- 02 12月, 2008 12 次提交
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由 Mark Brown 提交于
ASoC v2 does not use the struct snd_soc_device at runtime, using struct snd_soc_card as the root of the card. Begin removing data from snd_soc_device by pushing the workqueue data into snd_soc_card, using a backpointer to the snd_soc_device to keep things going for the time being. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
All outputs have dedicated gain controls except the HandsFree output. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Add Playback volume controls for all four DACs. All four paths has three levels of volume controls: Digital Fine gain, Digital Coarse gain, Analog gain. The controls are named to reflect their connection to the DACs. Per DAC volume can be performed, if needed: amixer sset 'DAC1 Analog' 5,10 DACL1 analog gain to 5 DACR1 analog gain to 10 Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
The digital Capture gain control has a range: 0 to 31 dB in 1 dB steps. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Currently ASoC card initialisation is completed by a function called snd_soc_register_card(). As part of the work to allow independant registration of cards, codecs and machines in ASoC v2 a new function of the same name has been added so rename the existing function to facilitate the merge of v2. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Takashi Iwai 提交于
Fix the old-style trigger callback in s3c2443-ac97.c: sound/soc/s3c24xx/s3c2443-ac97.c:378: warning: initialization from incompatible pointer type Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Fix the wrong shutdown callback type. Also removed the unused variables there: sound/soc/pxa/corgi.c: In function 'corgi_shutdown': sound/soc/pxa/corgi.c:114: warning: unused variable 'codec' sound/soc/pxa/corgi.c: At top level: sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
This reverts commit 9171e5e6. I can't reproduce the compile warnings any more. The warnings might be some weird cross-compiling set up. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The dependency on SND_SOC is already fulfilled in sound/soc/Kconfig, thus no more need in Kconfig of each sub directory. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 12月, 2008 2 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Hide annoying uninitialized warnings: sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 11月, 2008 1 次提交
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由 Daniel Mack 提交于
This patch enables more routing functions for tlv320aic3x codecs. It is now possible to - control the volume of the PGA bypass path for the HPL, HPR, HPLCOM and HPRCOM outputs individually - route right line1 input to the left ADC channel - route left line1 input to the right ADC channel - route right mic3 input to left DAC channel - route left mic3 input to right DAC channel - route left line1 input to right line1 output - route right line1 input to left line1 output Signed-off-by: NDaniel Mack <daniel@caiaq.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 26 11月, 2008 1 次提交
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 25 11月, 2008 9 次提交
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由 Qinghuang Feng 提交于
There is no argument named @clk_id in snd_soc_dai_set_fmt, remove its' comment. Signed-off-by: NQinghuang Feng <qhfeng.kernel@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Dmitry Baryshkov 提交于
Signed-off-by: NDmitry Baryshkov <dbaryshkov@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Misael Lopez Cruz 提交于
This patch add ASoC support for TI SDP3430. It's based on Gumstix Overo SoC code by Steve Sakoman. Signed-off-by: NMisael Lopez Cruz <mesak82@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Arun KS 提交于
Fixes Kconfig dependency of TWL4030 audio codec driver with TWL4030 core driver on both overo and omap2evm boards Signed-off-by: NArun KS <arunks@mistralsolutions.com> Acked-by: NDavid Brownell <dbrownell@users.sourceforge.net> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jarkko Nikula 提交于
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: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jarkko Nikula 提交于
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: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Special handling is required for suspend and resume of AC97 codecs due to the control path going over the data bus. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 11月, 2008 5 次提交
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由 Peter Ujfalusi 提交于
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: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
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: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
TWL4030 FGAIN volume control has a range: -62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
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: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 21 11月, 2008 8 次提交
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由 Mark Brown 提交于
Print something a bit more verbose to help make errors a little more obvious. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
It's not exported. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jarkko Nikula 提交于
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Karl Beldan 提交于
Signed-off-by: NKarl Beldan <karl.beldan@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Karl Beldan 提交于
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: NKarl Beldan <karl.beldan@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 11月, 2008 2 次提交
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Missed these during review. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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