - Fix ALC861 auto-mode parser
Fix the logic of ALC861 auto-mode parser for the outputs.
Instead of assuming the fixed DAC list, parse the conection and assign
the DAC dynamically.

Also, unmute the unused output connections to avoid noises on inputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:36:25 +02:00
Takashi Iwai
79452f0a28 ALSA: pcm - Fix regressions with VMware
VMware tends to report PCM positions and period updates at utterly
wrong timing.  This screws up the recent PCM core code that tries
to correct the position based on the irq timing.

Now, when a backward irq position is detected, skip the update
instead of rebasing.  (This is almost the old behavior before
2.6.30.)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 12:55:56 +02:00
Takashi Iwai
05ff7e11b7 ALSA: hda - Reduce click noise at power-saving
Add some tricks to reduce the click noise at powering down to D3
in the power saving mode on STAC/IDT codecs.
The key seems to be to reset PINs before the power-down, and some
delay before entering D3.  The needed delay is significantly long,
but I don't know why.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 12:39:24 +02:00
Joonyoung Shim
e458a48f87 ASoC: MAX9877: separate callback functions
The callback function to control register was used by whole controls in
MAX9877 driver, but this causes using many if statement for double
register control or invert.
So, the callback function for double register control is separate
differently, and the code for invert is added in the callback function.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-22 11:02:49 +01:00
javier Martin
25cbf46520 ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec.
This corrects a bug with ADC Inversion Switch in wm8974 codec.

Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-22 00:13:27 +01:00
John Bonesio
ed0f19b237 ASoC: MPC5200: Increase the delay time between resets
Reset was failing with the original udelay(50) between the code in
psc_ac97_cold_reset() and the call to psc_ac97_warm_reset(). Through testing
it was found that a delay of 1ms was necessary for the cold_reset code to
consistently complete successfully.

Signed-off-by: John Bonesio <bones@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-22 00:07:51 +01:00
Takashi Iwai
44f167d376 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: ca0106 - Fix the max capture buffer size
  ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
  ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
2009-07-21 19:03:22 +02:00
Takashi Iwai
a9d90c81b5 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
  ALSA: hda - Add quirk for Gateway T6834c laptop
  ALSA: hda_codec: Check for invalid zero connections
2009-07-21 19:03:20 +02:00
Takashi Iwai
36766835ed Merge branch 'fix/ctxfi' into for-linus
* fix/ctxfi:
  ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
2009-07-21 19:03:19 +02:00
Takashi Iwai
9c6c529a21 Merge branch 'fix/ctxfi' into topic/ctxfi 2009-07-20 17:08:01 +02:00
Frank Roth
55fe27f7e2 ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong. 
I double checked it with connector colors and creative soundblaster
windows drivers.

So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.

Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-20 17:07:10 +02:00
Takashi Iwai
34fdeb2d07 ALSA: ca0106 - Fix the max capture buffer size
The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.

This patch restricts the max buffer size less than that to make life
a bit easier.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-07-20 15:49:46 +02:00
Takashi Iwai
b04add9566 ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS.  Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.

This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides.  Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-07-20 15:12:41 +02:00
Takashi Iwai
bc5304b6fb ALSA: ctxfi - Native timer support for emu20k2
Added the native timer support for emu20k2, which gives much more
accurate update timing than the system timer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-20 13:41:35 +02:00
Hao Song
42b95f0c6b ALSA: hda - Add quirk for Gateway T6834c laptop
Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.

Signed-off-by: Hao Song <baritono.tux@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-20 09:05:53 +02:00
Jaswinder Singh Rajput
f96e080821 ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
When build SND_SEQUENCER in kernel then OSS sequencer(alsa_seq_oss_init)
is initialized before System (snd_seq_system_client_init) which leads to
memory leak :

unreferenced object 0xf6b0e680 (size 256):
  comm "swapper", pid 1, jiffies 4294670753
  backtrace:
    [<c108ac5c>] create_object+0x135/0x204
    [<c108adfe>] kmemleak_alloc+0x26/0x4c
    [<c1087de2>] kmem_cache_alloc+0x72/0xff
    [<c126d2ac>] seq_create_client1+0x22/0x160
    [<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
    [<c1485a05>] snd_seq_oss_create_client+0x86/0x142
    [<c1485920>] alsa_seq_oss_init+0xf6/0x155
    [<c1001059>] do_one_initcall+0x4f/0x111
    [<c14655be>] kernel_init+0x115/0x166
    [<c10032af>] kernel_thread_helper+0x7/0x10
    [<ffffffff>] 0xffffffff
unreferenced object 0xf688a580 (size 64):
  comm "swapper", pid 1, jiffies 4294670753
  backtrace:
    [<c108ac5c>] create_object+0x135/0x204
    [<c108adfe>] kmemleak_alloc+0x26/0x4c
    [<c1087de2>] kmem_cache_alloc+0x72/0xff
    [<c126f964>] snd_seq_pool_new+0x1c/0xb8
    [<c126d311>] seq_create_client1+0x87/0x160
    [<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
    [<c1485a05>] snd_seq_oss_create_client+0x86/0x142
    [<c1485920>] alsa_seq_oss_init+0xf6/0x155
    [<c1001059>] do_one_initcall+0x4f/0x111
    [<c14655be>] kernel_init+0x115/0x166
    [<c10032af>] kernel_thread_helper+0x7/0x10
    [<ffffffff>] 0xffffffff
unreferenced object 0xf6b0e480 (size 256):
  comm "swapper", pid 1, jiffies 4294670754
  backtrace:
    [<c108ac5c>] create_object+0x135/0x204
    [<c108adfe>] kmemleak_alloc+0x26/0x4c
    [<c1087de2>] kmem_cache_alloc+0x72/0xff
    [<c12725a0>] snd_seq_create_port+0x51/0x21c
    [<c126de50>] snd_seq_ioctl_create_port+0x57/0x13c
    [<c126d07a>] snd_seq_do_ioctl+0x4a/0x69
    [<c126d0de>] snd_seq_kernel_client_ctl+0x33/0x49
    [<c1485a74>] snd_seq_oss_create_client+0xf5/0x142
    [<c1485920>] alsa_seq_oss_init+0xf6/0x155
    [<c1001059>] do_one_initcall+0x4f/0x111
    [<c14655be>] kernel_init+0x115/0x166
    [<c10032af>] kernel_thread_helper+0x7/0x10
    [<ffffffff>] 0xffffffff

The correct order should be :

System (snd_seq_system_client_init) should be initialized before
OSS sequencer(alsa_seq_oss_init) which is equivalent to :

1. insmod sound/core/seq/snd-seq-device.ko
2. insmod sound/core/seq/snd-seq.ko
3. insmod sound/core/seq/snd-seq-midi-event.ko
4. insmod sound/core/seq/oss/snd-seq-oss.ko

Including sound/core/seq/oss/Makefile after other seq modules
fixes the ordering and memory leak.

Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-19 19:10:01 +02:00
Julia Lawall
fcb2954b96 ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
If spin_lock_irqsave is called twice in a row with the same second
argument, the interrupt state at the point of the second call overwrites
the value saved by the first call.  Indeed, the second call does not need
to save the interrupt state, so it is changed to a simple spin_lock.

The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@
expression lock1,lock2;
expression flags;
@@

*spin_lock_irqsave(lock1,flags)
... when != flags
*spin_lock_irqsave(lock2,flags)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-19 14:02:29 +02:00
Takashi Iwai
05e870d29a Merge branch 'fix/hda' into topic/hda 2009-07-19 13:52:31 +02:00
Jaroslav Kysela
2e9bf24706 ALSA: hda_codec: Check for invalid zero connections
To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-19 13:51:45 +02:00
Mark Brown
bca146578c ASoC: Fix checkpatch issues in AD1938
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-18 11:09:42 +01:00
Mark Brown
0c11f65555 ASoC: Fix FLL reference clock division setup in WM8993
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-17 22:13:01 +01:00
Mark Brown
8aa2df5308 ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.0
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL
configuration if the output frequency is zero.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-17 21:53:49 +01:00
Takashi Iwai
3f3b7c1aed ALSA: hda - Fix ALC268 parser for mono speaker
- Parse the mono output pin 0x16 correctly even as the primary output
- Create "Speaker" volume control if the primary output is a speaker
- Fix the wrong direction of (optional) "Mono" switch

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 14:36:59 +02:00
Takashi Iwai
82e1b804b0 ALSA: hda - Fix the previous sanity check in make_codec_cmd()
The newly added sanity-check for a codec verb can be better written
with logical ORs.  Also, the parameter can be more than 8bit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 12:47:34 +02:00
Candelaria Villareal, Jorge
c5910a7038 ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6
line, controlled by register INTBR_PMBR1. Machine driver takes care
of enabling gpio line through i2c and codec driver manipulates the
line during headset ramp up/down sequence.

Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-17 10:32:39 +01:00
Wu Fengguang
6430aeeb30 ALSA: hda - add bounds checking for the codec command fields
A recent bug involves passing auto detected >0x7f NID to codec command,
creating an invalid codec addr field, and finally lead to cmd timeout
and fall back into single command mode. Jaroslav fixed that bug in
alc880_parse_auto_config().

It would be safer to further check the bounds of all cmd fields.

Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 11:18:21 +02:00
Andiry Brienza
9176b672c2 ALSA: hda - Add support for new AMD HD audio devices
Add support for new AMD HD audio devices. Use generic driver to detect HD audio
devices with Vendor ID AMD.

Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 11:16:51 +02:00
Takashi Iwai
416c8fe3cd ASoC: Kill direct accesses to driver_data
Replaced with dev_{get|set}_drvdata().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 07:48:03 +02:00
Takashi Iwai
15c2ac051c Merge branch 'fix/usb-audio' into for-linus
* fix/usb-audio:
  sound: usb-audio: add workaround for Blue Microphones devices
2009-07-16 16:35:50 +02:00
Takashi Iwai
9d79b13691 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: riptide -  proper handling of pci_register_driver for joystick
2009-07-16 16:35:48 +02:00
Takashi Iwai
26887793b6 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking
2009-07-16 16:35:47 +02:00
Takashi Iwai
9d5b28d530 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free
2009-07-16 16:35:46 +02:00
Daniel Drake
0fb67e982a ALSA: hda - Add CX20582 and OLPC XO-1.5 support
This adds support for the Conexant CX20582 codec, based on code from
http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noarch.rpm.zip

This is the codec to be shipped in the OLPC XO-1.5, so this patch also
includes an XO-specific profile. Resultant configuration:
http://dev.laptop.org/~dsd/20090713/codec0.txt
http://dev.laptop.org/~dsd/20090713/codec0.svg

As the Linuxant code is structured differently to the other codecs,
I was unable to cleanly reimplement everything in the generic and Dell
profiles as more info is needed (e.g. codec graphs). I simplified those
profiles so that hopefully it will not break anyone's audio. If it does,
it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems,
and then fixing snd_hda_codec_configure() to fall back on the generic
parser, at least until support for other systems is figured out.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-16 16:03:08 +02:00
Barry Song
1274738d85 ASoC: new ad1938 codec driver based on asoc
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-16 11:14:39 +01:00
Kevin Hilman
3e46a44739 ASoC: davinci: don't use clock names
clock name strings are no longer passed on platform_data.  Instead,
we rely entirely on struct device and clkdev to find the right clock.

Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-16 10:59:52 +01:00
Joonyoung Shim
9db9ed977d ASoC: MAX9877: add MAX9877 amp driver
The MAX9877 combines a high-efficiency Class D audio power amplifier
with a stereo Class AB capacitor-less DirectDrive headphone amplifier.

The max9877_add_controls() is called to register the MAX9877 specific
controls on machine specific init() of the machine driver.

The datasheet for the MAX9877 can find at the following url:
http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf

[Slight edit to sort the ALL_CODECS entries -- broonie.]

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 16:59:31 +01:00