- 26 Nov, 2008 1 commit
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 25 Nov, 2008 6 commits
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Qinghuang Feng authored
There is no argument named @clk_id in snd_soc_dai_set_fmt, remove its' comment. Signed-off-by: Qinghuang Feng <qhfeng.kernel@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Dmitry Baryshkov authored
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Misael Lopez Cruz authored
This patch add ASoC support for TI SDP3430. It's based on Gumstix Overo SoC code by Steve Sakoman. Signed-off-by: Misael Lopez Cruz <mesak82@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Arun KS authored
Fixes Kconfig dependency of TWL4030 audio codec driver with TWL4030 core driver on both overo and omap2evm boards Signed-off-by: Arun KS <arunks@mistralsolutions.com> Acked-by: David Brownell <dbrownell@users.sourceforge.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Jarkko Nikula authored
Patch adds support for mono audio links so that McBSP DAI can operate with real mono codecs. In I2S, the signalling remains the same but only first frame (left channel) is transmitting audio data and second frame having null data. In DSP_A, only first frame is transmitted. Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Jarkko Nikula authored
Prepare for upcoming McBSP DAI update adding support for mono links by restricting number of channels to 2 in N810. This is due tlv320aic3x which claims channels_min = 1 and playing pure mono audio over I2S would cause it to be played only from left channel if both cpu and codec DAI's claim to support mono. Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 Nov, 2008 8 commits
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Mark Brown authored
Now that the ASoC resume has been punted to a workqueue for a release cycle without attracting bug reports it should be safe to make the log messages associated with it debug level, reducing noise and kernel size in production configurations. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Special handling is required for suspend and resume of AC97 codecs due to the control path going over the data bus. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
DAI type information is only ever used within ASoC in order to special case AC97 and for diagnostic purposes. Since modern CPUs and codecs support multi function DAIs which can be configured for several modes it is more trouble than it's worth to maintain anything other than a flag identifying AC97 DAIs so remove the type field and replace it with an ac97_control flag. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Some of the gain controls in TWL (mostly those which are associated with the outputs) are implemented in an interesting way: 0x0 : Power down (mute) 0x1 : 6dB 0x2 : 0 dB 0x3 : -6 dB Inverting not going to help with these. Custom volsw and volsw_2r get/put functions to handle these gains. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Add CGAIN (Coarse gain control) to TWL4030 codec. The range of the CGAIN is: 0 dB to 12 dB in 6 dB steps. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
TWL4030 FGAIN volume control has a range: -62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Keep Soft-volume disabled for now, since if it is enabled the FGAIN volume controls are not working in the current configuration: CODEC_MODE:OPT_MODE = 1 OPTION:ARXR2_EN = 1 OPTION:ARXL2_EN = 1 OPTION:ARXR1_EN = 0 OPTION:ARXL1_VRX_EN = 0 RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1) RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1) After the patch, FGAIN volume control works. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 21 Nov, 2008 11 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Mark Brown authored
Print something a bit more verbose to help make errors a little more obvious. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
It's not exported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Jarkko Nikula authored
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Implement support for the Marvell Zylonite PXA3xx reference platform, supporting standard AC97 stereo and AUX interfaces together with the auxiliary I2S interface of the WM9713. The board has two options for the MCLK of the WM9713: either the standard AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx can be used, selected via SW15 on the board. Currently only the AC97 system clock is supported by this driver. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Liam Girdwood's ASoC v2 work avoids having two different ops structures for DAIs by merging the members of struct snd_soc_ops into struct snd_soc_dai_ops, allowing per DAI configuration for everything. Backport this change. This paves the way for future work allowing any combination of DAIs to be connected rather than having fixed purpose CODEC and CPU DAIs and only allowing CODEC<->CPU interconnections. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
ASoC v2 factors most of the contents of soc.h out into separate headers, including soc-dai.h for the DAI. Factor the existing DAI API out into this file in order to prepare for backporting of the ASoC v2 DAI API. Also backport some of Liam's improvements to the documentation. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Karl Beldan authored
Signed-off-by: Karl Beldan <karl.beldan@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Karl Beldan authored
Clean up our record of the active streams in shutdown(), fixing subsequent failures of snd_pcm_hw_constraints_complete after closure of a stream. NOTE: - The ssm2602 allows pairs of non-matching PB/REC rates. - This is a fix for less evil: The logic is flawed (e.g. the slave might startup before the master's rate and sample_bits are set). Signed-off-by: Karl Beldan <karl.beldan@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 Nov, 2008 5 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Missed these during review. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Arun KS authored
This patch adds twl4030 audio support on omap2evm Signed-off-by: Arun KS <arunks@mistralsolutions.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Hugo Villeneuve authored
The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an FPGA that generates the bit clock and the master clock [Downgraded the rate debug print to pr_debug() in hw_params, converted 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>
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Hugo Villeneuve authored
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>
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- 18 Nov, 2008 9 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
For consistency with other ASoC codec drivers. 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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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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