- 18 Nov, 2008 8 commits
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Bryan Wu authored
Cc: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Bryan Wu <cooloney@kernel.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mike Frysinger authored
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>
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Michael Hennerich authored
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>
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Cliff Cai authored
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>
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Cliff Cai authored
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>
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Cliff Cai authored
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>
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Mike Frysinger authored
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>
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Naresh Medisetty authored
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>
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- 17 Nov, 2008 7 commits
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Mark Brown authored
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>
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Mark Brown authored
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>
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Mark Brown authored
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>
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Mark Brown authored
Only fully documented registers are cached in the WM8990 but additional registers exist. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Christian Pellegrin authored
Signed-off-by: Christian Pellegrin <chripell@fsfe.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Christian Pellegrin authored
Signed-off-by: Christian Pellegrin <chripell@fsfe.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
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>
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- 14 Nov, 2008 2 commits
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Mark Brown authored
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>
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Mark Brown authored
This reverts commit 8dc840f8. 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>
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- 13 Nov, 2008 2 commits
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Jarkko Nikula authored
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>
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Jarkko Nikula authored
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>
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- 12 Nov, 2008 1 commit
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Naresh Medisetty authored
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>
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- 10 Nov, 2008 2 commits
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Hugo Villeneuve authored
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>
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Christian Pellegrin authored
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>
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- 07 Nov, 2008 1 commit
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Troy Kisky authored
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>
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- 06 Nov, 2008 1 commit
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Grazvydas Ignotas authored
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>
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- 05 Nov, 2008 3 commits
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David Anders authored
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>
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Mark Brown authored
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>
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Marek Vasut authored
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>
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- 04 Nov, 2008 3 commits
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Takashi Iwai authored
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Huang Weiyi authored
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>
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Troy Kisky authored
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>
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- 31 Oct, 2008 8 commits
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Takashi Iwai authored
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Takashi Iwai authored
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Sedji Gaouaou authored
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>
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Sedji Gaouaou authored
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>
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Steve Sakoman authored
Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Steve Sakoman authored
Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Steve Sakoman authored
Signed-off-by: Steve Sakoman <steve@sakoman.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Stephen Rothwell authored
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>
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- 30 Oct, 2008 2 commits
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Takashi Iwai authored
The last change to Kconfig ca53fb24 added a wrong item SND_SOC_AC97, which must be SND_SOC_AC97_CODEC. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Timur Tabi authored
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>
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