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Commit Graph

11091 Commits

Author SHA1 Message Date
Axel Lin
04f45c493a ASoC: wm8994: Slightly optimize configure_clock
snd_soc_update_bits() will only write new register value
if the old value is different from the new value.
In additional, snd_soc_update_bits() returns 0 for no change.
No need to read WM8994_CLOCKING_1 register before calling snd_soc_update_bits().

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 13:09:55 +01:00
Peter Ujfalusi
b1b6cffeb7 ASoC: omap-pcm: Fix the no period wakeup implementation
After omap_request_dma the BLOCK_IRQ is enabled as default
configuration for the channel.
If we are requested for no period wakeup, we need to disable
the BLOCK_IRQ in order to not receive any interrupts.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 12:03:50 +01:00
Axel Lin
9cd113261b ASoC: wm8988: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 12:03:49 +01:00
Axel Lin
c6d43417dd ASoC: wm8971: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 12:03:49 +01:00
Axel Lin
11b9ce622a ASoC: wm8711: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 12:03:48 +01:00
Ryan Mallon
c855a1a7ff ASoC: max98095 codec: Catch driver bugs for biquad channel name
Move the biquad channel names to a separate array and iterate over it in
max98095_get_bq_channel rather than duplicating the hardcoded channel
names. Add an error message if an invalid channel is passed and check
the error in the callers.

Also added a BUILD_BUG_ON to ensure that the bq_mode_name and controls
arrays are the same size.

Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 12:03:47 +01:00
Ryan Mallon
8754f2263f ASoC: max98088 codec: Catch driver bugs for eq channel name
Move the EQ channel names to a separate array and iterate over it in
max98088_get_channel rather than duplicating the hardcoded channel
names. Add an error message if an invalid channel is passed and check
the error in the callers.

Also added a BUILD_BUG_ON to ensure that the eq_mode_name and controls
arrays are the same size.

Signed-off-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 12:03:35 +01:00
Mark Brown
81204c84ca ASoC: Add WM1811 support
The WM1811 is mostly register compatible with the WM8994 and WM8958,
providing a high performance audio hub CODEC in a small form factor
suitable for ultra compact system designs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 11:59:46 +01:00
Dan Carpenter
f92766bc89 ALSA: oss-mixer - use strlcpy() instead strcpy()
This is mostly a static checker fix more than anything else.  We're
copying from a 64 char buffer into a 44 char buffer.

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

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

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

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

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

Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-04 07:16:31 +02:00
Mark Brown
11c2b5f2dc ASoC: Fix typo in 24.576MHz rate in WM5100
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 21:34:57 +01:00
Mark Brown
0b07ab9244 ASoC: Instantiate DAPM widgets before we do the DAI link init
The DAI init function may want to do something that needs the widgets to
be instantiated.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 21:34:56 +01:00
Axel Lin
f3a54a283b ASoC: samsung: s3c-i2s-v2.c needs module.h
Include <linux/module.h> to fix below build error:

  CC      sound/soc/samsung/s3c-i2s-v2.o
sound/soc/samsung/s3c-i2s-v2.c:573: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:573: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:573: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:638: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:638: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:638: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:677: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:677: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:677: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c: In function 's3c_i2sv2_register_dai':
sound/soc/samsung/s3c-i2s-v2.c:736: warning: initialization discards qualifiers from pointer target type
sound/soc/samsung/s3c-i2s-v2.c: At top level:
sound/soc/samsung/s3c-i2s-v2.c:754: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:754: warning: type defaults to 'int' in declaration of 'EXPORT_SYMBOL_GPL'
sound/soc/samsung/s3c-i2s-v2.c:754: warning: parameter names (without types) in function declaration
sound/soc/samsung/s3c-i2s-v2.c:756: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/s3c-i2s-v2.c:756: warning: data definition has no type or storage class
sound/soc/samsung/s3c-i2s-v2.c:756: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE'
sound/soc/samsung/s3c-i2s-v2.c:756: warning: function declaration isn't a prototype
make[3]: *** [sound/soc/samsung/s3c-i2s-v2.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 21:34:12 +01:00
David Henningsson
eb335a40ca 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>
2011-10-03 17:25:38 +02:00
Arnd Bergmann
61e49bf194 ASoC: samsung: WM8994 depends on MFD_WM8994
Any driver that selects SND_SOC_WM8994 should also make sure that
MFD_WM8994 is set, since the codec relies on the mfd code:

  sound/built-in.o: In function `wm8994_read':
  last.c:(.text+0x20160): undefined reference to `wm8994_reg_read'
  sound/built-in.o: In function `wm8994_write':
  last.c:(.text+0x20e68): undefined reference to `wm8994_reg_write'

This solves the problem by selecting the MFD driver directly
and adding extra 'depends on' statements to make sure that we
respect the dependencies of that driver.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 15:39:58 +01:00
Pierre-Louis Bossart
14bc52b8fe 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>
2011-10-03 15:48:12 +02:00
Axel Lin
9b5999b1bc ASoC: Fix setting update bits for WM8741_DACRMSB_ATTENUATION
After checking the code and datasheet, I think what we want in the second
snd_soc_update_bits call is to update WM8741_DACRMSB_ATTENUATION register
instead of WM8741_DACRLSB_ATTENUATION.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 14:15:40 +01:00
Axel Lin
f6a74ced46 ASoC: txx9: Add __exit_p at necessary place
We have __exit annotation for txx9aclc_generic_remove(),
thus add __devexit_p to wrap it.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 14:15:40 +01:00
Axel Lin
1c6927f872 ASoC: Staticise ep93xx_ac97_dai
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mika Westerberg <mika.westerberg@iki.fi>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 14:15:32 +01:00
Axel Lin
2e4891a5ec ASoC: Staticise simtec_audio_resume()
It is exported via resume callback of struct dev_pm_ops rather than referenced
directly and so should be staticised.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 14:15:23 +01:00
Axel Lin
43419b80fa ASoC: Remove needless codec->dapm.bias_level assignment to SND_SOC_BIAS_OFF
This assignment is done by the snd_soc_register_codec so there is no need
to redo it in probe function of a codec driver.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 14:15:22 +01:00
Arnd Bergmann
ca7aceef21 ASoC: sh: use correct __iomem annotations
This removes a few unnecessary type casts and avoids
sparse warnings.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 21:51:27 +01:00
Arnd Bergmann
b381bc8ab8 ASoC: imx: eukrea_tlv320 needs i2c
Add a missing dependency that is required for random configurations.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 21:51:19 +01:00
Arnd Bergmann
b5c49d49b9 ASoC: omap_mcpdm_remove cannot be __devexit
omap_mcpdm_remove is used from asoc_mcpdm_probe, which is an
initcall, and must not be discarded when HOTPLUG is disabled.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 20:19:59 +01:00
Axel Lin
4b8713fd54 ASoC: Remove unused srate variable in tegra_spdif_hw_params
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:45 +01:00
Axel Lin
c6add3f0e6 ASoC: Remove unused rate variable in magician_playback_hw_params
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:45 +01:00
Axel Lin
8fccf46905 ASoC: Staticise sh4_ssi_dai
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:45 +01:00
Axel Lin
bc6bd90eb7 ASoC: Staticise samsung_spdif_dai
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:45 +01:00
Axel Lin
019cd3b25a ASoC: tegra: Staticise tegra_i2s_dai and tegra_spdif_dai
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:44 +01:00
Axel Lin
c4c5839f98 ASoC: samsung: Add __devexit_p at necessary places
According to the comments in include/linux/init.h:

"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the confi
options."

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Cc: Jaswinder Singh <jassi.brar@samsung.com>
Cc: Ben Dooks <ben@simtec.co.uk>
Cc: Seungwhan Youn <sw.youn@samsung.com>
Cc: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:44 +01:00
Axel Lin
48860e409d ASoC: kirkwood-i2s: Add __devexit_p at necessary place
According to the comments in include/linux/init.h:

"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."

We have __devexit annotation for kirkwood_i2s_dev_remove(), thus add __devexit_p
at necessary place.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:44 +01:00
Axel Lin
3f0456bfd7 ASoC: wm8782: Add __devexit_p at necessary place
According to the comments in include/linux/init.h:

"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."

We have __devexit annotation for wm8782_remove(), thus add __devexit_p at
necessary place.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:58:44 +01:00
Mark Brown
ea945ab4d2 Merge branch 'for-3.1' into for-3.2 2011-10-02 19:57:19 +01:00
Jarkko Nikula
a3c6dac201 ASoC: omap: Use single hw_params callback in sdp3430 and zoom2
There is no need to use two hw_params callbacks in sdp3430 and zoom2 as
thet are now identical. Use instead the same snd_soc_ops structure and
hw_params callback for both DAI links.

Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:57:08 +01:00
Jarkko Nikula
cf9feff28f ASoC: omap: Convert bunch of machine drivers to use init time DAI format
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:57:07 +01:00
Jarkko Nikula
4dd0417253 ASoC: omap-mcbsp: Prepare for init time DAI format setting
Before commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the
dai_link") expectation for omap-mcbsp was that snd_soc_dai_set_fmt is to be
called first in machine hw_params callback before other CPU DAI functions.
Thus it was enough that only omap_mcbsp_dai_set_dai_fmt cleared the
mcbsp->regs structure.  [Note that this was pure convention, it's always
been OK to set things on init -- broonie]

Now this doesn't hold anymore since machine drivers can set the DAI format
only once on init time and thus mcbsp->regs may get out of sync when other
CPU DAI functions are modifying them dynamically with different values
between the calls. Therefore clear the accessed mcbsp->regs bits and
bitfields in other functions too.

Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-02 19:57:07 +01:00
Axel Lin
21d17dd2a3 ASoC: Fix setting update bits for WM8753_LADC and WM8753_RADC
Current code set update bits for WM8753_LDAC and WM8753_RDAC twice,
but missed setting update bits for WM8753_LADC and WM8753_RADC.

I think it is a copy-paste bug in commit 776065
"ASoC: codecs: wm8753: Fix register cache incoherency".

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-10-02 19:55:48 +01:00
Arnd Bergmann
eff919ac0f ASoC: use a valid device for dev_err() in Zylonite
A recent conversion has introduced references to &pdev->dev, which does
not actually exist in all the contexts it's used in.

Replace this with card->dev where necessary, in order to let
the driver build again.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-10-02 19:16:52 +01:00
Axel Lin
177fdd89f9 ASoC: tlv320aic3x: Use driver_data field of struct i2c_device_id to identify models
Save model information in driver_data so we can simplify the implementation.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 13:56:35 +01:00
Peter Ujfalusi
009d196b47 ASoC: twl6040: Simplify code in out_drv_event for pending work check
Instead of checking, if the work is pending, it is safer to cancel
the pending work, or wait till the scheduled work finishes.
This way we can avoid modifying the variables used by the work
function.
Since we know that no work is pending, we can remove the two additional
checks in POST_PMU, and PRE_PMD for non pending works.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 13:55:08 +01:00
Peter Ujfalusi
6fbb32d175 ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event
None of the driver handled by out_drv_event have it's power
bit shifted by 3.
Remove the case for shift 3, and also add comment for the cases.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 13:55:08 +01:00
Peter Ujfalusi
93eebc6982 ASoC: twl6040: correct loop counters for HS/HF ramp code
The Headset gain range is 0 - 0xf (4 bit resolution)
The Handsfree gain range is 0 - 0x1d (5 bit resolution,
0x1e, and 0x1f values are invalid)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 13:55:08 +01:00
Peter Ujfalusi
a46737aee5 ASoC: twl6040: One workqueue should be enough
It is a bit overkill to have three (3) separate
workqueue for a single driver.
We can manage things with one workqueue nicely.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 13:55:08 +01:00
Jarkko Nikula
91a18ae8ff ASoC: omap-mcbsp: Fix FS polarity for LEFT_J, DSP_A and DSP_B formats
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.

Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions.

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>
2011-09-30 12:56:17 +01:00
Axel Lin
f34dafb287 ASoC: sn95031: Do not use static variable for channel_index
No reason to use static variable for channel_index.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 12:56:16 +01:00
Axel Lin
5992c58781 ASoC: Add missed regulator_unregister_notifier and regulator_bulk_free in wm8995_remove
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 12:56:16 +01:00
Axel Lin
6423aa9154 ASoC: Remove unused "control_data" field of struct aic3x_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 12:56:15 +01:00
Ujfalusi, Peter
5382ffbb17 ASoC: sdp4430: Fix string for FM input name
The name contains invalid valid character (/), which
causes problems when trying to create the debugfs
directory structure:
ASoC: Failed to create Aux/FM Stereo In debugfs file

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 12:56:12 +01:00
Jarkko Nikula
ad51f76544 ASoC: Davinci: Fix FS polarity for I2S format
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.

Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions

Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Vaibhav Bedia <vaibhav.bedia@ti.com>
Cc: Sekhar Nori <nsekhar@ti.com>
Cc: Kevin Hilman <khilman@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 11:56:56 +01:00
Axel Lin
00e982a6a3 ASoC: Remove unused "control_data" field of struct cs4271_private
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 11:56:56 +01:00
Mark Brown
6d447be014 ASoC: Remove unused function check_vdac_to_outmix from rt5631
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 11:19:47 +01:00
Mark Brown
f79e5e8ce2 ASoC: Staticise non-exported symbols in rt5631
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-30 11:19:40 +01:00
Takashi Iwai
798cb7e897 ALSA: hda - Fix a regression of the position-buffer check
The commit a810364a04
    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>
2011-09-30 08:57:15 +02:00
Susan Gao
fbc7c62a3f ASoC: Fix a bug in WM8962 DSP_A and DSP_B settings
Signed-off-by: Susan Gao <sgao@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmico.com>
Cc: stable@kernel.org
2011-09-29 11:13:53 +01:00
Axel Lin
c29429f3b7 ASoC: tlv320dac33: Add guarding parentheses to macros
Put parentheses around macro argument uses. This avoids pitfalls
for the programmer, where the argument expansion does not give the
expected result, for example:

SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1);

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-29 10:53:12 +01:00
Dan Carpenter
bb690c9e27 sound: oss: use strlcpy() in sound_timer_init()
sound_timer.info.name is a 32 character buffer.  This function only
has one caller (in sound/oss/ad1848.c) and it passes as 128 character
buffer as "name".  I don't know if this is a problem in real life,
and I doubt we're going to add more OSS drivers so it's unlikely to
become an issue.  But we may as well take care of it.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-29 08:12:33 +02:00
Mark Brown
a8fdac83a3 ASoC: Also count neighbour checks for supplies
Missed when the stat was originally added.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 19:42:12 +01:00
Takashi Iwai
ef940b0403 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>
2011-09-28 20:19:16 +02:00
Takashi Iwai
27fe48d972 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>
2011-09-28 20:11:36 +02:00
Takashi Iwai
18a2b96233 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>
2011-09-28 17:15:14 +02:00
Takashi Iwai
a597310331 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>
2011-09-28 16:43:36 +02:00
Mark Brown
44fdd43387 ASoC: Use dai_fmt in speyside_wm8962
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:22:33 +01:00
Mark Brown
75d9ac46b9 ASoC: Allow DAI formats to be specified in the dai_link
For almost all machines the DAI format is a constant, always set to the
same thing. This means that not only should we normally set it on init
rather than in hw_params() (where it has been for historical reasons) we
should also allow users to configure this by setting a variable in the
dai_link structure. The combination of these two will make many machine
drivers even more data driven.

Implement a new dai_fmt field in the dai_link doing just that. Since 0 is
a valid value for many format flags and we need to be able to tell if the
field is actually set also add one to all the values used to configure
formats.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:22:28 +01:00
Axel Lin
21326db156 ASoC: adau1701: Fix prototype for adau1701_set_sysclk
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:21:14 +01:00
Axel Lin
16b7a9aa9a ASoC: Remove unused "control_data" field of struct ak4671_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:19:59 +01:00
Axel Lin
63de012f35 ASoC: Remove unused "control_data" field of struct max98095_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:19:59 +01:00
Axel Lin
72a921da07 ASoC: Remove unused "control_data" field of struct max98088_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:19:59 +01:00
Axel Lin
6d4f7097df ASoC: Remove unused "control_data" field of struct cs42l51_private
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:19:58 +01:00
Axel Lin
217069ea9a ASoC: Remove unused "control_data" field of struct cs4270_private
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:19:58 +01:00
Axel Lin
44cb209d33 ASoC: Remove unused "control_data" field of struct alc5623_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 13:19:58 +01:00
Axel Lin
23d622b14b ASoC: adau1701: Initialize codec->control_data before using it
Currently codec->control_data is not initialized before calling
process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE).

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 09:58:06 +01:00
Axel Lin
458f6f6921 ASoC: Fix setting adau1373_dai->master for SND_SOC_DAIFMT_CBS_CFS
In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 09:57:58 +01:00
Axel Lin
4addfd88ea ASoC: Remove unused "control_data" field of struct wm8904_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:53:38 +01:00
Axel Lin
b6ba8cc287 ASoC: Remove unused "control_data" field of struct wm9090_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:53:38 +01:00
Axel Lin
6e34216490 ASoC: Remove unused "control_data" field of struct wm9081_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:53:38 +01:00
Axel Lin
8c0c459ced ASoC: Remove unused "control_data" field of struct wm8978_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:53:38 +01:00
Axel Lin
ec61bde573 ASoC: Remove unused "control_data" field of struct wm8960_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:53:38 +01:00
Axel Lin
c9241ec6af ASoC: Remove unused "control_data" field of struct wm8940_priv
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:53:38 +01:00
Mark Brown
644f1ff4ff ASoC: Add device ID for WM9093 to WM9090 driver
The WM9093 is an enhanced version of the WM9093.  Add the device ID to
the driver, further patches will add support for the additional features
in the WM9093.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:21:44 +01:00
Karl Tsou
bcec267a17 ASoC: Add DRC control for WM8996
Signed-off-by: Karl Tsou <karl@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 19:02:45 +01:00
Takashi Iwai
0fac25908f Merge branch 'fix/asoc' into for-linus 2011-09-27 18:21:41 +02:00
Joe Perches
bfb9035c98 treewide: Correct spelling of successfully in comments
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2011-09-27 18:08:04 +02:00
Paul Bolle
395cf9691d doc: fix broken references
There are numerous broken references to Documentation files (in other
Documentation files, in comments, etc.). These broken references are
caused by typo's in the references, and by renames or removals of the
Documentation files. Some broken references are simply odd.

Fix these broken references, sometimes by dropping the irrelevant text
they were part of.

Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2011-09-27 18:08:04 +02:00
Takashi Iwai
218264ae9a 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>
2011-09-27 17:33:45 +02:00
Lars-Peter Clausen
0289053526 ASoC: ssm2602: Support setting the oscillator and the clock output state
Currently the oscillator is always enabled and the clock output is always
disabled. This patch adds support for controlling the oscillator and clock
output state through snd_soc_dai_set_sysclk. Which makes it possible to
disable or enable them dynamically according to the requirements of the board
on which the CODEC is used.

This patch also slightly modifies the behavior as to when the oscillator is
going to be disabled in low-power states. Previously it would only be disabled
in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no
components which depend on it should be active in this state.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 13:30:48 +01:00
Lars-Peter Clausen
a9d1974ea1 ASoC: ssm2602: Set initial bias level to standby
Set the initial bias level to standby during CODEC probe instead of leaving the
CODEC powered off.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 11:21:40 +01:00
Axel Lin
d1b73287c2 ASoC: Staticise sst_platform_dai
It is not used outside this driver so no need to make the symbol global.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 11:21:40 +01:00
Axel Lin
fe0cc75193 ASoC: Remove unused fields in struct mfld_mc_private
Both *socdev and *codec of struct mfld_mc_private are not being used
in this driver, remove it.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 11:21:40 +01:00
Mark Brown
60c9e3178f Merge branch 'for-3.1' into for-3.2 2011-09-27 11:21:11 +01:00
Lars-Peter Clausen
9058020cd9 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
2011-09-27 11:20:38 +01:00
Clemens Ladisch
17d900c4a1 ALSA: usb-audio: increase control transfer timeout
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers.  Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.

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

Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-27 09:21:48 +02:00
Peter Ujfalusi
4d64bdca44 ASoC: twl6040: No need to change delay during HF ramp
The Handsfree gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x3 raw, at 16 the gain
is -26dB.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:33:31 +01:00
Peter Ujfalusi
8ff1e17098 ASoC: twl6040: No need to change delay during HS ramp
The Headset gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x0 raw at
one end of the range, and not in the middle.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:33:31 +01:00
Peter Ujfalusi
46dd0b93a0 ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data
The delayed_work named 'delayed_work' is for the headset detection,
so move it to the twl6040_jack_data struct.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:24:18 +01:00
Peter Ujfalusi
e71a5e5af6 ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF
The delayed works for the output can be moved within the
twl6040_output struct (from the twl6040_data) to be better
organized.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:33 +01:00
Peter Ujfalusi
a8cc7189cd ASoC: twl6040: Combine the custom volsw get, and put functions
We can manage with one set of get, and put function for the gain
controls we need to handle with custom code due to the shadowing
of the register.
For both get, and put function we can call decide based on the
mc->rreg value, if we need to call the volsw, or the vlosw_2r
variant (in 2r case rreg is not 0).
Handling of the shadow values are the same for both type of
controls.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:32 +01:00
Peter Ujfalusi
eb6b71e7d9 ASoC: twl6040: Rename pga_event to out_drv_event
This event handler is used with the OUT_DRV widgets.
The name pga_event was misleading.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:32 +01:00
Peter Ujfalusi
7bf3d92cdd ASoC: sdp4430: Configure McPDM offset cancellation
Based on the values from twl6040 codec (HSOTRIM L/R) we can configure
the McPDM offset cancellation.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:32 +01:00
Peter Ujfalusi
89b0d550a6 ASoC: omap-mcpdm: API to configure offset cancellation
The offset cancellation values can be different from board to board, even
on the same HW platform.
Provide a way for the machine drivers to configure the McPDM offset
cancellation.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:32 +01:00
Peter Ujfalusi
db4aabcc1f ASoC: twl6040: Function to fetch the TRIM values
Provide API to fetch the TRIM values (for machine drivers)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:32 +01:00
Peter Ujfalusi
f97217f18e ASoC: twl6040: Read the TRIM values from the chip
Update the reg_cache with values from chip regarding to TRIM.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 22:23:32 +01:00
Thomas Pfaff
61a6a108d1 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>
2011-09-26 15:48:47 +02:00
Takashi Iwai
5ec02a1cfa Merge branch 'fix/hda' into topic/hda 2011-09-26 15:27:10 +02:00
Takashi Iwai
e0d32e335f 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>
2011-09-26 15:24:57 +02:00
Takashi Iwai
5fe6e0151d 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>
2011-09-26 11:13:43 +02:00
Axel Lin
c0fd9c9c42 ASoC: Drop exporting ad1980_dai
ad1980_dai is not used outside this driver,
thus drop exporting it.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 09:58:23 +01:00
Axel Lin
575e498ae1 ASoC: Drop exporting sn95031_get_mic_bias
sn95031_get_mic_bias() is not used outside this driver
and it is a static function now.
Thus drop exporting sn95031_get_mic_bias.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 09:58:23 +01:00
Peter Ujfalusi
141947e6ce ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’
Fixes the following compile time warning:
omap-mcbsp.c:519: warning: suggest explicit braces to avoid ambiguous ‘else’

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-26 09:57:35 +01:00
Takashi Iwai
6b69a0e520 ALSA: aloop - Use vmalloc buffer
snd-aloop driver is virtual and has no need for allocating contiguous
pages.  It'll be more system-friendly to use vmalloc buffers.

Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-24 12:16:29 +02:00
David Henningsson
6656b15d67 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>
2011-09-24 09:56:59 +02:00
Mark Brown
a850260e47 ASoC: Set idle_bias_off for WM1250 EV1
The WM1250 EV1 is functionally digital in a system (the analogue I/O
is either ground referenced or always powered) so flag it as idle_bias_off.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 17:05:44 +01:00
Mark Brown
213eb0fb1e ASoC: Add platform data for WM1250 EV1 GPIOs
The WM1250 EV1 has some GPIOs which can be used to control the behaviour
at runtime. Request them all if supplied and add a set_bias_level()
function to start and stop the clocks.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 17:05:38 +01:00
Mark Brown
85a843c50f ASoC: Don't force bias on ground referenced devices
Currently we force all devices in the system to be at the same bias level.
This is due to concerns about power or pop/click impacts from either
ramping VMID or mismatching VMID on the analogue I/O lines between
connected devices but does mean we power devices up more often than we
really need to.

If a device flags idle_bias_off this will usually mean that it's either
all digital or ground referenced (in which case the idle and powered bias
levels are identical) so this concern does not apply and we can save some
power by leaving it off when not needed itself.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 17:04:36 +01:00
Mark Brown
8c75615834 ASoC: Add DMIC control to Speyside WM8962 board
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 16:59:55 +01:00
Mark Brown
7564463c64 ASoC: Add support for on-board analogue microphones on Speyside WM8962
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 16:59:49 +01:00
Mark Brown
086d7f804e ASoC: Convert WM8962 MICBIAS to a supply widget
A supply widget is generally clearer than a MICBIAS widget and a mic bias
is just a type of supply so use a supply widget for the MICBIAS. This also
avoids confusion with the routing when connected to multiple inputs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 16:59:43 +01:00
Mark Brown
3f7d55a19a ASoC: Rename WM8962 DMIC widget to DMIC_ENA
Matches the register name and avoids confusion with board widgets.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 16:59:38 +01:00
Mark Brown
ef473fee67 ASoC: Support a wider range of sample rates on Speyside WM8962
As we've only got one audio interface and it is symmetric we can just set
SYSCLK based on the sample rate requested by the application layer. Provide
a default so bypass paths work before audio playback.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 16:59:37 +01:00
Takashi Iwai
3127b6aa75 Merge branch 'fix/asoc' into for-linus 2011-09-23 15:26:37 +02:00
Thomas Pfaff
362e4e49ab 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>
2011-09-23 15:26:06 +02:00
Raymond Yau
34588709af 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>
2011-09-23 15:21:29 +02:00
Andy Shevchenko
49957f3966 ALSA: 6fire: don't use custom hex_to_bin()
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 15:18:52 +02:00
Axel Lin
0a742681e6 ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 11:54:24 +01:00
Axel Lin
0010bcc226 ASoC: Remove unneeded mutex_init in wl1273_probe()
Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support",
snd_soc_register_codec() now does all the codec list and mutex init.
Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 11:54:24 +01:00
Axel Lin
a436089b77 ASoC: Staticize sn95031_dais
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 11:54:24 +01:00
Axel Lin
51e19fb385 ASoC: Staticize rt5631_dai
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 11:54:24 +01:00
Peter Ujfalusi
f34c660662 ASoC: twl6040: No need to read the INTID register
Since our irq handler has been called, it is granted, that
the reason was either PLUGINT, or UNPLUGINT.
The INTID register has been checked in the MFD part of
twl6040 driver (twl6040-irq.c).
We have no reason to read from chip again here.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 11:54:24 +01:00
Peter Ujfalusi
3b5b516fbf ASoC: omap-mcpdm: Correct the supported number of channels
OMAP4 McPDM supports 5 downlink (playback), and
3 uplink (capture) channels.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-23 11:52:32 +01:00
Mark Brown
ebca813cf0 Merge branch 'for-3.1' into for-3.2 2011-09-23 11:52:09 +01:00
Jarkko Nikula
34c869855a 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
2011-09-23 11:48:55 +01:00
Dan Carpenter
2ca595ab7a ALSA: hdspm - cleanup __user tags in ioctl()
This makes the code cleaner and silences a Sparse complaint:
sound/pci/rme9652/hdspm.c:6341:23: warning: incorrect type in assignment (incompatible argument 4 (different address spaces))
sound/pci/rme9652/hdspm.c:6341:23:    expected int ( *ioctl )( ... )
sound/pci/rme9652/hdspm.c:6341:23:    got int ( static [toplevel] *<noident> )( ... )
sound/pci/rme9652/hdspm.c:6102:44: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6225:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6264:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6283:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6289:59: warning: dereference of noderef expression

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

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

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

The hdspm_version struct had holes in it as well.

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

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

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 07:18:22 +02:00
Axel Lin
689b956e2c ASoC: Add Kconfig and Makefile entries for rt5631 codec
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 18:18:28 +01:00
Mark Brown
d73ec75cc4 ASoC: Add missed BCLK rate to WM5100 driver
Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:48:01 +01:00
Mark Brown
e56235e099 ASoC: Add another DAPM stat for neighbour checks
The number of times we look at a potentially connected neighbour is just
as important as the number of times we actually recurse into looking at
that neighbour so also collect that statistic.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:24:40 +01:00
Mark Brown
7aefb086c1 ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100
Allow the DBVDD2 and DBVDD3 rails to be powered down when idle, helping
fully power down connected devices when idle.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:24:32 +01:00
Peter Ujfalusi
ab6cf13943 ASoC/MFD: twl6040: Combine bit definitions for Headset control registers
Use one set of defines for the HS bits, since they are identical in both
control register.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:22 +01:00
Peter Ujfalusi
d13f1fe044 ASoC: twl6040/sdp4430: Change legacy DAI name
Change the legacy DAI name from "twl6040-hifi" to "twl6040-legacy" to
be more intuitive.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:22 +01:00
Peter Ujfalusi
fdb625ffd2 ASoC: twl6040: Support for AUX L/R output
AUX L/R outputs can be driver from the Handsfree PGA output.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:22 +01:00
Peter Ujfalusi
45b0f60de2 ASoC: twl6040: Use consistent names for Headset path
Use "Headset XYZ" for user visible controls, while the internal DAPM
widgets can use "HS XYZ".
In this way we can group the Headset related controls in UI
(alsamixer for example).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:22 +01:00
Peter Ujfalusi
df11ce295a ASoC: twl6040: Use consistent names for Handsfree path
Use "Handsfree XYZ" for user visible controls, while the internal DAPM
widgets can use "HF XYZ".
In this way we can group the Handsfree related controls in UI
(alsamixer for example).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:22 +01:00
Peter Ujfalusi
317596a694 ASoC: twl6040: Earphone path correction
Fix the DAPM routing for the earphone path.
Convert the DAPM_SWITCH_E to DAPM_OUT_DRV_E, so we can have correct
power up, and down sequence for EP.
Introduce mute control (Earphone Playback Switch) for users to
enable/disable the EP path.
Note: the EP does not have it's own dedicated DAC. EP is connected to
HSL DAC.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:22 +01:00
Peter Ujfalusi
d17bf31832 ASoC: twl6040: Introduce SW only shadow register
Software only shadow register to be used by the driver.
For example Earpiece path will need this shadow register.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:21 +01:00
Peter Ujfalusi
5bf692d972 ASoC: twl6040: Remove strings "NULL" from DAPM route
Replace the string with plain NULL.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:21 +01:00
Peter Ujfalusi
4548dc3c05 ASoC: twl6040: Fix comments for register names
Change the register name strings in the comments for the
twl6040_reg table, so it is easier to search for specific
register.

This is cosmetic change.

Before we had for example:
TWL6040_REG_HSLCTL as register definition.

At the register table we had:
TWL6040_HSLCTL

Searching for TWL6040_HSLCTL resulted no hits.

While if we look for REG_HSLCTL, we can find the places
the register has been used.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:21 +01:00
Peter Ujfalusi
3acef6854c ASoC: twl6040: Lower the power on gain values at startup
The default gains on outputs/inputs are set to 0dB.
This is fixing the pop noise issue at the first playback, which
caused by the wrong starting point of the ramp code.
The ramp code for the outputs expects the gains to be in
their lowest configuration in order to be effective.
After the playback stops, the ramp code takes care of
ramping down the gains to their minimum.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 17:20:21 +01:00
Takashi Iwai
272a487056 Merge branch 'fix/misc' into topic/misc 2011-09-22 16:41:52 +02:00
Ben Hutchings
c37279b92a ALSA: fm801: Gracefully handle failure of tuner auto-detect
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.

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

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

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

Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-22 15:51:46 +02:00
Mark Brown
fadd81b52c Merge branch 'peter/topic/for-mark/mcpdm_for-3.2' of git://gitorious.org/omap-audio/linux-audio into for-3.2 2011-09-22 11:25:58 +01:00
Mark Brown
8af0894546 ASoC: Include delay.h in 88pm860x
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 11:16:10 +01:00
Yong Zhang
88e24c3a4b sound: irq: Remove IRQF_DISABLED
Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).

So now this flag is a NOOP and can be removed.

Signed-off-by: Yong Zhang <yong.zhang0@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-22 11:19:20 +02:00
Takashi Iwai
af1910a817 Merge branch 'topic/asoc' into topic/remove-irqf_disable 2011-09-22 09:56:12 +02:00
Misael Lopez Cruz
f5f9d7bf6e ASoC: omap-mcpdm: Replace legacy driver
Reasons for the replacement:
The current driver for McPDM was developed to support the legacy mode only.
In preparation for the ABE support the current driver stack need the be
replaced.
The new driver is much simpler, easier to extend, and it also fixes some of the
issues with the old stack.

Main changes:
- single file for omap-mcpdm (mcpdm.c/h removed)
- Define names for registers, bits cleaned up, prefixed
- Full-duplex audio operation (arecord | aplay) has been fixed
- Less code

McPDM need to be turned off after all streams has been stopped.
This might cause pop noise on the output, if the codec's DAC is
still powered at this time.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-22 09:22:50 +03:00
Peter Ujfalusi
3a98cd6b2b ASoC: OMAP4: McPDM: Convert to hwmod/omap_device
In order to probe, and operate correctly, the OMAP McPDM driver needs to
be converted to use hwmod.
The device name has been changed to probe the driver.
Replace the clk_* with pm_runtime_* calls to manage the clocks correctly.
Missing request_mem_region/release_mem_region added.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-22 09:19:15 +03:00
Peter Ujfalusi
b199adfdff ASoC: omap-mcpdm: Fix threshold and dma configuration
DMA packet_size must be configured based on the McPDM FIFO threshold
value, number of channels.
Due to the FIFO operation the DMA muse be configured differently for
playback, and capture.
At the same time fix the McPDM threshold values used for playback, and
capture to avoid broken code.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-22 09:19:15 +03:00
Peter Ujfalusi
0722d055ac ASoC: tpa6130a2: Remove model_id from platform data
The model_id is no longer needed within the platform_data
for the TPA driver since the model of TPA specified
with the device name (tpa6130a2/tpa6140a2).

Also update rx51 (the only affected user) to use the device name rather
than platform data.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 16:07:45 +01:00
Peter Ujfalusi
07441006b2 ASoC: tpa6130a2: Model support cleanup
Use the device name and driver_data to identify
the TPA model supported by the driver.
Board files should use either "tpa6130a2" or
"tpa6140a2" as device name to specify the model
in used on the specific board.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 16:07:40 +01:00
Dong Aisheng
17841020e9 ASoC: soc-core: symmetry checking for each DAIs separately
The orginal code does not cover the case that one DAI such as codec
may be shared between other two DAIs(CPU).
When do symmetry checking, altough the codec DAI requires symmetry,
the two CPU DAIs may still be configured to run on different rates.

We change to check each DAI's state separately instead of only checking
the dai link to prevent this issue.

Signed-off-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 15:59:46 +01:00
Bas Vermeulen
548aae8cc4 ASoC: 88pm860x-codec - reset the codec correctly
Reset the codec according to the Audio power-up delay errata for the 88PM8607.

Signed-off-by: Bas Vermeulen <bas.vermeulen@novero.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 15:52:28 +01:00
Bas Vermeulen
06c15baf90 ASoC: 88pm860x-codec - Allow independent use of both I2S playback and capture
Introduce a I2S CLK supply so playback and capture can operate independently.

Signed-off-by: Bas Vermeulen <bas.vermeulen@novero.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 15:52:28 +01:00
johnnyhsu@realtek.com
d80852223e ASoC: Add driver for rt5631
Signed-off-by: Johnny Hsu <johnnyhsu@realtek.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 15:52:23 +01:00
Axel Lin
49acf73bd0 ASoC: Staticise nuc900_dma_getposition()
It is not used outside this driver so no need to make the symbol global.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 14:55:50 +01:00
Axel Lin
bb75463684 ASoC: Staticise jz4740_pcm_new()
It is not used outside this driver so no need to make the symbol global.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 14:55:50 +01:00
Axel Lin
3b70e3b55d ASoC: Staticise bf5xx_pcm_i2s_new()
It is not used outside this driver so no need to make the symbol global.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 14:55:50 +01:00
Axel Lin
3156cf73bd ASoC: Staticise bf5xx_pcm_ac97_new()
It is not used outside this driver so no need to make the symbol global.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 14:55:50 +01:00
Mark Brown
92868de624 Merge branch 'for-3.1' into for-3.2 2011-09-21 14:54:34 +01:00
Mark Brown
f0e8ed858e 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>
2011-09-21 14:54:23 +01:00
Mark Brown
7c81beb048 ASoC: Factor out per-widget DAPM power checks
The indentation is getting a little deep. Should be straight code motion,
no functional changes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 14:53:45 +01:00
Mark Brown
de02d0786d ASoC: Trace and collect statistics for DAPM graph walking
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.

In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-21 14:53:44 +01:00
David Henningsson
0b6c49b59f 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>
2011-09-21 10:31:00 +02:00
David Henningsson
42cf0d0155 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>
2011-09-20 18:22:33 +02:00
Mark Brown
6d4baf084f ASoC: Add WM5100 driver
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation.  This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-20 16:02:16 +01:00
Mark Brown
f648de832d Merge branch 'for-3.1' into for-3.2 2011-09-20 12:59:35 +01:00
Lars-Peter Clausen
26806a4266 ASoC: ssm2602: Do not dereference codec->control_data
The driver assumes that control_data points to the drivers i2c_client struct,
but this is no longer the case since the ASoC core has switched to regmap.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-20 12:59:07 +01:00
Axel Lin
d890a1a42d ASoC: fsl: Fix error handling if platform_device_add fails
Call platform_device_put() instead of platform_device_unregister() if
platform_device_add() fails.

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>
2011-09-20 12:58:59 +01:00
Mark Brown
f93dc4b6c9 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>
2011-09-20 12:57:48 +01:00
Mark Brown
ded71dcb77 ASoC: Refcount WM8996 bandgap from FLL too
For digital only paths we need to make sure the bandgap is enabled prior
to starting the FLL which isn't tied into DAPM.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-20 12:56:53 +01:00
Mark Brown
c9d023adb6 ASoC: Fix unused variable warning in WM8996
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-20 12:56:36 +01:00
Takashi Iwai
290b421f69 Merge branch 'fix/hda' into topic/hda 2011-09-20 09:14:04 +02:00
David Henningsson
46724c2e02 ALSA: HDA: Add support for IDT 92HD93
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 09:13:18 +02:00
Clemens Ladisch
5495ffbd7b ALSA: via82xx: allow to disable the SRC
Add the PCM rule to allow disabling the PCM playback SRC.

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

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

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

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

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:56:34 +02:00
Peter Ujfalusi
cdd5054c3e ASoC: twl6040: Correct supported number of playback channels
twl6040 supports 5 playback, and 2 capture channels

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-19 23:15:59 +01:00
Peter Ujfalusi
d8dd032d53 ASoC: twl6040: Fix the number of channels for vibra
Only mono audio can be used for vibra (DL4 channel).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-19 23:15:59 +01:00
Peter Ujfalusi
2c27ff41d8 ASoC: twl6040: Use chip defaults in the initial reg_cache
Reset the twl6040_reg array to hold the chip default values.
The only changed values were for the microphone input selection.
Select no input for the microphones in the twl6040_init_chip function.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-19 23:15:58 +01:00
Peter Ujfalusi
a52762eee9 ASoC: twl6040: Chip initialization cleanup
There is no need to write to the vio registers at probe time, since most
them either read only, or shared with MFD or not used.
On the other hand it is a good idea to updated the ASICREV register in
the cache at this time.

After power up we need to restore some registers. Clean up the list to
contain only the registers we are going to restore.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-19 23:15:57 +01:00
Dong Aisheng
76067540c6 ASoC: mxs-saif: add record function
1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.

The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.

2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.

Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-19 18:31:59 +01:00
Axel Lin
5d42940c25 ASoC: sn95031: Staticize sn95031_pcm_hw_params
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-19 18:31:00 +01:00
Mark Brown
45cf367e80 ASoC: Add line loads to the list of supported detections for Speyside
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-19 18:30:59 +01:00
Mark Brown
0b684cc14a ASoC: Initial WM8996 headphone impedance measurement support
The WM8996 can measure the impedance of accessories connected to the
headphone output. Implement initial support for this, measuring the
left channel impedance when an accessory is detected and using this
to distinguish between a line load and a headphone load.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-19 18:30:59 +01:00
Mark Brown
8259df12fd ASoC: WM8996 only needs bandgap for analogue functionality
Rather than managing the bandgap in the bias level control use a supply
widget as we only actually need to enable it for analogue paths, not
fully digital ones.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-19 18:30:58 +01:00
Mark Brown
53daf20893 ASoC: Display the error code when we fail to add a DAPM control
Useful for diagnostics.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-19 11:27:22 +01:00
Takashi Iwai
8974bd51a7 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration
When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work.  It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.

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

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

Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Cc: stable@kernel.org (3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-19 11:31:34 +02:00
Timur Tabi
0016226d03 ASoC: support all possible sample rates in the WM8776 driver
The WM8776 supports a continuous range of sample rates rather than
discrete values and supports a wider range of sample rates on the
playback path than is currently supported.  Update the constraints on
the DAIs to reflect this.

Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 17:35:18 +01:00
Axel Lin
0547d0f3da ASoC: wm8995: Remove unused i2c variable in wm8995_remove()
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>
2011-09-16 10:49:21 +01:00
Axel Lin
6fa0c25bf4 ASoC: wm8995: Return -EINVAL if device ID mismatch
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>
2011-09-16 10:49:09 +01:00
Axel Lin
275708f88d ASoC: tpa6130a2: 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>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:13:53 +01:00
Ben Gardiner
be4ff96122 ASoC: davinci-pcm: trivial: replace link with actual chan/link
The ambiguously named variable 'link' is used as a temporary throughout
davinci-pcm -- its presence makes grepping (and groking) the code
difficult.

Replace link with the value of link in almost all sites. The exception
is a couple places where the last-assigned link/chan needs to be
returned by a function -- in these cases, rename to last_link.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:12:47 +01:00
Mika Westerberg
8a386ca26d ASoC: edb93xx: convert to use snd_soc_register_card()
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.

Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:11:32 +01:00
Mika Westerberg
5a0a03c5ef ASoC: simone: convert to use snd_soc_register_card()
Current method for machine driver to register with the ASoC core is to
use snd_soc_register_card() instead of creating a "soc-audio" platform device.

In addition we use platform_device_register_simple() to create a platform
device for the codec. This function will handle putting and deleting the
device automatically which simplifies the error handling in the machine
driver.

Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:11:26 +01:00
Mika Westerberg
9306816954 ASoC: ep93xx-pcm: add MODULE_ALIAS
To get the PCM module loaded automatically by udev et al. we need to add a
proper MODULE_ALIAS.

Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:11:22 +01:00
Mika Westerberg
62e4f7d138 ASoC: snappercl15: convert to use snd_soc_register_card()
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.

Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:10:49 +01:00
Timur Tabi
d1dc698a54 ASoC: support sample sizes properly in the WM8776 codec driver
Use snd_pcm_format_width() to determine the sample size, instead of
checking specify sample formats and assuming that those are the only
valid format.

This change adds support for big-endian architectures (which use the _BE
formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE).

[Fixed single letter variable name legibility problem -- broonie]

Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 10:06:31 +01:00
Daniele Guerrieri
14515a0829 ALSA: usb-audio: Added support for Roland UM-ONE midi-usb interface
Roland UM-ONE midi usb interface differs from Roland UM-1.

Signed-off-by: Daniele Guerrieri <d.guerrieri@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-16 08:31:45 +02:00
Takashi Iwai
0308110615 Merge branch 'fix/misc' into topic/misc 2011-09-16 08:29:04 +02:00
Mark Brown
cc2115cbfc Merge branch 'for-3.1' into for-3.2 2011-09-16 00:54:25 +01:00
Mark Brown
f998f257c9 ASoC: Fix WM8996 DC servo operation without IRQ
We need to count the timeout down.

Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-16 00:54:08 +01:00
Axel Lin
4f6c7e1593 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>
2011-09-16 00:50:16 +01:00
Axel Lin
7803e329bb ASoC: samsung: Fix checking return value of clk_get
clk_get() returns a pointer to the struct clk or an ERR_PTR().
This patch also use PTR_ERR() for return value.

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>
2011-09-16 00:05:57 +01:00
Timur Tabi
5e538ecade ASoC: improve asynchronous mode support in the fsl_ssi driver
The Freescale SSI audio controller supports "synchronous" and "asynchronous"
modes.  In synchronous mode, playback and capture use the same input clock,
so sample rates must be the same during simultaneous playback and capture.
Unfortunately, the code which supports asynchronous mode is just broken in
various ways.  In particular, it was constraining sample sizes as well as
the sample rate.

The fix also allows us to simplify the code by eliminating the 'asynchronous',
'playback', and 'capture' variables that were used to keep track of playback
and capture streams.

Unfortunately, it turns out that simulataneous playback and record does not
actually work on the only platform that supports asynchronous mode: the
Freescale P1022DS reference board.  If a second stream is started, the SSI
grinds to halt for both streams.  This is true even if the P1022 is configured
for synchronous mode, so it's likely a hardware problem that needs to be
worked around.

Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-16 00:05:29 +01:00
Arjan van de Ven
763437a9e7 ALSA: pcm - fix race condition in wait_for_avail()
wait_for_avail() in pcm_lib.c has a race in it (observed in practice by an
Intel validation group).

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

However there are two races in the existing code

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

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

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

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

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

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

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

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

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:01 +02:00
David Henningsson
2e1210bc3d ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 13:45:12 +02:00
Mark Brown
32d2a0c17d ASoC: Correct channel numbers for WM8996 AIF2
The AIF1 channels are numbered from zero than one; do the same thing for
AIF2 too.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-14 11:00:07 +01:00
Mark Brown
c83495af63 ASoC: Disable WM8996 CPVDD supply when not in use
The WM8996 only requires CPVDD when the charge pump is active so control
it separately to the other supplies, only enabling it when the charge pump
is active. This will result in a small power saving on systems which are
able to provide independent software control of the supply.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-14 10:59:36 +01:00
Clemens Ladisch
dba8b46992 ALSA: mpu401: clean up interrupt specification
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive:  To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero.  At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller.  This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.

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

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

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

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

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

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

Also remove a unneeded NULL checking for kfree.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:35:47 +02:00
Raymond Yau
356aab7d41 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>
2011-09-12 10:34:48 +02:00
Raymond Yau
89f3325a6e ALSA: ymfpci: add "Playback" to FM Legacy Volume control
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth.

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

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

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

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:30:20 +02:00
Lars-Peter Clausen
30ab1e7886 ASoC: ad193x: Setup regmap read and write flag masks for SPI
Currently register read-back for the ad193x is broken, because it expects bit 0
of the upper byte to be set to indicate a read operation, while the regmap
default for SPI is to use bit 7.

This patch also addresses another oddity of the device. There are SPI and I2C
versions of this codec. In both cases the registers are 8-bit wide and numbered
from 0x0 to 0x10, but in the SPI case there is also a so called
'global address' which is prefixed in-front of the register address. The global
address mimics I2C behaviour and includes a static device address the and the
read/write flag. This basically extends the register address to an 16-bit value
numbered from 0x800 to 0x810. These are the register numbers which are
currently used by the driver. This works, because I2C will ignore the upper
8 bits of the register, but it is still a bit confusing, as there are no such
register numbers in the I2C case.

The approach taken by this patch is to number the registers from 0x00 to 0x10
and encode the global address for SPI mode into the read and write flag masks.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-09 11:13:31 -07:00
Axel Lin
694741471b ASoC: playpaq_wm8510: Return proper error if clk_get fails
Return proper error instead of 0 if clk_get fails.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-09 09:42:00 -07:00
Lu Guanqun
283e42e011 ASoC: sst_platform: fix memory leak
snd_pcm_hw_constraint_integer() could return -1, in this case, sst platform is
not opened successfully.  However the corresponding close callback isn't able
to be called later on to release these two allocated memories, thus resulting
in memory leak.

This patch moves the check for hardware contraints earlier, thus resolving this
issue.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-08 15:43:03 -07:00
Lu Guanqun
22be504aaa ASoC: sst_platform: using builtin function
Use the builtin snd_soc_set_runtime_hwparams() instead of assigning it by
myself.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-08 15:43:03 -07:00
Lu Guanqun
c2f6fce33e ASoC: sst_platform: trivial coding style fix
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-08 15:43:02 -07:00
Axel Lin
0f73644f37 ASoC: ad1980: Return proper error if vendor id mismatch
Return -ENODEV instead of 0 if vendor id mismatch.

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>
2011-09-08 15:39:51 -07:00
Mark Brown
3ed464659a ASoC: Remove unused step size from debugfs CODEC write function
We don't use the step size so there's no need to work it out.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-08 15:38:18 -07:00
Axel Lin
c8f4b7fd68 ASoC: alc5623: Remove unused mutex
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>
2011-09-05 18:15:15 -07:00
Mark Brown
27b6d92a24 ASoC: Check that WM8996 FLL started even if we don't have the IRQ
We can directly read the FLL lock status on WM8996 so even if we don't
have an interrupt wired up we can still verify that the FLL started
successfully.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-09-05 18:14:04 -07:00
Mark Brown
11323e3283 Merge branch 'for-3.1' into for-3.2 2011-09-05 18:13:31 -07:00
Axel Lin
4ed0d012c9 ASoC: Add missing platform_device_put in raumfeld_audio_init error path
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-05 18:13:04 -07:00
Lars-Peter Clausen
c5d2e650bd ASoC: Blackfin: bf5xx-ad193x: Fix codec device name
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.

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

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-09-05 18:10:52 -07:00
Axel Lin
7b4615ba81 ASoC: sn95031: Fix the logic to find free channel
In the case of no free channel available,
current implementation returns 0 instead of negative errno.

This patch fixes the logic to return -EINVAL if no free channel available.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-04 08:53:39 -07:00
Mark Brown
efd614ac06 Merge branch 'for-3.1' into for-3.2 2011-08-31 09:57:44 +01:00
Mark Brown
da1c6ea6cf ASoC: Allow source specification for CODEC level sysclk
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.

Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:57:35 +01:00
Mark Brown
d2dd0540c1 ASoC: Add device tree binding for WM8804
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:52:57 +01:00
Mark Brown
b6de431556 ASoC: Add device tree binding for WM8776
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:52:50 +01:00
Mark Brown
13c7d08f54 ASoC: Add device tree binding for WM8770
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-31 09:52:43 +01:00