- 20 Aug, 2009 7 commits
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Eduardo Valentin authored
This patch export through sysfs two properties to configure maximum threshold for transmission and reception on each mcbsp instance. Also, it exports two helper functions to allow mcbsp users to read this values. Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Eduardo Valentin authored
This patch adds a way to handle transmit/receive threshold. It export to mcbsp users a callback registration procedure. Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Eduardo Valentin authored
Increasing startup delay value as worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec Although, 100us may give enough time for two CLKSRG, due to some unknown PM related, clock gating etc. reason, this patch increases it to 500us. Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Eduardo Valentin authored
Adding McBSP register definition for IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Eero Nurkkala authored
ASoC has an annoying bug letting either L or R channel to be played on L channel. In other words, L and R channels can switch at random. This provides McBSP funtionality that may be used to fix this feature. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Kuninori Morimoto authored
This driver is very simple. It support playback only now. This patch is tested by ms7724se board. Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Shine Liu authored
> Then it's a driver bug. If unaligned period size is allowed, it means > that the irq is really generated in that period, not at the buffer > boundary. Otherwise, it must have a proper hw-constraint to align the > period size to the buffer size. This patch will fix the bug metioned in the above mail. Force the peroid size to be aligned with the buffer size. Based and tested on linux-2.6.31-rc6. Signed-off-by: Shine Liu <shinel@foxmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 Aug, 2009 2 commits
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Mark Brown authored
Used for applications such as direct bluetooth connections on smartphones which don't go via the CPU. This used to be supported before the refactoring to share code but this check was removed during that move. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
If the CODEC does not provide a set_bias_level() then update the bias_level variable for it since other parts of the system expect that to be maintained. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 Aug, 2009 3 commits
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Mark Brown authored
These need to be in the CODEC since the DAIs supported by the CODECs aren't static. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Joonyoung Shim reports that S3C64xx I2S is working on the NCP boards so allow it to be selected in Kconfig. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmciro.com>
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Joonyoung Shim authored
The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD register. This patch changes to the defined values it. And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 Aug, 2009 7 commits
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Mark Brown authored
Note that the number of slots used internally is specified in terms of stereo slots while the external API works with mono slots. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
When used without the PLL we were accidentally clearing the MCLK/2 divider, resulting in a double rate SYSCLK when the divider should have been used. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Speaker and headphone outputs do not need to be handled separately since they can't be part of the same path. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
If the system doesn't have any DAPM widgets then we can't use their state to check if the bias level for the codec should be up. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Shine Liu authored
There is a mistake in current uda134x_mute function: mute_reg has been changed in line 162 or line 164, so uda134x_write should write "mute_reg" but not "mute_reg & ~(1<<2)" to UDA134X_DATA010. Signed-off-by: Shine Liu <shinel@foxmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Janusz Krzysztofik authored
Enhance period_index accuracy, particularly just before buffer rewind, by making use of DMA interrupt status flags in addition to simply counting up interrupts. Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Janusz Krzysztofik authored
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP models. Remove unnecessary DMA transfer restart from interrupt handler routine. The interrupt routine used to maintain a period index, originally needed for counting up periods up to a full buffer in order to restart the DMA transfer. For some time, this counter is also used as a replacement for hardware DMA progress counter that has been found unusable on OMAP1510 in case of playback. Thus, the period index calculation cannot be omitted completely. However, the accuracy of this counter can still suffer from missing DMA interrupts. In order to work correctly, it requires patch 1 from this series also applied: [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 Aug, 2009 6 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
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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
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Conflicts: sound/soc/Makefile
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- 14 Aug, 2009 2 commits
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Barry Song authored
Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 Aug, 2009 9 commits
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Marek Vasut authored
This patch adds support for passing platform data to ac97 bus devices from PXA2xx-AC97 driver.. Signed-off-by: Marek Vasut <marek.vasut@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Chaithrika U S authored
There is one instance of McASP on DA850/OMAP-L138 SoC. This is connected to TLV320AIC3106 codec for audio playback and capture. This patch adds audio support on this platform. Some of the structure prefix names which are common for DA830/OMAP-L137 EVM and DA850/OMAP-L138 EVM have been renamed to da8xx from da830. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Chaithrika U S authored
The patch adds a DAI format: Codec bit clock master and frame sync slave, to the driver. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Chaithrika U S authored
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO support. This FIFO provides additional data buffering. It also provides tolerance to variation in host/DMA controller response times. The read and write FIFO sizes are 256 bytes each. If FIFO is enabled, the DMA events from McASP are sent to the FIFO which in turn sends DMA requests to the host CPU according to the thresholds programmed. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber= sprufm1&fileType=pdf This patch adds support for FIFO configuration. The platform data has a version field which differentiates the McASP on different SoCs. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
The WM8993 analogue control is shared with other devices in the same product line. Since this is a very substantial proportion of the driver move the definitions of these controls into a new wm_hubs module which allows them to be shared between the two. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
- Build in SND_SOC_ALL_CODECS. - Remove null suspend/resume stuff. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Barry Song authored
There has been an ad1836 driver in sound/blackfin based on traditional alsa. The new driver is based on asoc. The architecture of ad1836 codec driver is very much like ad1938. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Peter Ujfalusi authored
Dynamically control and control only the needed output amplifier muting/un-muting. The original code was muting and un-muting the following output amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time regardless which pin is actually in use at the given moment. Move these as separate PGA so only the needed amplifier will be touched. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Barry Song authored
According to the function dapm_dac_check_power() in sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any output widget as sink. And according to dapm_adc_check_power(), adc power can't be on/off stand-alone without any input widget as source. So we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC to hope their power can be managed dynamically. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 12 Aug, 2009 1 commit
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Mark Brown authored
It's only actually paying attention to the slot count anyway. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 11 Aug, 2009 3 commits
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Mark Brown authored
Store the TDM slot width then if it's set use that rather than the sample size to calculate BCLK. Leave imposing constraints to the core (which should do this but doesn't yet) or machine driver. Also allow 0 TDM slots to be configure (for use when disabling TDM). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
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Randy Dunlap authored
Fix soc build errors when I2C is built as a loadable module: (.text+0x5d26b): undefined reference to `i2c_master_send' soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer' Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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