- 25 Jan, 2010 1 commit
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Guennadi Liakhovetski authored
Many macros from include/sound/soc-dapm.h take an array and a number of elements in it as arguments, whereas most users use static arrays and use "x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of those macros, calling ARRAY_SIZE() internally. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 21 Jan, 2010 3 commits
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Mark Brown authored
This provides a small power saving when audio is inactive. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
The same information is now visible via debugfs and with large modern devices dumping everything to the console can be very resource intensive, causing more harm than good. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 20 Jan, 2010 2 commits
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Peter Ujfalusi authored
Add possibility to configure the burst mode BCLK divider through platform data structure. The BCLK divider changes the actual speed of the serial bus in burst mode, which is faster than the sampling frequency of the running stream. In this way platforms can experiment with the optimal burst speed without the need to modify the codec driver itself. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
The BCLK divider was not configured in case of mode7. This leads to unpredictable behavior when switching between FIFO modes. Configure the BCLK divider depending on the fifo_mode (FIFO is in use, or FIFO bypass). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 Jan, 2010 2 commits
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Guennadi Liakhovetski authored
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 Jan, 2010 7 commits
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Mark Brown authored
Conflicts: arch/arm/plat-mxc/Makefile (dual add) sound/soc/imx/mx27vis_wm8974.c (API updates & removal)
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Mark Brown authored
Currently they don't build due to cross tree dependencies, they will be reenabled once the arch/arm side has merged. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
This has been superceeded by Sascha's new driver but was not removed in the patch series due to cutdowns for review. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
The clocks for the SSI block need handling before this can work. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
While dynamically allocated DAIs are the way forward the core doesn't yet support anything except matching with a pointer to the actual DAI so convert to doing that so that machine drivers don't have to jump through hoops to register themselves. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Javier Martin <javier.martin@vista-silicon.com>
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Mark Brown authored
Don't unconditionally include the i.MX2x DMA driver, the arch/arm functions it uses aren't available for i.MX3x. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Javier Martin <javier.martin@vista-silicon.com>
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Sascha Hauer authored
The old driver has the number of SSI units in the system hardcoded, does not make use of the device model and works only on i.MX21/27. This driver replaces it. It works in DMA mode on i.MX21/27 and using an FIQ handler on other systems. It also supports AC97 mode of the SSI units. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Javier Martin <javier.martin@vista-silicon.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 Jan, 2010 2 commits
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Daniel Mack authored
Add support for sample rates other than 44100Khz on raumfeld audio devices. At startup time, call snd_soc_dai_set_sysclk() with 0 as 'freq' argument so it offers all the sample rates. Later, the function is called again to give proper constraints. Use the external audio clock generator to provide double data rate clocks as the PXA's internal baud generator does anything but what's described in the datasheets. Signed-off-by: Daniel Mack <daniel@caiaq.de> Cc: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Daniel Mack authored
For setups with variable MCLKs, the current logic of limiting the available sampling rates at startup time is not sufficient. We need to be able to change the setting at a later point, and so the codec must offer all possible rates until the hw_params are given. This patches allows that by passing 0 as 'freq' argument to cs4270_set_dai_sysclk(). Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 14 Jan, 2010 1 commit
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Thomas Weber authored
Signed-off-by: Thomas Weber <weber@corscience.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 Jan, 2010 2 commits
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Daniel Mack authored
The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 12 Jan, 2010 3 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
This reverts commit afe1c2cd since it doesn't build.
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Mark Brown authored
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- 11 Jan, 2010 1 commit
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Ilkka Koskinen authored
tpa6140a2 uses different names for the regulators. Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 08 Jan, 2010 1 commit
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Mark Brown authored
We need to set the LRCLK inversion bit to select DSP mode. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
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- 04 Jan, 2010 7 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Manuel Lauss authored
Initialize the glue by calling snd_soc_new_ac97_codec() as is done in other ASoC AC97 codecs. Fixes an oops caused by dereferencing uninitialized members in snd_soc_new_pcms(). Run-tested on Au1250. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Ilkka Koskinen authored
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 31 Dec, 2009 8 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Better integration of the core with the device model means that we now no longer get the ASoC suspend and resume callbacks without the card having been set up. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Peter Ujfalusi authored
The currently available FIFO modes (mode1 and mode7) require master mode from the codec. Do not allow the slave configuration when the FIFO is in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Mode 7 of tlv320dac33 operates in the following way: The codec is in master mode. Host configures upper and lower thresholds in tlv320dac33 During playback the codec will clock in the data until the upper threshold is reached in FIFO. At this point the codec stops the colocks on the serial bus. When the FIFO fill is reaching the lower threshold limit the codec will enable the clocks on the serial bus, and clocks in data till the upper threshold is reached. In this mode, we can also request interrupts for threshold events (upper, lower and alarm), which could be used for power management. At this point the interrupts are not enabled for this mode, but it can be taken into use in the future, when the surrounding code makes it possible to use it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Use switch instead of if statements to configure FIFO bypass and mode1. With this change adding new FIFO mode is going to be easier, and cleaner. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Ensure that the code is going to be readable, when new FIFO modes are introduced later. Move the prefill and playback state handling to inlined functions. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
In order to have support for more FIFO modes supported by tlv320dac33, the switch for enabling/disabling the FIFO use has to be replaced with an enum. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Barry Song authored
Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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