- 13 4月, 2009 4 次提交
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由 Mark Brown 提交于
This ensures that we sync with the DAPM powerdown sequencing properly and don't need to bounce the power on the voice DAC so often. 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
This is simple code motion, intended to support future refactoring of the DAPM algorithms and (more immediately) the additon of events for DACs and ADCs. 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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- 12 4月, 2009 1 次提交
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由 Alexander Beregalov 提交于
Signed-off-by: NAlexander Beregalov <a.beregalov@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 4月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
Add DSP_A interface format support by setting the LRP bit in DSP mode. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 08 4月, 2009 4 次提交
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由 Mark Brown 提交于
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由 Mark Brown 提交于
The WM8988 is a low power, high quality stereo CODEC designed for portable digital audio applications. The device integrates complete interfaces to 2 stereo headphone or line out ports. External component requirements are drastically reduced as no separate headphone amplifiers are required. Advanced on-chip digital signal processing performs graphic equaliser, 3-D sound enhancement and automatic level control for the microphone or line input. The WM8988 can operate as a master or a slave, with various master clock frequencies including 12 or 24MHz for USB devices, or standard 256fs rates like 12.288MHz and 24.576MHz. Different audio sample rates such as 96kHz, 48kHz, 44.1kHz are generated directly from the master clock without the need for an external PLL. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Many devices require symmetric configurations of capture and playback data formats, often due to shared clocking but sometimes also due to other shared playback and record configuration in the device. Start providing core support for this by allowing the DAIs or the machine to specify that the sample rates used should be kept symmetric. A flag symmetric_rates is provided in the snd_soc_dai and snd_soc_dai_link structures. If this is set in either of the DAIs or in the machine then a constraint will be applied when a stream is already open preventing any changes in sample rate. 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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- 06 4月, 2009 2 次提交
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由 Daniel Glöckner 提交于
According to the data sheet data is clocked out on the falling edge and latched on the rising edge of the bit clock. While the left sample is transmitted the word clock line is low. Signed-off-by: NDaniel Glöckner <dg@emlix.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Dan Carpenter 提交于
ak4535_remove() from sound/soc/codecs/ak4535.c calls i2c_unregister_device() with a possibly null pointer. This bug was found by smatch (http://repo.or.cz/w/smatch.git/). Signed-off-by: NDan Carpenter <error27@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 04 4月, 2009 2 次提交
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由 Daniel Glöckner 提交于
This patch adds machine specific code for the audio part of the Stretch s6105 IP camera reference design. The device uses the tlv320aic31(01) codec to generate the clock for both I2S ports of the soc. While the master clock is generated by a configurable PLL chip, the code assumes the factory default settings. An additional kcontrol has been added to handle the special routing of the board, connecting both HPLCOM and HPROUT to the same pin of the audio jack. One of these should always be switched off. Signed-off-by: NDaniel Glöckner <dg@emlix.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Daniel Glöckner 提交于
This patch adds a driver for the I2S interface found on Stretch s6000 family processors. Signed-off-by: NDaniel Glöckner <dg@emlix.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 03 4月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
Adds the needed code to be able to use 96KHz playback. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 02 4月, 2009 12 次提交
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由 Mark Brown 提交于
Without this the WM9705 driver fails badly when resuming. Tested-by: NRussell King <linux@arm.linux.org.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Ensure that any AC97 devices that bind to the CODEC are below the ASoC device in the device tree so the suspend and resume code can figure out what order to handle them in. Reported-by: NRussell King <linux@arm.linux.org.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
AC97 devices may have other drivers hanging off them directly so need to have resumed when the resume function returns meaning that we can't defer the resume - complete it immediately for them. Non-AC97 devices should not have other drivers hanging directly off the ASoC devices. We only really need the deferral for non-AC97 devices - it's there since some I2C buses are very slow and non-AC97 codecs often have large numbers of registers to restore and require delays to bring the codec up cleanly leading to a substantial impact on overall resume time. Reported-by: NRussell King <linux@arm.linux.org.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
A brief overview of how the components of the API fit together. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jarkko Nikula 提交于
McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During initial playback startup, this FIFO is keeping the DMA request active until the FIFO is full. So now if ALSA buffer size is smaller, DMA is looping around it while filling up the HW FIFO, generating burst of interrupts as well and SW doesn't have any change to fill enough data. Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. 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 supports 96KHz sample playback, but only playback. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Timur Tabi 提交于
Optimize the display of SSI statistics in the Freescale MPC8610 sound driver to display the status count only of the interrupts that were actually enabled. Previously, it would display the counts of all SISR status bits, even those that were not enabled. Signed-off-by: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Luotao Fu 提交于
the variable gsr_bit is set in isr. It is however set to 0 and interrupts are disabled prior to reset. Hence it doesn't make a lot of sense to show the content of gsr_bit in case of a reset timeout. Signed-off-by: NLuotao Fu <l.fu@pengutronix.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Timur Tabi 提交于
Remove the delay from the trigger function in the Freescale MPC8610 sound driver when capture is started. This delay was used to ensure that the DMA controller was active when ALSA call the .pointer function to request a DMA transfer status. A better approach is for the .pointer function to detect that DMA has not started, and return zero instead. This change eliminates the need for the delay. Also add some related code to check for a DMA programming error, and report XRUN if it occurs. Signed-off-by: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Philipp Zabel 提交于
HTC Magician has a Philips UDA1380 codec connected via SSP1 (playback) and I2S (capture). There is a flip-flop between the SSP frame clock output and the codec's word select input pin. To make the codec see proper I2S input, the SSP has to send two frames per sample. Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Philipp Zabel 提交于
Now magician and similar boards can use network mode with only one active slot to explicitly set 16 bit frame width, even for S16_LE stereo sound. Signed-off-by: NPhilipp Zabel <philipp.zabel@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 3月, 2009 7 次提交
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由 Takashi Iwai 提交于
Fix the wrong device pointer passed to dev_err(). Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Lopez Cruz, Misael 提交于
Headset was declared previously as a Headphone widget connecting HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver. The capture path becomes invalid as the Headphone widget is not a valid input endpoint. Instead of that, the Headset is declared as separate Microphone and Headphone widgets. Current patch modifies audio map: - Headset Mic: HSMIC with bias - Headset Stereophone: HSOL, HSOR Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jarkko Nikula 提交于
Add functions "Headset" and "Mic" to the control "Jack Function" for activating and de-activating codec input pin LINE1L which is connected to the mic pin of 4-pole Nokia AV connecter. Note there is no mic bias voltage management here since bias is coming from Nokia ASIC and driver for it is not in mainline. 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
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
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
There is an AVDD supply as well, normally one or more of the other upplies would be tied to it. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
The active discharge does not bring sufficient benefit to justify the lengthy times involved so don't do that. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 3月, 2009 1 次提交
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由 Mark Brown 提交于
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- 17 3月, 2009 1 次提交
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由 Atsushi Nemoto 提交于
The commit 14fa43f5 ("ASoC: Only register AC97 bus if it's not done already") added a condition for calling of soc_ac97_dev_register() but not added for calling of soc_ac97_dev_unregister(). This patch adds same condition for soc_ac97_dev_unregister(). Without this fix, kernel crashes when unloading an asoc driver. Signed-off-by: NAtsushi Nemoto <anemo@mba.ocn.ne.jp> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 16 3月, 2009 4 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
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Joonyoung Shim 提交于
CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Robert Jarzmik 提交于
As the PXA27x series allow 2 gpios to reset the ac97 bus, allow through platform data configuration the definition of the correct gpio which will reset the AC97 bus. This comes from a silicon defect on the PXA27x series, where the gpio must be manually controlled in warm reset cases. Signed-off-by: NRobert Jarzmik <rjarzmik@free.fr> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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