- 18 Feb, 2009 10 commits
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Peter Ujfalusi authored
This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. 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
Completely untested. 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
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Move the WM8731 driver to use a more standard device registration scheme where the device can be registered independantly of the ASoC probe. As a transition measure push the current manual code for registering the WM8731 into the individual machine driver probes. This allows separate patches to update the relevant architecture files with less risk of merge issues. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
This is a pure code motion patch intended to improve reviewability of a following patch moving WM8731 to use more standard device registration. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
This is a bit more idiomatic and makes identifying a configuration based on the board type work better. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
We have software control of the MCLK for the WM8731 so save a bit of power by actively managing it within the machine driver, enabling it only while the codec is active. Once ASoC supports multiple boards and doesn't require the soc-audio device the initial clock setup should be pushed down into the arch/arm code but for now this reduces merge issues. Tested-by: Sedji Gaouaou <sedji.gaouaou@atmel.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
We should display an error by default if we fail to register. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 Feb, 2009 3 commits
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Mark Brown authored
The WM8731 bias level configuration function was written slightly obscurely - streamline the code a little and refresh the comments. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
It makes boot a bit more noisy and I never remember to update it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Paul Fertser authored
WM8753 uses a tricky way to switch DAIs "on the fly", for that it registers 2 dummy DAIs and substitutes them depending on mixer control. List element of registered dummy DAIs should be preserved to allow unregistering of DAIs on module unload. Signed-off-by: Paul Fertser <fercerpav@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 Feb, 2009 5 commits
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Kevin Hilman authored
Fix for the error when the audio module is unloaded. On unregistering the platform_device, platform_device_release will free the platform data.If platform data is static the kernel panics when it is freed. Instead use the platform device helper function to add data. This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Takashi Iwai authored
Merge branch 'for-2.6.30' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc
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Mark Brown authored
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Mark Brown authored
ASoC supports both explicit codec drivers for AC97 devices and a simple driver which uses the standard ALSA AC97 framework for codec support. When used with the generic AC97 codec support that will provide the ad hoc AC97 device for drivers like touchscreens to attach to so the core shouldn't do so. Reported-by: Manuel Lauss <mano@roarinelk.homelinux.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Timur Tabi authored
Update the CS4270 codec driver to allow applications to use the mixer to control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute. Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first initializes the hardware, but these features either don't work or interfere with normal ALSA behavior. However, they can now be re-enabled by an application if desired. Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute bits. The driver previously and erroneously assumed that these bits control only external muting circuitry, but they also control internal muting circuitry, so they should always be used. Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 11 Feb, 2009 1 commit
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Takashi Iwai authored
The snd_soc_codec was moved into socdev->card, but this change wasn't applied in some places. Fixed now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 10 Feb, 2009 2 commits
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Mark Brown authored
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Lopez Cruz, Misael authored
This patch replaces "snd_soc_machine" structure by "snd_soc_card" in SP3430 driver. This change is needed in SDP3430 driver to reflect changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch (87506549). Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 Feb, 2009 2 commits
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Jarkko Nikula authored
TLV320AIC3X volume controls are logarithmic. Export their dB ranges. 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
This is a minor fix but helps to define dB ranges for volume controls. Only DAC digital volume has full register value range from 0 to 127 but ADC PGA gain and output stage volume controls don't. For ADC PGA, maximum value is 119 and then it saturates to the same gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB and is muted for values 118 and above. Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 08 Feb, 2009 5 commits
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Philipp Zabel authored
This removes the calls to pxa_gpio_mode from the pxa2xx-i2s driver. Pin setup should be done during board init via pxa2xx_mfp_config instead. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Eric Miao <eric.miao@marvell.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Philipp Zabel authored
The spitz has a WM8750 codec connected as I2S slave but doesn't use the PXA I2S system clock. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Eric Miao <eric.miao@marvell.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Philipp Zabel authored
The iPAQ h5000 has an AK4535 codec connected as I2S slave, PXA I2S providing SYSCLK. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Eric Miao <eric.miao@marvell.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Roel Kluin authored
With a postfix increment count reaches 10001, not 10000. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Robert Jarzmik authored
This machine driver enables sound functions on Mitac mio a701 smartphone. Build upon ASoC v1, it handles : - rear speaker - front speaker - microphone - GSM A global "Mio Mode" switch is not yet provided to cope with audio path setup. As balance on audio chip line is no more assured, an incorrect setup can produce a lot of heat and even fry the battery behind the wm9713 and the speaker amplifier. It doesn't cope with : - headset jack - mio master mode - master volume control This driver is backported from ASoc v2, and amputated from scenario setups and master volume control. [Minor mods for terminology in comments -- broonie] Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 06 Feb, 2009 6 commits
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Mark Brown authored
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Timur Tabi authored
In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mike Frysinger authored
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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Mike Frysinger authored
- make sport number handling more dynamic as not all Blackfins have a linear sport map starting at 0 - indexes can be macroed away too Signed-off-by: Mike Frysinger <vapier.adi@gmail.com> 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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Jarkko Nikula authored
Function wm899x_outpga_put_volsw_vu misuses the kcontrol's private value by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it as a pointer into struct soc_mixer_control after the commit 4eaa9819. This is very similar fix than fix to TLV320AIC3X codec made by Eero Nurkkala <ext-eero.nurkkala@nokia.com>. This fix is compile tested only. Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Cc: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Eero Nurkkala authored
Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it as a pointer into struct soc_mixer_control after the commit 4eaa9819. This was causing arbitrary register writes when touching the controls defined with SOC_DAPM_SINGLE_AIC3X. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 04 Feb, 2009 2 commits
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Philipp Zabel authored
PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function requested the wrong SSP port. Correcting this unveiled another bug where ssp_init tries to request the already-requested SSP port again. So this patch replaces the ssp_init/exit calls with their internals from mach-pxa/ssp.c, leaving out the redundant ssp_request and the unneeded IRQ request. Effectively, that leaves us with not much more than enabling/disabling the SSP clock. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Philipp Zabel authored
This patch splits set_dai_fmt into three variants (single interface, dual interface playback only, dual interface capture only) so that data input and output formats can be configured separately for dual interface setups. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Tested-by: Vasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 03 Feb, 2009 3 commits
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Vasily Khoruzhick authored
Without this fix driver switches to WSPLL in uda1380_pcm_prepare even if SYSCLK was chosen (uda1380_pcm_prepare modifies UDA1380_CLK register to disable R00_DAC_CLK before flushing reg cache) Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Takashi Iwai authored
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Liam Girdwood authored
This just updates my email address on some drivers I'd forgotten in a previous patch. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 02 Feb, 2009 1 commit
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Timur Tabi authored
Replace printk calls with dev_xxx calls. Set the 'dev' field of the codec and codec_dai structures so that these calls work. Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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