- 18 2月, 2009 4 次提交
-
-
由 Mark Brown 提交于
This is a bit more idiomatic and makes identifying a configuration based on the board type work better. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
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: NSedji Gaouaou <sedji.gaouaou@atmel.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
We should display an error by default if we fail to register. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 17 2月, 2009 3 次提交
-
-
由 Mark Brown 提交于
The WM8731 bias level configuration function was written slightly obscurely - streamline the code a little and refresh the comments. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
It makes boot a bit more noisy and I never remember to update it. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Paul Fertser 提交于
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: NPaul Fertser <fercerpav@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 14 2月, 2009 1 次提交
-
-
由 Kevin Hilman 提交于
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: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NKevin Hilman <khilman@deeprootsystems.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 13 2月, 2009 4 次提交
-
-
由 Takashi Iwai 提交于
Merge branch 'for-2.6.30' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc
-
由 Mark Brown 提交于
-
由 Mark Brown 提交于
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: NManuel Lauss <mano@roarinelk.homelinux.net> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Timur Tabi 提交于
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: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 11 2月, 2009 3 次提交
-
-
由 Takashi Iwai 提交于
The snd_soc_codec was moved into socdev->card, but this change wasn't applied in some places. Fixed now. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
-
由 Mark Brown 提交于
-
由 Lopez Cruz, Misael 提交于
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: NMisael Lopez Cruz <x0052729@ti.com> Acked-by: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 10 2月, 2009 2 次提交
-
-
由 Jarkko Nikula 提交于
TLV320AIC3X volume controls are logarithmic. Export their dB ranges. Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Jarkko Nikula 提交于
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: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 09 2月, 2009 5 次提交
-
-
由 Philipp Zabel 提交于
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: NPhilipp Zabel <philipp.zabel@gmail.com> Acked-by: NEric Miao <eric.miao@marvell.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Philipp Zabel 提交于
The spitz has a WM8750 codec connected as I2S slave but doesn't use the PXA I2S system clock. Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com> Acked-by: NEric Miao <eric.miao@marvell.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Philipp Zabel 提交于
The iPAQ h5000 has an AK4535 codec connected as I2S slave, PXA I2S providing SYSCLK. Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com> Acked-by: NEric Miao <eric.miao@marvell.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Roel Kluin 提交于
With a postfix increment count reaches 10001, not 10000. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Robert Jarzmik 提交于
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: NRobert Jarzmik <robert.jarzmik@free.fr> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 06 2月, 2009 6 次提交
-
-
由 Mark Brown 提交于
-
由 Timur Tabi 提交于
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: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mike Frysinger 提交于
Signed-off-by: NMike Frysinger <vapier.adi@gmail.com> Signed-off-by: NBryan Wu <cooloney@kernel.org> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mike Frysinger 提交于
- 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: NMike Frysinger <vapier.adi@gmail.com> Signed-off-by: NCliff Cai <cliff.cai@analog.com> Signed-off-by: NBryan Wu <cooloney@kernel.org> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Jarkko Nikula 提交于
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: NJarkko Nikula <jarkko.nikula@nokia.com> Cc: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Eero Nurkkala 提交于
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: NEero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 04 2月, 2009 3 次提交
-
-
由 Philipp Zabel 提交于
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: NPhilipp Zabel <philipp.zabel@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Philipp Zabel 提交于
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: NPhilipp Zabel <philipp.zabel@gmail.com> Tested-by: NVasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Vasily Khoruzhick 提交于
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: NVasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 03 2月, 2009 4 次提交
-
-
由 Takashi Iwai 提交于
-
由 Liam Girdwood 提交于
This just updates my email address on some drivers I'd forgotten in a previous patch. Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
-
由 Timur Tabi 提交于
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: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Timur Tabi 提交于
Fix a oversight in the CS4270 codec driver that caused a build break. Signed-off-by: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 02 2月, 2009 2 次提交
-
-
由 Mark Brown 提交于
-
由 Eero Nurkkala 提交于
omap_pcm_trigger is called also in interrupt context so CPU flags must be restored when returning. Signed-off-by: NEero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 31 1月, 2009 2 次提交
-
-
由 Grazvydas Ignotas 提交于
Update pandora board file for recent TWL4030 codec changes. Also move output related snd_soc_dapm_nc_pin() calls to omap3pandora_out_init(), where they belong. Signed-off-by: NGrazvydas Ignotas <notasas@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Timur Tabi 提交于
Spruce up the documentation in the CS4270 codec. Use kerneldoc where appropriate. Fix incorrect comments. Signed-off-by: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 30 1月, 2009 1 次提交
-
-
由 Timur Tabi 提交于
ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself. When a codec driver registers with ASoC, a probe function is called. Most codec drivers call ASoC first, and then register with the I2C bus in the ASoC probe function. However, in order to support multiple codecs on one board, it's easier if the codec driver is probed via the I2C bus first. This is because the call to i2c_add_driver() can result in the I2C probe function being called multiple times - once for each codec. In the current design, the driver registers once with ASoC, and in the ASoC probe function, it calls i2c_add_driver(). The results in the I2C probe function being called multiple times before the driver can register with ASoC again. The new design has the driver call i2c_add_driver() first. In the I2C probe function, the driver registers with ASoC. This allows the ASoC probe function to be called once per I2C device. Also add code to check if the I2C probe function is called more than once, since that is not supported with the current ASoC design. Signed-off-by: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-