- 18 8月, 2009 1 次提交
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 8月, 2009 5 次提交
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由 Mark Brown 提交于
Speaker and headphone outputs do not need to be handled separately since they can't be part of the same path. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Shine Liu 提交于
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: NShine Liu <shinel@foxmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 8月, 2009 8 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
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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 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Conflicts: sound/soc/Makefile
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由 Barry Song 提交于
Signed-off-by: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Change the strings related to capture in order to be interpreted correctly by alsamixer and possible other UI based mixer applications. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 14 8月, 2009 5 次提交
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由 Marek Vasut 提交于
This patch adds support for passing platform data to ac97 bus devices from PXA2xx-AC97 driver.. Signed-off-by: NMarek Vasut <marek.vasut@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Chaithrika U S 提交于
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: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Chaithrika U S 提交于
The patch adds a DAI format: Codec bit clock master and frame sync slave, to the driver. Signed-off-by: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Chaithrika U S 提交于
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: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 8月, 2009 4 次提交
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由 Mark Brown 提交于
- Build in SND_SOC_ALL_CODECS. - Remove null suspend/resume stuff. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Barry Song 提交于
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: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
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: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Barry Song 提交于
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: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 12 8月, 2009 1 次提交
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由 Mark Brown 提交于
It's only actually paying attention to the slot count anyway. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 11 8月, 2009 3 次提交
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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由 Randy Dunlap 提交于
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: NRandy Dunlap <randy.dunlap@oracle.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 8月, 2009 2 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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- 08 8月, 2009 3 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 Chaithrika U S 提交于
Add support for audio on DA830 EVM- here McASP1 is interfaced to TLV320AIC3106 codec. Signed-off-by: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Uwe Kleine-König 提交于
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig): CC [M] sound/soc/s3c24xx/s3c2443-ac97.o sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX' sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read': sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function) sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.) sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write': sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function) Signed-off-by: NUwe Kleine-König <u.kleine-koenig@pengutronix.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 8月, 2009 6 次提交
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由 Janusz Krzysztofik 提交于
The patch changes the line discipline name registered in include/linux/tty.h and updates the ams-delta machine driver to use it. Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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由 Mark Brown 提交于
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由 Troy Kisky 提交于
The dma setup code assumes that the buffer size is a multiple of the period size. Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Troy Kisky 提交于
dai is a parameter to the functions, so use it instead of looking it up. Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jarkko Nikula 提交于
Simultaneous audio playback and capture on OMAP1510 can cause that second stream is stalled if there is enough delay between startup of the audio streams. Current implementation of the omap_mcbsp_start is starting both transmitter and receiver at the same time and it is called only for firstly started audio stream from the OMAP McBSP based ASoC DAI driver. Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is missed if there is no DMA transfer set up at that time when the first word after McBSP startup is transmitted. The problem hasn't noted before since later OMAPs are using level sensitive DMA request lines. Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by allowing to start and stop individually McBSP transmitter and receiver logics. Then call those functions individually for both audio playback and capture streams. This ensures that DMA transfer is setup before transmitter or receiver is started. Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA request line behavior differences between the OMAP generations. Reported-and-tested-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NTony Lindgren <tony@atomide.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 06 8月, 2009 2 次提交
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由 Daniel Ribeiro 提交于
Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: NDaniel Ribeiro <drwyrm@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
This patch is a workaround for the problem of several subsequent control statements not being applied correctly to the codec controller (modem). In order to follow the hook switch state change from handset to handsfree while in full duplex mode, two consecutive +VLS control commands were sent to the modem. The first one was M1 (microphone only), the seconds one was M1S1 (both microphone and speaker). As there was no real modem handshaking procedure implemented, neither in the codec nor in the machine driver part of the line discipline, the modem was having the second command missed. Since a possibility to switch to microphone only mode (and speaker only mode as well) seams of no value, I have modified the code to issue single M1S1 command only for any of those cases. Tested on my Amstrad Delta. Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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