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Commit Graph

483 Commits

Author SHA1 Message Date
Hugo Villeneuve
1c0090c280 ASoC: Add PCM3008 ALSA SoC driver
The PCM3008 is a 16-bit stereo audio codec. It accepts
left-justified format for ADC, and right-justified format
for DAC. Independent power-down modes for ADC and DAC are
provided, as well as a digital de-emphasis filter (4 modes).

[Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie]

Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-19 13:16:38 +00:00
Mark Brown
72f2b89445 ASoC: Move uda134x_codec.h to uda134x.h
For consistency with other ASoC codec drivers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 12:32:33 +00:00
Mike Frysinger
a0bd65f45f ASoC: Blackfin: always set a default value for that GPIO range
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 12:32:33 +00:00
Bryan Wu
27b9be5a78 ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 12:32:32 +00:00
Mike Frysinger
caa45836d6 ASoC: Blackfin: do not force TWI bus for ssm2602 codec
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 12:32:32 +00:00
Michael Hennerich
0cade26e36 ASoC: Fix Blackfin AC97 DAI probe function return code
A probe function should have a clean return 0 path.

Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Michael Hennerich <michael.hennerich@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 12:32:11 +00:00
Cliff Cai
a89e611a1d ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled
clean up redudent code and correct building problem in non-mmap mode

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 11:40:21 +00:00
Cliff Cai
67f854b910 ASoC: Blackfin: add multi-channel function support
This patch provides a option for users to enable multi-channel function support
in Blackfin ASoC driver. Because Blackfin is without MMU, it is easy for us and
the user to enable this function at compiling stage not dynamically on the fly.

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 11:40:19 +00:00
Cliff Cai
9905ed35fd ASoC: AD1980 codec: add multi-channel function support
We added multi-channel function to this codec driver and Blackfin ASoC driver as well.
It was tested on Blackfin hardware.

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 11:40:18 +00:00
Mike Frysinger
a11311d71d ASoC: Blackfin: updates Kconfig for SPORT
tweak SPORT range for non-BF54x so we get proper behavior for BF52x parts

Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 11:40:18 +00:00
Naresh Medisetty
cb6e206369 ASoC: DaVinci: Fix audio stall when doing full duplex
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.

Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 11:40:04 +00:00
Mark Brown
8d702f2376 ASoC: Build tlv320aic23 cleanly
Also merge down a couple of last minute style changes that got lost in the
shuffle.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 21:46:24 +00:00
Mark Brown
2adb9833d1 ASoC: Manage VMID mode for WM8990
A small additional power saving can be achieved for the WM8990 by
maintaining VMID using a 2*250k divider when in standby mode.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 17:26:51 +00:00
Mark Brown
be1b87c70a ASoC: Enable WM8990 ADC clocking workaround
Enable a hardware workaround which avoids problems with the clocking of
the ADCs in certain configurations.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 17:24:54 +00:00
Mark Brown
ba533e95b9 ASoC: Allow writes to uncached registers in WM8990
Only fully documented registers are cached in the WM8990 but additional
registers exist.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 17:24:50 +00:00
Christian Pellegrin
7ad933d7a6 ASoC: Machine driver for for s3c24xx with uda134x
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 11:45:53 +00:00
Christian Pellegrin
1cad1de1b2 ASoC: UDA134x codec driver
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 11:45:39 +00:00
Peter Ujfalusi
6e5d9db271 ASoC: Fix for master playback/capture volume range for TWL4030 codec
FGAIN for playback is in range of 0-0x3f, while for capture GAIN it
is in the range of 0-0x1f.
The original value of 128 (0x7f) would modify the CGAIN also for
playback.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-17 11:02:21 +00:00
Mark Brown
71cfc9028d ASoC: Add WM8728 codec driver
The WM8728 is a high performance stereo DAC designed for applications
such as DVD, home theatre and digital TV.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-14 14:44:53 +00:00
Mark Brown
2bef901071 ASoC: Revert "ASoC: Add new parameter to s3c24xx_pcm_enqueue"
This reverts commit 8dc840f88d.  Christian
Pellegrin <chripell@gmail.com> reported that on some systems the patch
caused DMA to fail which is much more serious than the original skipped
audio issue.  Further investigation by Dave shows that the behaviour
depends on the clock speed of the SoC - a better fix is neeeded.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-14 14:40:46 +00:00
Jarkko Nikula
0b6048561d ASoC: OMAP: Add more supported sample rates into McBSP DAI driver
Originally it was put too tight limits to support only 44.1 kHz and 48 kHz
sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With
96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?).

Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas
Instruments Beagle with TWL4030 from rates 8 - 48 kHz.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-13 10:31:44 +00:00
Jarkko Nikula
bbba944410 ASoC: Fix supported sample rates of TWL4030 audio codec
TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec
mode register accordingly in twl4030_hw_params. Expose this info so that
ASoC can match other rates than 44.1 kHz or 48 kHz as well.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-13 10:31:22 +00:00
Naresh Medisetty
fb0ef645f2 ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback
Fixes swapping of channels at start of stereo playback.

Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.

Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-12 11:50:29 +00:00
Hugo Villeneuve
b402dff873 ASoC: Add Right-Justified mode and Codec clock master to davinci-i2s
The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the
Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.

Signed-off-by: Hugo Villeneuve <hugo.villeneuve@lyrtech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-10 11:41:18 +00:00
Christian Pellegrin
53599bbc30 ASoC: s3c24xx 8 bit sound fix
fixes playing/recording of 8 bit audio files.

Generated on  20081108  against v2.6.27

Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-10 11:40:57 +00:00
Troy Kisky
26df91c36f ASoC: TLV320AIC23B Support more sample rates
Add support for more sample rates, different crystals
and split playback/capture rates.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-07 13:27:47 +00:00
Grazvydas Ignotas
e18c94d202 ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
According to TRM, 256*Fs clock output should be enabled
when TWL4030 is in slave mode, not master.
This allows sound to work on OMAP3 Pandora, which uses
256*Fs clock.

Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-06 11:27:30 +00:00
David Anders
8dc840f88d ASoC: Add new parameter to s3c24xx_pcm_enqueue
The S3C24xx dma does not allow more than one buffer to be enqueue prior to
the dma transfers starting. This patch adds an additional parameter to
s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load
value.

Signed-off-by: David Anders <danders at amltd.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-05 22:11:31 +00:00
Mark Brown
ea913940c3 ASoC: Remove core version number
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it.  It was much more useful
when ASoC was out of tree.

Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
2008-11-05 22:11:29 +00:00
Marek Vasut
74e722015f ASoC: Add Palm/PXA27x unified ASoC audio driver
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.

[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-05 22:11:16 +00:00
Takashi Iwai
0ee4663617 ALSA: ASoC - Remove unnecessary inclusion of linux/version.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-11-04 18:06:23 +01:00
Huang Weiyi
3865675c60 ALSA: ASoC codec: remove unused #include <version.h>
The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION.
  sound/soc/codecs/ad73311.c

This patch removes the said #include <version.h>.

Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-11-04 18:03:21 +01:00
Troy Kisky
dce908e26f ALSA: SOC: Fix setting codec register with debugfs filesystem merge error
Call device_create_file only once in snd_soc_dapm_sys_add function.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-11-04 08:40:55 +01:00
Takashi Iwai
7b3b6e4203 Merge commit 'v2.6.28-rc2' into topic/asoc 2008-10-31 17:13:10 +01:00
Takashi Iwai
04172c0b9e Merge branch 'topic/fix/asoc' into topic/asoc 2008-10-31 14:39:49 +01:00
Sedji Gaouaou
5b99e6ccf9 ASoC: Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731).
Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731).
It is based on the former eti_b1_wm8731.c file, using the atmel scc API.

Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 13:13:20 +00:00
Sedji Gaouaou
6c7425095c ASoC: Merge AT91 and AVR32 support into a single atmel architecture
The Ateml AT91 and AVR32 SoC share common IP for audio and can share the
same driver code using the atmel-ssc API provided for both architectures.
Do this, creating a new unified atmel ASoC architecture to replace the
previous at32 and at91 ones.

[This was contributed as a patch series for reviewability but has been
squashed down to a single commit to help preserve both the history and
bisectability.  A small bugfix from Jukka is included.]

Tested-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 13:12:26 +00:00
Steve Sakoman
dc06102a0c ASoC: Add support for Beagleboard
Signed-off-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:33:43 +00:00
Steve Sakoman
4e20787373 ASoC: Add support for Gumstix Overo
Signed-off-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:33:38 +00:00
Steve Sakoman
cc17557e78 ASoC: Add support for TWL4030 audio codec
Signed-off-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-31 12:33:27 +00:00
Stephen Rothwell
57b41898c2 ALSA: ASoC - restore removed variable declaration
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_sys_add':
sound/soc/soc-dapm.c:828: error: 'ret' undeclared (first use in this function)

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-10-31 07:32:12 +01:00
Takashi Iwai
0763722d28 ALSA: ASoC - Fix a typo in Kconfig
The last change to Kconfig ca53fb24dd
added a wrong item SND_SOC_AC97, which must be SND_SOC_AC97_CODEC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-10-30 17:53:19 +01:00
Timur Tabi
0c235d1e83 ASoC: Disable automatic volume control in the CS4270 sound driver
Disable the automatic volume control feature of the CS4270 audio codec.  This
feature, which is enabled by default, causes volume change commands to be
delayed.  Sometimes the volume change happens after playback is started.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 15:54:11 +00:00
Mark Brown
ca53fb24dd ASoC: Use finer grained dependencies in SND_SOC_ALL_CODECS
Move the bus dependencies in SND_SOC_ALL_CODECS into the individual
codec options rather than have them centrally. This allows the
inclusion of AC97 codecs when testing on platforms with AC97 support
and will also handle codecs on multi-function devices more gracefully.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 15:54:10 +00:00
Mark Brown
e775f6c0fb ASoC: Do a warm reset after cold when resetting the WM9713
The WM9713 comes out of cold reset in low power mode so always requires
a warm reset to bring up the AC97 link after a cold reset.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 15:54:10 +00:00
Mark Brown
1b340bd7e4 ASoC: Add PXA SSP support
The SSP ports PXA series processors can be used to implement a variety of
audio interface formats. This patch implements support for I2S, DSP A and
DSP B modes on these ports.

This patch is based on the previous out of tree pxa2xx-ssp driver (which
was originally written by Liam Girdwood with updates from Philipp Zabel
and Nicola Perrino) and pxa3xx-ssp driver (originally written by Seth
Forsee based on the pxa2xx-ssp driver). Testing coverage is not complete
currently.

Tested-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 15:45:21 +00:00
Mark Brown
219b93f525 ASoC: Remove DAPM restriction on mixer control name lengths
As well as ensuring that UI-relevant parts of control names don't get
truncated in the DAPM code this avoids conflicts in long control names
that differ only at the end of a long string.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 14:34:03 +00:00
Mark Brown
f24368c2fb ASoC: Convert core to use standard debug print macros
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 14:34:03 +00:00
Mark Brown
d45f6219d2 ASoC: Fix handling of DAPM suspend work
Since we can query the playback stream power state directly we do not
need to infer if it is powered up from the timer being scheduled.  Doing
this avoids problems that previously existed with streams being
incorrectly determined to be powered up caused when the timer is
scheduled when streams are closed after being partially set up.

Reported-by: Nobin Mathew <nobin.mathew@gmail.com>
Reported-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 14:34:03 +00:00
Troy Kisky
12ef193d58 ASoC: Allow setting codec register with debugfs filesystem
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 14:34:02 +00:00