Commit Graph

12849 Commits

Author SHA1 Message Date
Takashi Iwai 23d30f2827 ALSA: hda - Move up the fixup helper functions to the library module
Move the fixup helper functions in patch_realtek.c to hda_auto_parser.c
so that they can be used in other codec drivers like patch_conexant.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 18:01:33 +02:00
Takashi Iwai 5536c6d693 ALSA: hda - Protect the power-saving count with spinlock
To avoid some races.  Still not perfect, but now a bit safer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 18:01:30 +02:00
Takashi Iwai 339876d70a ALSA: hda - Clear the power-saving states properly at reset
Some power-saving states have been left unchanged in
snd_hda_codec_reset(), and this is a potential danger because the
function may be called in various situations including the continuous
operation after that call.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 18:01:01 +02:00
Takashi Iwai 7f30830b7b ALSA: hda - Always resume the codec immediately
This is a fix for the problem in commit 785f857d1c, the pop noise
issue on some machines with ALC269.  The problem was the uninitialized
state after the resume due to the delayed resume of the codec chips.
In that commit, we tried to fix by forcibly putting the codec to D3 at
suspend.  But, this still also leaves the uninitialized state after
resume, and it _might_ be still problematic with some BIOS.  Since the
commit turned out to regress another issues, we reverted it in the
end.

Now, in this fix, try to fix by turning on the codec immediately at
the resume path.  We need to take care of the power-saving in this
case.  When the device is woken up at the power-saved state, it should
go power-saving again after the resume.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 18:00:47 +02:00
Takashi Iwai 2abb80176c sound: allow the unit search until 256 in sound_core.c
The upper limit of the available minors isn't necessarily 128 + unit,
but it's rather up to 256.  Fixing this allows more than 8 devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 17:27:03 +02:00
Takashi Iwai 779ae5a083 ALSA: Fix the card number limit of OSS-emulation
There are left-over codes from the ancient days with the static device
number limitation of 8.  Actaully OSS can support up to 16 cards.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 17:25:56 +02:00
Takashi Iwai c382a9f009 ALSA: hda - Fix possible access to uninitialized work struct
The work struct must be initialized before the possible call in the
destructor.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 16:39:57 +02:00
Takashi Iwai 3de9517356 ALSA: hda/realtek - Call a common helper for alc_spec initialization
Just a clean up by calling the same helper function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 16:38:14 +02:00
Takashi Iwai ffd344444f Merge branch 'fix/hda' into topic/hda 2012-05-08 16:38:02 +02:00
Takashi Iwai 619a341b78 Revert "ALSA: hda - Set codec to D3 forcibly even if not used"
This reverts commit 785f857d1c.

The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.

Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence.  But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.

In short, the safest option now is to revert this affecting commit.

Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.

Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 16:35:42 +02:00
Takashi Iwai af741c150f ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup
The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization.  Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 14:10:31 +02:00
Mark Brown 0fb7d0c30b ASoC: wm9081: Hook DAC up via DAPM rather than stream
More current API usage.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-08 12:29:18 +01:00
Mark Brown 55b2784730 ASoC: lowland: Support digital link for WM9081
The WM9081 on Lowland is connected to AIF3 on the WM5100.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-08 12:29:17 +01:00
Mark Brown 277b6fdac1 ASoC: lowland: Convert to dai_fmt
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-08 12:29:16 +01:00
Mark Brown b3bba9a1a8 ASoC: pcm: Fix DPCM for aux_devs
When we instantiate an aux_dev we use a fake rtd as part of the process
which doesn't have a dai_link associated with it. Fix the dpcm startup
code to cope with this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-05-08 12:29:15 +01:00
Andre Schramm 42eb92380f ALSA: hdsp - Provide ioctl_compat
snd_hdsp uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels.

Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Reviewed-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-08 07:27:22 +02:00
Peter Ujfalusi 3bb8a819c6 ASoC: twl6040: Remove HS/HF gain ramp feature
None of the machines uses the gain ramp possibility for HS/HF.
This code path is mostly unused and it does not reduces the pop
noise on the output (it alters it to sound a bit different).
The preferred method to reduce pop noise is to use ABE.
Remove the gain ramp, and related features form the driver.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-07 18:27:36 +01:00
Mark Brown a2e888f0d7 ALSA: jack: Update documention to reflect other userspace interfaces
Since this is a generic API which should support any userspace interface
for reporting jacks update the documentation a little to make that a bit
clearer.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-07 18:11:37 +02:00
guoyh d93ca1ae61 ASoC: pxa: allocate the SSP DMA parameters in startup
Allocating the SSP DMA parameters in startup, freeing it in
shutdown instead of freeing and re-allocating it in hw_params.
After doing that, the logic is clear and more safe.

Signed-off-by: guoyh <guoyh@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-07 12:55:35 +01:00
Takashi Iwai bca4013855 ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo
Reported-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-07 11:14:53 +02:00
Takashi Iwai f5c53d898c ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G
Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-07 10:07:33 +02:00
Mark Hills c914f55f7c ALSA: echoaudio: Remove incorrect part of assertion
This assertion seems to imply that chip->dsp_code_to_load is a pointer.
It's actually an integer handle on the actual firmware, and 0 has no
special meaning.

The assertion prevents initialisation of a Darla20 card, but would also
affect other models. It seems it was introduced in commit dd7b254d.

ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting...
ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000
ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware...
ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20
------------[ cut here ]------------
WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]()
Hardware name: Dell DM051
BUG? (!chip->dsp_code_to_load || !chip->comm_page)

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-06 12:54:20 +02:00
Linus Torvalds 1c2f954806 sound fixes for 3.4-rc6
As good as nothing exciting here; just a few trivial fixes for
 various ASoC stuff.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound sound fixes from Takashi Iwai:
 "As good as nothing exciting here; just a few trivial fixes for various
  ASoC stuff."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: omap-pcm: Free dma buffers in case of error.
  ASoC: s3c2412-i2s: Fix dai registration
  ASoC: wm8350: Don't use locally allocated codec struct
  ASoC: tlv312aic23: unbreak resume
  ASoC: bf5xx-ssm2602: Set DAI format
  ASoC: core: check of_property_count_strings failure
  ASoC: dt: sgtl5000.txt: Add description for 'reg' field
  ASoC: wm_hubs: Make sure we don't disable differential line outputs
2012-05-05 10:07:06 -07:00
Clemens Ladisch 76bc7a0d0a ALSA: oxygen: add Xonar DGX support
Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-05 14:24:12 +02:00
Takashi Iwai e9e7183fd2 Merge branch 'fix/asoc' into for-linus 2012-05-05 11:27:26 +02:00
Takashi Iwai b339583c57 Merge branch 'for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc into fix/asoc 2012-05-05 11:26:50 +02:00
Takashi Iwai 20c76945d0 ASoC: Updates for 3.4
Nothing terribly exciting here, a bunch of small and simple fixes
 scattered around the place.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for 3.4

Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
2012-05-05 11:25:17 +02:00
Oleg Matcovschi fad9365bcc ASoC: omap-pcm: Free dma buffers in case of error.
Signed-off-by: Oleg Matcovschi <oleg.matcovschi@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2012-05-04 12:09:28 +01:00
Ashish Chavan 3cb81651d0 ASoC: da7210: Minor improvements and a bugfix
This patch improves playback quality for few sample rates like 8000 and
11025 Hz.

This also fixes an issue observed during testing of pll slave mode. Due
to the issue, on some rare occasions there was no sound output for first
time playback after system boot, though all subsequent playbacks were
fine. It was mainly because of the sequence in which SRM bit was
enabled.

Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-03 18:53:52 +01:00
Mark Brown 9b5231247c ASoC: wm5100: Set the DAI base address in the DAI drivers
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-05-02 15:44:11 +01:00
Mark Brown 94aa733a47 ASoC: wm_hubs: Cache multiple DCS offsets
Rather than invalidating the cached DCS value every time the headphone
gain changes store multiple values, indexed by gain. This allows the
optimisation we get from the cache to take effect more often.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-01 19:21:07 +01:00
Stephen Warren 6264f668d5 ASoC: tegra: add device tree support for TrimSlice
This binding doesn't include the nvidia,model or nvidia,audio-routing
properties the other Tegra audio DT bindings have, because this binding
is targetted at a single machine, rather than for any machine using the
tlv320aic23 codec.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:47:54 +01:00
Heiko Stübner 06412088ce ASoC: s3c2412-i2s: Fix dai registration
As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised
s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai.

Without this call the snd_soc_dai_ops structure isn't initialised correctly.

Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:45:25 +01:00
Mark Brown 3a96c77ef7 ASoC: wm8350: Replace use of custom I/O with snd_soc_read()/write()
Makes the code more standard and prepares for better framework usage.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:36:48 +01:00
Mark Brown 3e4ba82cac ASoC: wm8350: Remove check for clocks in trigger()
This is now very standard behaviour for CODECs so shouldn't be device
specific and we shouldn't really be trying to peer into the register
cache from atomic context anyway.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:36:47 +01:00
Mark Brown b9c374b26c ASoC: cs42l52: Remove duplicate module exit code
In the conversion to module_init_i2c() the original open coded module
exit function was left.  Remove it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:36:47 +01:00
Brian Austin dfe0f98b8d ASoC: Add support for CS42L52 Codec
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:36:20 +01:00
Mark Brown 30facd4d51 ASoC: wm8350: Don't use locally allocated codec struct
The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 23:34:42 +01:00
Liam Girdwood cd0f8911c5 ASoC: core: Fix dai_link dereference.
We should check dailess before dereferencing.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-30 11:09:13 +01:00
Eric Bénard e875c1e3e7 ASoC: tlv312aic23: unbreak resume
* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value

* this patch solves the problem by only working on the 9 bits the
register contains.

Signed-off-by: Eric Bénard <eric@eukrea.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-04-30 10:06:44 +01:00
Richard Zhao 81e8e49261 ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000
It tries to clk_get the clock. And if it failed, it assumes the clock
by default enabled.

Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:44:08 +01:00
Richard Zhao 717071dc27 ASoC: imx-sgtl5000: add of_node_put when probe fail.
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:44:06 +01:00
Mark Brown 04de57c153 ASoC: wm_hubs: Enable class W for output mixer paths
Class W can be used for any path where only data from the DAC is routed
to the headphones. Currently we only enable it when the direct DAC to
headphone path is used but it can also be enabled for paths that go via
the output mixer providing the DAC is the only input to the output mixer.
Implement support for this, including updates to the class W status when
the output mixer configuration is changed. This also allows us to enable
the DC servo optimisations for DAC to headphone paths where the output
mixer is used.

In general the direct DAC path is still preferred as this will offer
better performance on most wm_hubs devices but these additional paths
can simplify use case management.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:42:12 +01:00
Mark Brown c340304dd8 ASoC: wm_hubs: Factor out class W management
Since the analogue portions of the checks for class W are the same over
all the devices factor out these checks into wm_hubs and while we're at
it also use wm_hubs_dac_hp_direct() to enable class W optimisations on
more paths.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:42:11 +01:00
Mark Brown af31a227e1 ASoC: wm_hubs: Special case headphones for digital paths in more use cases
The optimisations which we can do with caching the headphone DCS result in
wm_hubs have only been enabled in cases where class W is enabled. However,
there are more use cases which can benefit from the cache, especially with
WM8994 series devices with their more advanced digital routing.

Rather than keying off the class W information from the CODECs have a
check in wm_hubs for a suitable path and use that to determine if we can
deploy our headphone optimisations.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:42:10 +01:00
Liam Girdwood f57b8488bc ASoC: dpcm: Fixup debugFS for DPCM state.
Remove writable debugFS permission, use simple_open() and
fix indentation.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:38:47 +01:00
Ashish Chavan 604bb229b5 ASoC: da7210: Minor bugfix for non pll slave mode
This patch fixes a bug discovered during testing of non pll slave mode.
Due to the bug chip was not getting correctly configured and as a result
there was no sound output while playback. After applying this patch,
both pll and non pll modes work fine.

Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-27 18:38:47 +01:00
Mark Brown 9747cec21e ASoC: dapm: Move CODEC<->CODEC params off stack
Reduce our stack consumption by moving the params off the stack, they
are reasonably large and might be an issue on platforms with small stacks.

Reported-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ackeded-by: Liam Girdwood <lrg@ti.com>
2012-04-27 18:38:32 +01:00
Linus Torvalds 2390c0fca6 sound fixes for 3.4-rc5
A workaround for an ASUS laptop and a few ASoC changes;
 most of the commits are tagged for stable, too.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A workaround for an ASUS laptop and a few ASoC changes; most of the
  commits are tagged for stable, too."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: wm8994: Improve sequencing of AIF channel enables
  ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
  ASoC: fsi: update for dmaengine prep_slave_sg fallout.
  ASoC: core: Fix card RTD count for deferred probe.
  ASoC: cs42l73: don't use negative array index
  ASoC: dapm: Ensure power gets managed for line widgets
2012-04-26 15:32:39 -07:00
Mark Brown 3a334adab0 ASoC: wm8994: Add trace showing wm8958_micd_set_rate()
This can be helpful to users when tuning their systems.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 18:08:56 +01:00
Mark Brown fcdc4de7ad ASoC: wm8994: Allow rate configuration with custom mic callback
If a driver using a custom mic detection callback has provided a table
of mic detection rates via platform data then use it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 18:08:45 +01:00
Mark Brown e9d9a968e7 ASoC: wm8994: Tune debounce rates for jack detect mode
Use a slightly larger debounce when identifying accessory type and a
slightly smaller one when detecting buttons in response to user feedback
from large scale testing.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 18:08:39 +01:00
Mark Brown 501bf0354d ASoC: wm8996: Put the microphone biases into bypass mode when idle
When we're not actively doing audio we don't need the microphone biases
to be regulated, noise is not important when we are not looking at the
audio signal. Save some power by putting the MICBIAS regulators into
bypass mode when not doing audio.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 18:06:56 +01:00
Liam Girdwood be3f3f2ce6 ASoC: pcm: Add pcm operation for pcm ioctl.
Provide an ioctl marshaller for ASoC platform drivers.
This will use the default ALSA handler if no platform
handler exists.

This is also required for DPCM BE PCMs as snd_pcm_info()
will call the ioctl as part of stream startup.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:43 +01:00
Liam Girdwood 07bf84aaf7 ASoC: dpcm: Add bespoke trigger()
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.

A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:42 +01:00
Liam Girdwood 47c88ffff7 ASoC: dpcm: Add API for DAI link substream and runtime lookup
Some component drivers will need to be able to look up their
DAI link substream and RTD data. Provide a mechanism for this.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:19 +01:00
Liam Girdwood 618dae11f8 ASoC: dpcm: Add runtime dynamic route update
This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.

This patchs adds/changes the following :-

 o Adds DPCM runtime update core.
 o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
   to return if a change has occured rather than 0. No other users check
   atm.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:19 +01:00
Liam Girdwood f86dcef87b ASoC: dpcm: Add debugFS support for DPCM
Add debugFS files for DPCM link management information.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:19 +01:00
Liam Girdwood 01d7584cd2 ASoC: dpcm: Add Dynamic PCM core operations.
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.

Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.

e.g. pcm:0,0 routing digital data to 2 external codecs.

FE pcm:0,0  ----> BE (McBSP.0) ----> CODEC 0
             +--> BE (McPDM.0) ----> CODEC 1

e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.

FE pcm:0,0 ---
             +--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---

The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.

DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.

Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.

This patch adds the core DPCM code and contains :-

 o The FE and BE PCM operations.
 o FE and BE DAI link support.
 o FE and BE PCM creation.
 o BE support API.
 o BE and FE link management.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 17:48:19 +01:00
Fabio Estevam f20c2cb999 ASoC: core: Remove unused variable 'min'
commit 4183eed2 (ASoC: core: Add signed multi register control) introduced
the variable 'min',but it is not used.

Remove it to fix the following build warning:

sound/soc/soc-core.c: In function 'snd_soc_put_xr_sx':
sound/soc/soc-core.c:2990: warning: unused variable 'min'

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-26 10:29:13 +01:00
Takashi Iwai 1a442cc3df ALSA: asihpi - Revert module_pci_driver conversion for asihpi.c
It contains non-standard call.

Reported-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-26 07:19:39 +02:00
Lars-Peter Clausen bec3d9a973 ASoC: SSM2602: Convert to direct regmap API usage
Mostly a one to one converion. On one occasion the patch replaces a
snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps
to keep the conversion simple.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 11:28:10 +01:00
Lars-Peter Clausen d86a11d68c ASoC: SSM2602: Remove driver specific version
We have never really updated that version number and probably never will, so
just remove it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 11:27:57 +01:00
Lars-Peter Clausen 8b3f39dab5 ASoC: SSM2602: Add sysclk based rate constraints
Not all advertised rates are available for all sysclk frequencies. Add
additional sysclk based rate constraints.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 11:27:53 +01:00
Lars-Peter Clausen d9ca8e76f3 ASoC: bf5xx-ssm2602: Setup sysclock in init callback
The sysclock is fixed, so just set it up once in the init callback instead of
setting it repeatably in the hw_params callback.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 11:19:31 +01:00
Lars-Peter Clausen a3a53fe154 ASoC: bf5xx-ssm2602: Set DAI format
Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC
machine drivers to use the dai_links dai_fmt field to setup their DAI format.
For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt,
but missed to set the dai_links dai_fmt field.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 11:14:44 +01:00
Kyung-Kwee Ryu e05854ddaa ASoC: wm8994: Make sure we disable FLL bypass when stopping the FLL
If FLL bypass is left enabled when we disable the CODEC then the output
clock will be left running which consumes a small amount of additional
current. Only enable bypass when there is an output.

Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 09:50:50 +01:00
Daniel Mack 07a5e9d4fd ALSA: snd-usb: fix some typos in endpoint.c documentation
Also be more specific about some details while at it.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 20:16:18 +02:00
Richard Zhao c34ce320d9 ASoC: core: check of_property_count_strings failure
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-04-24 12:06:27 +01:00
Takashi Iwai e9f66d9b9c ALSA: pci: clean up using module_pci_driver()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 12:25:00 +02:00
Andrew Morton 68853fa30c ALSA: usb-audio: sound/usb/endpoint.c: suppress warning
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function

Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:10:10 +02:00
Takashi Iwai baba2e0d2b ALSA: usb-audio: Add missing error checks in snd_ebox44_create_mixer()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:07:38 +02:00
Felix Homann d34bf14851 ALSA: usb-audio: M-Audio Fast Track Ultra: Add effect controls
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.

[fixed missing break by tiwai]

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:06:06 +02:00
Felix Homann cfe8f97c82 ALSA: usb-audio: Rename Fast Track Ultra mixer quirk functions
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:02:11 +02:00
Felix Homann 25ee7ef8fa ALSA: usb-audio: Add TLV to M-Audio Fast Track Ultra controls
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:01:46 +02:00
Felix Homann 285de9c08b ALSA: usb-audio: Rename and export mixer_vol_tlv
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:01:27 +02:00
Felix Homann 8a4d1d397b ALSA: usb-audio: Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks.
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().

[fixed minor checkpatch.pl warnings by tiwai]

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:00:45 +02:00
Mark Brown de050acaa1 ASoC: wm_hubs: Make sure we don't disable differential line outputs
While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-23 20:20:00 +01:00
Kristoffer KARLSSON dd7b10b30c ASoC: core: Add strobe control
Added support for a control that strobes a bit in
a register to high then back to low (or the inverse).

This is typically useful for hardware that requires
strobing a singe bit to trigger some functionality
and where exposing the bit in a normal single control
would require the user to first manually set then
again unset the bit again for the strobe to trigger.

Added convenience macro.

SOC_SINGLE_STROBE

Added accessor implementations.

snd_soc_get_strobe
snd_soc_put_strobe

Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-23 20:05:06 +01:00
Kristoffer KARLSSON 4183eed288 ASoC: core: Add signed multi register control
Added control type that can span multiple consecutive codec registers
forming a single signed value in a MSB/LSB manner.
The control dynamically adjusts to the register word size configured
in driver.

Added convenience macro.

SOC_SINGLE_XR_SX

Added accessor implementations.

snd_soc_info_xr_sx
snd_soc_get_xr_sx
snd_soc_put_xr_sx

Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-23 20:05:06 +01:00
Jesper Juhl c1a4ecd921 ASoC: wm8994: Delete trailing whitespace from sound/soc/codecs/wm8994.c
While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-23 19:02:20 +01:00
Mark Brown fbe5c580a6 ASoC: Update regmap access for WM5100 DSP control registers
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-23 18:52:31 +01:00
Takashi Iwai cff7873554 ASoC: updates for 3.4
Slightly larger than normal - the DAPM fix is a "this should always have
 worked" type of thing which is very clear and should have no impact on
 systems that don't need it.  The WM8994 fix is driver specific but
 pretty important for that driver.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: updates for 3.4

Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it.  The WM8994 fix is driver specific but
pretty important for that driver.
2012-04-23 18:39:47 +02:00
Mark Brown 1a38336b86 ASoC: wm8994: Improve sequencing of AIF channel enables
This ensures a clean startup of the channels, without this change some
use cases could result in issues in a small proportion of cases.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-04-23 12:55:52 +01:00
Linus Torvalds 9f24ff6f42 First MFD pull request for 3.4 fixes
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Merge tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6

Pull MFD fixes from Samuel Ortiz:
 "We have 3 build fixes, a OMAP USB host PHY reset fix and the twl6040
  conversion to an i2c driver.  The latter may not sound like a fix but
  the twl6040 MFD driver won't probe without it, triggering an OMAP4
  audio regression."

* tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6:
  mfd: Fix modular builds of rc5t583 regulator support
  mfd: Fix asic3_gpio_to_irq
  ARM: OMAP3: USB: Fix the EHCI ULPI PHY reset issue
  mfd: Convert twl6040 to i2c driver, and separate it from twl core
  mfd : Fix dbx500 compilation error
2012-04-21 12:42:12 -07:00
Daniel Mack c89a5d9cac ALSA: snd-usb: remove refactorization left-overs
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-21 17:40:28 +02:00
Linus Torvalds a54769c505 sound fixes for 3.4-rc4
Fixes for a few regressions of HD-audio, originated partly from 3.4
 and partly 3.3.
 The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
 are based on 3.3 then merged back to 3.4, so that they can be merged
 to stable tree cleanly.  The non-trivial merge conflicts are because
 of this action.
 
 In addition, a copule of trivial fixes for documentation and a long-
 statnding issue in the listing of built-in sound driver at boot time.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Fixes for a few regressions of HD-audio, originated partly from 3.4
  and partly 3.3.

  The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
  are based on 3.3 then merged back to 3.4, so that they can be merged
  to stable tree cleanly.  The non-trivial merge conflicts are because
  of this action.

  In addition, a couple of trivial fixes for documentation and a long-
  standing issue in the listing of built-in sound driver at boot time."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/conexant - Set up the missing docking-station pins
  ALSA: hda/conexant - Don't set HP pin-control bit unconditionally
  ALSA: workaround: change the timing of alsa_sound_last_init()
  ALSA: hda/sigmatel - Fix inverted mute LED
  ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1
  ALSA: fix core/vmaster.c kernel-doc warning
2012-04-20 10:41:00 -07:00
Takashi Iwai 6942c103fb ALSA: hda - Skip pin capability sanity check for bogus values
Some old codecs like ALC880 seem to give a bogus pin capability value 0
occasionally.  This breaks the new sanity check in snd_hda_set_pin_ctl().
Skip the sanity checks in such a case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-20 13:08:40 +02:00
Takashi Iwai 4740860b53 ALSA: hda - Add snd_hda_get_default_vref() helper function
Add a new helper function to guess the default VREF pin control bits
for mic in.  This can be used to set the pin control value safely
matching with the actual pin capabilities.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-20 13:06:53 +02:00
Takashi Iwai cdd03cedc5 ALSA: hda - Introduce snd_hda_set_pin_ctl*() helper functions
For setting the pin-control values more safely to match with the
actual pin capability bits, a copule of new helper functions,
snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are
introduced.  These are simple replacement of the codec verb write with
AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out
superfluous pin-control bits if they don't fit with the corresponding
pin capabilities.

Some codecs are screwed up or ignore the command when such a wrong bit
is set.  These helpers will avoid such secret errors.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-20 12:38:48 +02:00
David Henningsson 5ac57550f2 ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/950490
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-20 10:08:08 +02:00
Takashi Iwai d398011057 Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_conexant.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-19 17:20:13 +02:00
Takashi Iwai c817eebec5 Merge branch 'fix/cxt-stable' into fix/hda
Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4.  These were committed in that branch to make the stable
merging easier.

Conflicts:
	sound/pci/hda/patch_conexant.c
2012-04-19 17:13:03 +02:00
Takashi Iwai d70f363222 ALSA: hda/conexant - Set up the missing docking-station pins
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.

The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.

Bugzilla entries:
	https://bugzilla.redhat.com/show_bug.cgi?id=808559
	https://bugzilla.redhat.com/show_bug.cgi?id=806217
	https://bugzilla.redhat.com/show_bug.cgi?id=810697

Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-19 17:10:34 +02:00
Takashi Iwai ca3649de02 ALSA: hda/conexant - Don't set HP pin-control bit unconditionally
Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.

Check the pin-capability and avoid the HP bit if not supported.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-19 15:15:25 +02:00
Mark Brown fde39a6b15 ASoC: wm1250-ev1: Support sample rate configuration
The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-19 14:10:21 +01:00
Mark Brown 5f6ac59f70 ASoC: wm1250-ev1: Support stereo
Springbank can support stereo, though it is primarily intended for mono
use cases.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-19 14:10:19 +01:00
Kuninori Morimoto 590b4775d6 ALSA: workaround: change the timing of alsa_sound_last_init()
Current alsa_sound_last_init() was called as __initcall().
So, on current ALSA, only devices that had been properly
registered at this point were shown.
So, it will show "No soundcards found" if driver requests
probe deferment. it's often misleading.
This patch delays the timing of alsa_sound_last_init()
as workaround.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviwed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-19 13:51:54 +02:00
Takashi Iwai 3e843196c6 ALSA: hda/sigmatel - Fix inverted mute LED
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way.  The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.

Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-19 12:04:03 +02:00
Takashi Iwai 118cb4a408 ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1
Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken.  There are two problems behind it:

- This machine doesn't like the default COEF setup for ALC260 we take
  now as default

- BIOS doesn't set the pins correctly at all; especially the machine
  uses only the pin 0x0f for both headphone and speaker

This patch adds the fixup as a workaround for these issues.

Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-19 07:33:27 +02:00
Liam Girdwood ec2e3031b6 ASoC: dapm: Add API call to query valid DAPM paths
In preparation for ASoC DSP support.

Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.

This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-18 18:23:00 +01:00
Mark Brown 0cbe4b36b0 ASoC: samsung: Hook up AIF2 to the CODEC on Littlemill
Connect the WM1250-EV1 baseband simulator on Littlemill systems up to
the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider
range of use cases to be represented.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-18 18:20:58 +01:00
Masanari Iida 59bf896406 Fix "the the" in various Kconfig
Fix typo "the the" in various Kconfig.

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2012-04-18 14:12:27 +02:00
Paul Mundt cdf27f3737 ASoC: fsi: update for dmaengine prep_slave_sg fallout.
Leading up to the ->device_prep_slave_sg change in
185ecb5f4f 'dmaengine: add context
parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was
added in place to guard against the API change, though the fsi driver
wasn't updated in the process (presumably its dmaengine support hadn't
been merged yet at the time). This trivially switches over to the new
wrapper and gets it building again.

Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-18 09:16:13 +01:00
Takashi Iwai 56599bb020 Merge branch 'topic/usb-endpoint' into topic/misc 2012-04-18 07:57:32 +02:00
Randy Dunlap f2ec52d4c3 ALSA: fix core/vmaster.c kernel-doc warning
Fix kernel-doc warning in sound/core/vmaster.c:

Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data'

Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-18 07:56:15 +02:00
Clemens Ladisch 7bdbff6762 firewire: move rcode_string() to core
There is nothing audio-specific about the rcode_string() helper, so move
it from snd-firewire-lib into firewire-core to allow other code to use it.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de> (fixed sound/firewire/cmp.c)
2012-04-17 22:54:55 +02:00
Mark Brown 8c5b842b83 ASoC: wm8994: Keep AIF3 tristated when not in use
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-17 20:53:56 +01:00
Ashish Chavan c4b14e70a1 ASoC: da7210: Minor update for PLL and SRM
This patch converts multiple if conditions in to single if with "&&"s.

Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-17 20:52:42 +01:00
Liam Girdwood a7dbb60342 ASoC: core: Fix card RTD count for deferred probe.
Currently we increment the number of RTD's per card during the DAI link
bind. This can cause an incorrect RTD count when we cannot find a component
and defer the probe (and hence perform the DAI link bind for the card again).

Fix the count so that it is cleared before every card registration
and bind attempt.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-17 20:52:19 +01:00
Ashish Chavan 570aa7bae5 ASoC: da7210: Add support for PLL and SRM
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.

This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,

(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled

This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.

Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.

Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-17 14:43:48 +01:00
Mark Brown 26e6781155 ASoC: Use dai_fmt in Speyside
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-16 20:00:00 +01:00
Mark Brown d5efccd5b6 Linux 3.4-rc3
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ASoC: Merge tag 'v3.4-rc3' into for-3.5

Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.

Conflicts:
	sound/soc/soc-core.c
	sound/soc/tegra/tegra_i2s.c
	sound/soc/tegra/tegra_spdif.c
2012-04-16 19:40:27 +01:00
Fabio Estevam 516541a00c ASoC: soc-dapm: Use '%llx' with 'u64' type.
Fix the following build warning:

sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'

'%llx' should be used with 'u64' type.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-16 19:36:46 +01:00
Mark Brown c74184ed30 ASoC: core: Support transparent CODEC<->CODEC DAI links
Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct.  If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.

This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.

This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there.  It is expected that the bias
level callbacks will be used for clock configuration.

Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute().  This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here.  We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.

At present we are also restricted to a single DAPM link for the entire
DAI.  Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-04-16 19:36:29 +01:00
Mark Brown 054880febe ASoC: core: Bind DAIs to CODECs at registration time
We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.

This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-16 19:36:29 +01:00
Mark Brown f04209a7b0 ASoC: core: Flip master for CODECs in the CPU slot of a CODEC<->CODEC link
When two CODEC DAIs are linked directly to each other then if we give the
same master mode settings to both devices things won't work as either
neither will drive or they'll drive against each other. Flip the settings
for the DAI in the CPU slot of the DAI link.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-16 19:36:29 +01:00
Mark Brown 1eee1b3833 ASoC: dapm: Allow DAI widgets to be routed through
In order to allow CODEC<->CODEC links to function we will need to allow
DAPM paths to be created that pass through DAIs rather than only ones
that are source or sunk at the DAI.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-04-16 19:36:29 +01:00
Mark Brown 04570c628f ASoC: core: Return -ENOTSUPP instead of -EINVAL if mute is not supported
This helps us ignore errors in callers if the operation failed due to not
being available as opposed to an error.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-04-16 19:36:29 +01:00
Peter Ujfalusi 8eaeb93933 mfd: Convert twl6040 to i2c driver, and separate it from twl core
Complete the separation of the twl6040 from the twl core since
it is a separate chip, not part of the twl6030 PMIC.

Make the needed Kconfig changes for the depending drivers at the
same time to avoid breaking the kernel build (vibra, ASoC components).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2012-04-16 16:45:34 +02:00
Linus Torvalds 218a8c2b57 another sound fixes for 3.4-rc3
A few regression fixes for Realtek HD-audio codecs, mainly specific to
 some laptop models.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull another round of sound fixes from Takashi Iwai:
 "A few regression fixes for Realtek HD-audio codecs, mainly specific to
  some laptop models."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
  ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines
  ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G
  ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co
  ALSA: hda/realtek - Add a few ALC882 model strings back
2012-04-15 11:14:07 -07:00
Mark Hills 7536c301f8 ALSA: snd-usb-audio: Replace mixer for Electrix Ebox-44
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-15 15:40:08 +02:00
Mark Hills 284a8dd6f0 ALSA: snd-usb-audio: Skip un-parseable mixer units instead of erroring
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.

This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-15 15:39:55 +02:00
Dan Carpenter 60884c2767 ASoC: dapm: release lock on error paths
We added locking here but there were a couple error paths where we
forgot to drop the lock before returning.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-15 10:46:17 +01:00
Stephen Warren 7203a62562 ASoC: convert Tegra20 DAS driver to regmap
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 19:16:53 +01:00
Stephen Warren 5939ae7475 ASoC: convert Tegra20 SPDIF driver to regmap
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 19:16:53 +01:00
Stephen Warren c1607416aa ASoC: convert Tegra20 I2S driver to regmap
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 19:16:53 +01:00
Stephen Warren d19e779b84 ASoC: tegra: select REGMAP_MMIO
All Tegra ASoC drivers will be reworked to use MMIO regmaps. Select
this in Kconfig.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 18:30:24 +01:00
Takashi Iwai 22026c1a7b ALSA: usb: Remove obsoleted fields from struct snd_usb_substream
Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 12:57:39 +02:00
Takashi Iwai 85f71932e5 ALSA: usb: Fix fill_max flag set
ep->fill_max is a 1 bit flag, thus it has to be boolean.
  sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
  sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 12:41:54 +02:00
Kuninori Morimoto cd04461e2f ASoC: sh: fsi: select simple-card on Kconfig
Current SuperH FSI require simple-card driver as sound card.
This patch select it on Kconfig when FSI was selected.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:28 +01:00
Kuninori Morimoto 064bfada66 ASoC: sh: fsi: use simple-card instead of fsi-da7210
This patch uses simple-card driver instead of fsi-da7210 on each board.
To select DA7210 driver, each boards select it on Kconfig.

This patch removes fsi-da7210 driver which is no longer needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:27 +01:00
Kuninori Morimoto fa063b4804 ASoC: sh: fsi: use simple-card instead of fsi-hdmi
This patch uses simple-card driver instead of fsi-hdmi on each board.
This patch removes fsi-hdmi driver which is no longer needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:26 +01:00
Kuninori Morimoto af8a2fe12f ASoC: sh: fsi: use simple-card instead of fsi-ak4642
This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.

This patch removes fsi-ak4642 driver which is no longer needed

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:26 +01:00
Kuninori Morimoto f2390880ec ASoC: add generic simple-card support
Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:25 +01:00
Stephen Warren cdc04fd1e9 ASoC: tegra: add Kconfig and Makefile support for Tegra30
This adds Kconfig options for the Tegra30 AHUB and I2S controller, and
updates the Tegra+WM8903 machine driver Kconfig to select those.

Includes a squashed bugfix from Sumit Bhattacharya <sumitb@nvidia.com>

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:23 +01:00
Stephen Warren 4fb0384f3d ASoC: tegra: add tegra30-i2s driver
This provides an ASoC DAI interface for Tegra 30's I2S controller.

Includes a squashed bugfix from Sumit Bhattacharya <sumitb@nvidia.com>

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:23 +01:00
Stephen Warren be944d42cc ASoC: tegra: add tegra30-ahub driver
The AHUB (Audio Hub) is a mux/crossbar which links all audio-related
devices except the HDA controller on Tegra30. The devices include the
DMA FIFOs, DAM (Digital Audio Mixers), I2S controllers, and SPDIF
controller. Audio data may be routed between these devices in various
combinations as required by board design/application.

Includes a squashed bugfix from Nikesh Oswal <noswal@nvidia.com>
Includes squashed bugfixes from Sumit Bhattacharya <sumitb@nvidia.com>

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 11:29:22 +01:00
Jesper Juhl 86fc499823 ASoC: cs42l73: don't use negative array index
If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in
sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using
the (negative) return value as array index on the very next line of
code - that's bad.

Catch the negative return value and propagate it to the caller (which
checks for it) and things are a bit more sane :-)

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-13 10:01:38 +01:00
Takashi Iwai c5ee4ec828 ALSA: usb: Remove unused variable
sound/usb/endpoint.c: In function ‘deactivate_urbs’:
sound/usb/endpoint.c:520:16: warning: unused variable ‘flags’ [-Wunused-variable]

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:27:28 +02:00
Daniel Mack 94c27215bc ALSA: snd-usb: add some documentation
Document the new streaming code and some of the functions so that
contributers can catch up easier.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:25:24 +02:00
Daniel Mack c75a8a7ae5 ALSA: snd-usb: add support for implicit feedback
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:32 +02:00
Daniel Mack d399ff9593 ALSA: snd-usb: remove old streaming logic
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:23 +02:00
Daniel Mack edcd3633e7 ALSA: snd-usb: switch over to new endpoint streaming logic
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:08 +02:00
Daniel Mack 8fdff6a319 ALSA: snd-usb: implement new endpoint streaming model
This patch adds a new generic streaming logic for audio over USB.

It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.

A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.

With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.

In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:23:42 +02:00
Daniel Mack 596580d0ee ALSA: snd-usb: add snd_usb_audio-wide mutex
This is needed for new card-wide list operations.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:21:55 +02:00
Jesper Juhl 7d7eb9ea31 ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.

While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 07:35:57 +02:00
Mark Brown 7e1f7c8a6e ASoC: dapm: Ensure power gets managed for line widgets
Line widgets had not been included in either the power up or power down
sequences so if a widget had an event associated with it that event would
never be run. Fix this minimally by adding them to the sequences, we
should probably be doing away with the specific widget types as they all
have the same priority anyway.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-04-12 19:36:52 +01:00
Josh Boyer 29ebe40284 ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines
A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine.  Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option.  All Mac machines are now explicitly
handled with a quirk and the auto-parser.  This adds a quirk for the device
found on the Mac Pro 5,1 machines.

This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559

[sorted the new entry in the ID number order by tiwai]

Reported-by: Gabriel Somlo <somlo@cmu.edu>
Signed-off-by: Josh Boyer <jwboyer@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-12 20:00:48 +02:00
Takashi Iwai fe97da1f70 ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G
It's compatible with 8930G.
Using the same fixup gives the proper 5.1 sound back.

Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-12 19:57:18 +02:00
Takashi Iwai 038d4fef37 ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co
Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-12 07:58:19 +02:00
Linus Torvalds a1ada08606 sound fixes for 3.4-rc3
- A series of fixes for Conexant 20549 HD-audio codec chip
 - A workaround for HDMI hotplug debug prints that annoyed people
 - A fix for the new support of platform DAPM contexts
 - Many driver-specific minor fixes
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 - A series of fixes for Conexant 20549 HD-audio codec chip
 - A workaround for HDMI hotplug debug prints that annoyed people
 - A fix for the new support of platform DAPM contexts
 - Many driver-specific minor fixes

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
  ALSA: sound/isa/sscape.c: add missing resource-release code
  sound: sound/oss/msnd_pinnacle.c: add vfrees
  ALSA: hda - clean up CX20549 test mixer setup
  ALSA: hda - CX20549 doesn't need pin_amp_workaround.
  ALSA: hda - Remove CD control from model=benq for CX20549
  ALSA: hda - fix record volume controls of CX20459 ("Venice")
  ALSA: hda - Rename capture sources of CX20549 to match common conventions
  ALSA: hda - Fix proc output for ADC amp values of CX20549
  ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
  ASoC: set idle_bias_off=1 for all platform DAPM contexts
  ASoC: imx-audmux: Check for NULL pointer
  ASoC: imx-audmux: Fix ssi port numbers in sysfs
  ASoC: ak4642: fixup: mute needs +1 step
  MAINTAINERS: Don't list everyone working on Wolfson drivers
  MAINTAINERS: Add missing ASoC OMAP co-maintainer
  ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro
  ASoC: tegra: ensure clocks are enabled when touching registers
  ASoC: sgtl5000: Enable VAG when DAC/ADC up
  ALSA: asihpi - fix return value of hpios_locked_mem_alloc()
2012-04-11 11:07:38 -07:00
Linus Torvalds 39f86a608a Merge branch 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media fixes from Mauro Carvalho Chehab:

 - dvb core: there is a regression found when used with xine.  For
   whatever unknown reason, xine (and xine-lib clients) wants that the
   frontend to tell what frequency he is using even before the PLL lock
   (or at least, it expects a non-zero frequency).

   On DVB, the frequency is only actually known after a frequency
   zig-zag seek, done by the DVB core.  Anyway, the fix was trivial.
   That solves Fedora BZ#808871.

 - ivtv: fix a regression when selecting the language channel

 - uvc: fix a race-related crash

 - it913x: fixes firmware loading

 - two trivial patches (a dependency issue at a radio driver at sound
   Kconfig, and a warning fix on dvb).

* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
  [media] uvcvideo: Fix race-related crash in uvc_video_clock_update()
  [media] Drivers/media/radio: Fix build error
  [media] dvb_frontend: fix compiler warning
  [media] it913x: fix firmware loading errors
  [media] ivtv: Fix AUDIO_(BILINGUAL_)CHANNEL_SELECT regression
  [media] dvb_frontend: regression fix: userspace ABI broken for xine
2012-04-11 11:05:34 -07:00
Takashi Iwai 912093bc7c ALSA: hda/realtek - Add a few ALC882 model strings back
Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-11 14:10:57 +02:00
Fabio Estevam 019ec5059b ASoC: wm9705: Fix build due to removal of 'runtime' definition
sound/soc/codecs/wm9705.c: In function 'ac97_prepare':
sound/soc/codecs/wm9705.c:251: error: 'runtime' undeclared (first use in this function)

This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-11 11:43:19 +01:00
Fabio Estevam 7a0a289c5f ASoC: ac97: Fix build due to removal of 'runtime' definition
Fix the following build error:

sound/soc/codecs/ac97.c: In function 'ac97_prepare':
sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function)

This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-11 11:43:19 +01:00
Fabio Estevam b46ac308bf ASoC: wm9712: Fix build due to missing definition of "runtime"
Fix the following build error:

sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function)
sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function)

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-10 22:35:18 +01:00
Stephen Warren 8127bf5529 ASoC: tegra: utils: Don't use of_have_populated_dt()
Recent list discussions concluded that drivers should not be calling
of_have_populated_dt(), and hence of_have_populated_dt() should not be
exported. Use a different mechanism to detect DT vs. non-DT boot.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-10 22:33:19 +01:00
David Henningsson 83b0c6ba99 ALSA: hda - Fix oops caused by recent commit "Fix internal mic for Lenovo Ideapad U300s"
Make sure we don't dereference the "quirk" pointer when it is null.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-10 14:55:01 +02:00
Fengguang Wu fae3d88a5c ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
Also remove two warnings when CONFIG_SND_DEBUG is not set:

sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-10 14:53:55 +02:00
Mark Brown 20dc24a951 ASoC: wm8994: Implement FLL bypass support
Later WM8994 class devices can bypass the FLL from BCLK. Do this
automatically when the FLL input and output frequencies match up.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-10 11:11:22 +01:00
Jesper Juhl 34ff0f95b1 ASoC: wm8994: Don't test for NULL before release_firmware()
release_firmware() does its own NULL ptr testing, it's redundant to
also test before calling it.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-10 11:07:58 +01:00
Mark Brown b920eb41a8 Linux 3.4-rc2
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Merge tag 'v3.4-rc2' into regulator-drivers

Linux 3.4-rc2 contains some fixes that further patches depend upon.
2012-04-10 09:52:59 +01:00
Mark Brown bbdd391556 ASoC: wm1250-ev1: Convert to module_i2c_driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-10 09:31:56 +01:00
Jesper Juhl 507230c999 ALSA: riptide: remove redundant NULL test before release_firmware()
release_firmware() deals gracefully with NULL pointers, no need to check first.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-10 08:42:33 +02:00
Julia Lawall 38be95dd3d ALSA: sound/isa/sscape.c: add missing resource-release code
At the point of this error-handling code, both regions and the dma have
been allocated, so free it as done in previous and subsequent
error-handling code.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-10 08:42:07 +02:00
Julia Lawall 156d14da4c sound: sound/oss/msnd_pinnacle.c: add vfrees
At the point of this error-handling code, HAVE_DSPCODEH may be undefined,
so free INITCODE and PERMCODE as done elsewhere.  A jump and label are
introduced to avoid code duplication.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-10 08:41:54 +02:00
Stephen Warren 82ef0ae46b ASoC: tegra: add runtime PM support
To the Tegra I2S and SPDIF drivers

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-09 17:42:48 +01:00
Hans Verkuil d3a92d6248 [media] Drivers/media/radio: Fix build error
On Sunday, April 01, 2012 21:09:34 Tracey Dent wrote:
> radio-maxiradio depends on SND_FM801_TEA575X_BOOL to build or will
> result in an build error such as:
>
> Kernel: arch/x86/boot/bzImage is ready  (#1)
> ERROR: "snd_tea575x_init" [drivers/media/radio/radio-maxiradio.ko] undefined!
> ERROR: "snd_tea575x_exit" [drivers/media/radio/radio-maxiradio.ko] undefined!
> WARNING: modpost: Found 6 section mismatch(es).
> To see full details build your kernel with:
> 'make CONFIG_DEBUG_SECTION_MISMATCH=y'
> make[1]: *** [__modpost] Error 1
> make: *** [modules] Error 2
>
> Select CONFIG_SND_TEA575X to fixes problem and enable
> the driver to be built as desired.
>
> v2:
> instead of selecting CONFIG_SND_FM801_TEA575X_BOOL, select
> CONFIG_SND_TEA575X, which in turns selects CONFIG_SND_FM801_TEA575X_BOOL
> and any other dependencies for it to build.

No, this is the correct patch:

RADIO_MAXIRADIO should be treated just like RADIO_SF16FMR2, I just didn't
realize at the time that it had to be added as a SND_TEA575X dependency.

Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Tested-by: Shea Levy <shea@shealevy.com>
Acked-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
2012-04-09 10:12:54 -03:00
Mark Brown c172708d38 regulator: core: Use a struct to pass in regulator runtime configuration
Rather than adding new arguments to regulator_register() every time we
want to add a new bit of dynamic information at runtime change the function
to take these via a struct. By doing this we avoid needing to do further
changes like the recent addition of device tree support which required each
regulator driver to be updated to take an additional parameter.

The regulator_desc which should (mostly) be static data is still passed
separately as most drivers are able to configure this statically at build
time.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-09 12:37:09 +01:00
Mark Brown 1eecb8280b Linux 3.4-rc2
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Merge tag 'v3.4-rc2' into for-3.5

Linux 3.4-rc2 contains some bug fixes we need, including the addition of
an export for regcache_sync_region().
2012-04-09 11:53:45 +01:00
Jiri Kosina e75d660672 Merge branch 'master' into for-next
Merge with latest Linus' tree, as I have incoming patches
that fix code that is newer than current HEAD of for-next.

Conflicts:
	drivers/net/ethernet/realtek/r8169.c
2012-04-08 21:48:52 +02:00
Takashi Iwai c38f62b08d ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
 for platform DAPM contexts - we were picking the wrong default for the
 idle_bias_off setting which was meaning we weren't actually achieving
 any useful runtime PM on platform devices.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: fixes for 3.4

A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
2012-04-07 12:28:00 +02:00
Takashi Iwai 5ff5c3a4ab Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_conexant.c
2012-04-07 12:26:55 +02:00
Michael Karcher 250f32747e ALSA: hda - clean up CX20549 test mixer setup
name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.

remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1

really select MIC1, not Mixer out for recording

"Mixer out" for recording is not a "pin", adjust comment

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:56 +02:00
Michael Karcher 51969d62c3 ALSA: hda - CX20549 doesn't need pin_amp_workaround.
CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:

  $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
  nid = 0x10, verb = 0x300, param = 0xb126
  value = 0x0
  $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
  nid = 0x10, verb = 0xb00, param = 0x8001
  value = 0x0

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:47 +02:00
Michael Karcher e6e03daecd ALSA: hda - Remove CD control from model=benq for CX20549
The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.

Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:40 +02:00
Michael Karcher cbf2d28e83 ALSA: hda - fix record volume controls of CX20459 ("Venice")
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:34 +02:00
Michael Karcher 3edbbb9ec5 ALSA: hda - Rename capture sources of CX20549 to match common conventions
This includes renaming "Line In" to line, also in the mixer settings.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:25 +02:00
Michael Karcher 4f32456e5e ALSA: hda - Fix proc output for ADC amp values of CX20549
The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:02 +02:00
Stephen Warren c2f6702d31 ASoC: tegra: utils: add support for Tegra30 devices
Tegra30 has some additional clocks that need to be manipulated, names
some clocks differently, runs PLLs at different base rates, etc. The
utility code needs to handle this.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 18:31:05 +01:00
Stephen Warren a9005b67b3 ASoC: tegra: set a sensible initial clock rate
Initialize the audio clock tree appropriately for some reasonable rate.
This makes sure the PLLs etc. are actually programmed to something
reasonable when the audio driver is loaded.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 18:25:16 +01:00
Stephen Warren 17933db2e4 ASoC: tegra: use devm_ APIs in SPDIF driver
The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 18:22:06 +01:00
Stephen Warren 7613c508df ASoC: tegra: sort includes, remove mach/iomap.h
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 18:22:06 +01:00
Maarten ter Huurne cd11536742 ASoC: JZ4740: Replaced comma operators with semicolons.
They were harmless but also unnecessary, probably a leftover from earlier code.

Signed-off-by: Maarten ter Huurne <maarten@treewalker.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 18:22:05 +01:00
Stephen Warren 896637ac1b ASoC: tegra: complete Tegra->Tegra20 renaming
Rename Tegra20-specific Kconfig variables, module filenames, all internal
symbol names, clocks, and platform devices, to reflect the fact the DAS
and I2S drivers are for a specific HW version.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 17:36:15 +01:00
Stephen Warren ef280d3907 ASoC: tegra: rename Tegra20-specific driver files
Rename these files so they include a specific hardware version in their
filenames. The contents is only touched minimally so that git's rename
tracking operates correctly; renaming all symbols in the files results
in a diff so large that the rename detection fails.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-06 14:17:45 +01:00
Stephen Warren 8abe05c6eb ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed
the prototype of tegra_i2s_debug_add, but didn't update the dummy inline
used when !CONFIG_DEBUG_FS. Fix that.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # 3.3
2012-04-06 12:28:49 +01:00
Linus Torvalds 5d32c88f0b Merge branch 'akpm' (Andrew's patch-bomb)
Merge batch of fixes from Andrew Morton:
 "The simple_open() cleanup was held back while I wanted for laggards to
  merge things.

  I still need to send a few checkpoint/restore patches.  I've been
  wobbly about merging them because I'm wobbly about the overall
  prospects for success of the project.  But after speaking with Pavel
  at the LSF conference, it sounds like they're further toward
  completion than I feared - apparently davem is at the "has stopped
  complaining" stage regarding the net changes.  So I need to go back
  and re-review those patchs and their (lengthy) discussion."

* emailed from Andrew Morton <akpm@linux-foundation.org>: (16 patches)
  memcg swap: use mem_cgroup_uncharge_swap fix
  backlight: add driver for DA9052/53 PMIC v1
  C6X: use set_current_blocked() and block_sigmask()
  MAINTAINERS: add entry for sparse checker
  MAINTAINERS: fix REMOTEPROC F: typo
  alpha: use set_current_blocked() and block_sigmask()
  simple_open: automatically convert to simple_open()
  scripts/coccinelle/api/simple_open.cocci: semantic patch for simple_open()
  libfs: add simple_open()
  hugetlbfs: remove unregister_filesystem() when initializing module
  drivers/rtc/rtc-88pm860x.c: fix rtc irq enable callback
  fs/xattr.c:setxattr(): improve handling of allocation failures
  fs/xattr.c:listxattr(): fall back to vmalloc() if kmalloc() failed
  fs/xattr.c: suppress page allocation failure warnings from sys_listxattr()
  sysrq: use SEND_SIG_FORCED instead of force_sig()
  proc: fix mount -t proc -o AAA
2012-04-05 15:30:34 -07:00
Stephen Boyd 234e340582 simple_open: automatically convert to simple_open()
Many users of debugfs copy the implementation of default_open() when
they want to support a custom read/write function op.  This leads to a
proliferation of the default_open() implementation across the entire
tree.

Now that the common implementation has been consolidated into libfs we
can replace all the users of this function with simple_open().

This replacement was done with the following semantic patch:

<smpl>
@ open @
identifier open_f != simple_open;
identifier i, f;
@@
-int open_f(struct inode *i, struct file *f)
-{
(
-if (i->i_private)
-f->private_data = i->i_private;
|
-f->private_data = i->i_private;
)
-return 0;
-}

@ has_open depends on open @
identifier fops;
identifier open.open_f;
@@
struct file_operations fops = {
...
-.open = open_f,
+.open = simple_open,
...
};
</smpl>

[akpm@linux-foundation.org: checkpatch fixes]
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-04-05 15:25:50 -07:00
Mark Brown 5fa87d3484 ASoC: wm8400: Use snd_soc_write() and snd_soc_read()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 22:05:31 +01:00
M R Swami Reddy 7e811ae74b ASoC: lm49453: fix build warnings
sound/soc/codecs/lm49453.c: In function 'lm49453_set_dai_fmt':
sound/soc/codecs/lm49453.c:1189:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1193:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1197:4: warning: overflow in implicit
constant conversion [-Woverflow]

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 22:01:19 +01:00
Stephen Warren a7fda2ba82 ASoC: tegra: make Tegra20 drivers depend on Tegra20 support
Without this, the Tegra20 drivers can be built into a kernel that's
built only for Tegra30.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 21:57:37 +01:00
Stephen Warren 063dd9d448 ASoC: tegra: drop Kconfig description for SND_SOC_TEGRA_DAS
The DAS, I2S, and SPDIF Kconfig options are intended to be selected by
the Kconfig options for ASoC machine drivers. As such, they don't need
to be user-visible themselves. Drop the description from the DAS variable
to achieve this. I2S and SPDIF already don't have a description.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 21:57:37 +01:00
Stephen Warren 3fec6b6d5a ASoC: set idle_bias_off=1 for all platform DAPM contexts
The ASoC core currently defaults to using STANDBY rather than OFF for
idle ASoC platform devices, which causes a permanent pm_runtime_get() on
them. This keeps the device active unnecessarily. This can be especially
problematic when the ASoC platform device and DAI device are the same
device.

The distinction between OFF and STANDBY is likely not relevant for ASoC
platform drivers, since they aren't analog devices. So, solve this issue
by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this
turns out to be a problem, this value could be sourced from the
snd_soc_platform_driver, similarly to soc_probe_codec().

Note: Prior to this change, this caused a large (10) runtime_active count
for the Tegra I2S controller even when not in use, and a leak in that
value as streams were started and stopped. This change probably hides a
bug.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 21:55:49 +01:00
David Henningsson 18dcd3044e ALSA: hda - Fix internal mic for Lenovo Ideapad U300s
The internal mic input is phase inverted on one channel.
To avoid people in userspace summing the channels together
and get zero result, use a separate mixer control for the
inverted channel.

BugLink: https://bugs.launchpad.net/bugs/903853
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-05 17:54:22 +02:00
Fabio Estevam 66bb2a7f83 ASoC: imx-audmux: Check for NULL pointer
Check for NULL pointer before accessing it.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 15:23:20 +01:00
Fabio Estevam 00792ac4e0 ASoC: imx-audmux: Fix ssi port numbers in sysfs
Doing a 'cat /sys/kernel/debug/audmux/ssi7' causes the following oops to be printed by the kernel:

Uhandled fault: external abort on non-linefetch (0x008) at 0xf53b003c
Internal error: : 8 [#1] PREEMPT
Modules linked in:
CPU: 0    Not tainted  (3.3.0-00033-gecc726e-dirty #307)
PC is at audmux_read_file+0x68/0x2f4
LR is at clk_enable+0x3c/0x48
pc : [<c001b8c8>]   lr : [<c00190a0>]    psr: a0000013
sp : c3ad3f38  ip : c30a4000  fp : 00000003
r10: 00001000  r9 : be83fb00  r8 : c3ad3f80
r7 : c3ad3f80  r6 : 00000007  r5 : 00031010  r4 : c30a5000
r3 : f53b0000  r2 : 0000003c  r1 : 380fa100  r0 : c068dda0
Flags: NzCv  IRQs on  FIQs on  Mode SVC_32  ISA ARM  Segment user
Control: 0005317f  Table: 83034000  DAC: 00000015
Process cat (pid: 1042, stack limit = 0xc3ad2270)
Stack: (0xc3ad3f38 to 0xc3ad4000)
3f20:                                                      c3139180 00000000
3f40: c3bc6500 00001000 be83fb00 c3ad3f80 00001000 c3ad2000 00000000 c0095f3c
3f60: 00000003 c3bc6508 c3bc6500 be83fb00 00000000 00000000 00001000 c0096010
3f80: 00000000 00000000 b6fe2050 00000000 00001000 be83fb00 00000003 00000003
3fa0: c000eb88 c000e9e0 00001000 be83fb00 00000003 be83fb00 00001000 00000000
3fc0: 00001000 be83fb00 00000003 00000003 00000001 00000001 00000000 00000003
3fe0: 000bec8c be83fae0 0000f808 b6ea8d5c 60000010 00000003 7dff7ede 749bedf1
[<c001b8c8>] (audmux_read_file+0x68/0x2f4) from [<c0095f3c>] (vfs_read+0xb0/0x144)
[<c0095f3c>] (vfs_read+0xb0/0x144) from [<c0096010>] (sys_read+0x40/0x70)
[<c0096010>] (sys_read+0x40/0x70) from [<c000e9e0>] (ret_fast_syscall+0x0/0x2c)
Code: e1a02186 e2822004 e3500000 e7935186 (e7937002)
---[ end trace 4d046e31309023de ]---

Fix the ssi port numbers in sysfs to fix this problem.

Reported-by: Joan Carles <joancarles@fqingenieria.es>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 15:23:17 +01:00
Kuninori Morimoto 1f99e44cf0 ASoC: ak4642: fixup: mute needs +1 step
ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute.
But current settings didn't care +1 step for mute.
This patch adds it

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-04-05 11:08:45 +01:00
Mark Brown e6968a1719 ASoC: codecs: Remove rtd->codec usage from CODEC drivers
In order to support CODEC<->CODEC links remove the assumption that there
is only a single CODEC on a DAI link by removing the use of the CODEC
pointer in the rtd from the CODEC drivers. They are already being passed
their DAI whenever they are passed an rtd and can get the CODEC from
there.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-04 15:59:12 +01:00