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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
The control had an extra space at the end of the name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix missing unsigned for irqsave flags in psc i2s driver
Make attribute visiblity static
Collect all sysfs errors before checking status
[Word wrapped DEVICE_ATTR() lines for 80 columns -- broonie]
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Setting the TFS pin selector for SPORT 0 based on whether the selected
port id F or G. If the port is F then no conflict should exist for the
TFS. When Port G is selected and EMAC then there is a conflict between
the PHY interrupt line and TFS. Current settings prevent the conflict
by ignoring the TFS pin when Port G is selected. This allows both
ssm2602 using Port G and EMAC concurrently.
- some code cleanup
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Codec doesn't support to configure bit clock and frame sync polarities
- Codec doesn't support DSP_A format but DSP_B with inverted bit clock
polarity
- Match also other formats with their signal polarities
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix "ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver" was not correct
due misunderstanding of DSP_A format and similar error in TLV320AIC33
codec which was used to test the original fix.
This patch corrects now DSP_A format in OMAP McBSP DAI driver and is
verified with TLV320AIC23 codec that's implementing DSP_A correctly.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix word clock length which must equal to one bit clock cycle in DSP mode.
Surprisingly McBSP is able synchronize into wrong length when it's
slave but e.g. TLV320AIC33 codec in slave configuration is outputting
some amount of noise if word clock length is longer than one bit clock
cycle.
Fix also bit clock and frame sync polarities in DSP mode since they are
opposite from I2S.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
debugfs_create_dir() returns NULL if an error occurs, returns -ENODEV
when debugfs is not enabled in the kernel.
Signed-off-by: Zhao Lei <zhaolei@cn.fujitsu.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was marked as deprecated in 2.6.27 and all users except for
playpaq_wm8510 fixed in that release.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the wm8900 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the wm8580 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the visibility of the tristate conditional on having the OpenFirmware
helper code enabed so that users who can't use it don't see the visible
option. Kconfig ignores dependencies for select so other users are
unaffected.
Thanks to Takashi for the suggestion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
ALSA: ASoC codec: remove unused #include <version.h>
ALSA: ASoC: update email address for Liam Girdwood
ALSA: hda: corrected invalid mixer values
ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
ALSA: ASoC: Add destination and source port for DMA on OMAP1
ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
ALSA: ASoC: Fix build of GTA01 audio driver
ALSA: ASoC: Add widgets before setting endpoints on GTA01
ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
ALSA: ASoC: Make TLV320AIC26 user-visible
ALSA: ASoC - clean up Kconfig for TLV320AIC2
ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
ALSA: ASoC: Implement WM8510 bias level control
ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
ALSA: ASoC: Add WM8510 SPI support
ALSA: ASoC: Add WM8753 SPI support
...
Fixes this warning:
sound/soc/codecs/tlv320aic23.c: In function 'tlv320aic23_write':
sound/soc/codecs/tlv320aic23.c:104: warning: passing argument 2 of
'codec->hw_write' makes pointer from integer without a cast
Replaces i2c smbus write function with standard i2c write function
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
