- 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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- 10 Aug, 2009 1 commit
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 Aug, 2009 1 commit
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 08 Aug, 2009 3 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Chaithrika U S authored
Add support for audio on DA830 EVM- here McASP1 is interfaced to TLV320AIC3106 codec. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Uwe Kleine-König authored
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: Uwe Kleine-König <u.kleine-koenig@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 Aug, 2009 6 commits
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Janusz Krzysztofik authored
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: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
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Mark Brown authored
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Troy Kisky authored
The dma setup code assumes that the buffer size is a multiple of the period size. Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Troy Kisky authored
dai is a parameter to the functions, so use it instead of looking it up. Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Jarkko Nikula authored
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: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Tony Lindgren <tony@atomide.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 06 Aug, 2009 4 commits
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Daniel Ribeiro authored
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: Daniel Ribeiro <drwyrm@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Janusz Krzysztofik authored
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: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Janusz Krzysztofik authored
This patch adds debugging statement that can help in tracing how the driver is trying to control the codec device. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
The WM8776 is a high performance, stereo audio CODEC with five channel input selector. The WM8776 is ideal for surround sound processing applications for home hi-fi, DVD-RW and other audio visual equipment. This driver implements support for most WM8776 features - currently the ADC automatic level control/limiter functionality is omitted. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 05 Aug, 2009 3 commits
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Daniel Mack authored
Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model and let the I2C functions only call the SoC core function for resume and suspend, which do nothing currently but will do later. Signed-off-by: Daniel Mack <daniel@caiaq.de> Cc: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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John Bonesio authored
The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones@secretlab.ca> Acked-by: Grant Likely <grant.likely@secretlab.ca> 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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- 03 Aug, 2009 4 commits
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Mark Brown authored
This converts all the Wolfson drivers using this format (the only devices that do) except WM8753 to use it. 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
As part of this refactoring the type of the CODEC hw_read operation is changed to match the regular read operation. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
While writes tend to be able to use a fairly bus independant format to do the writes reads are all bus specific. To allow us to factor out this code include the bus type as a parameter when setting up the cache. Initially just use this to factor out hw_write_t for I2C. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 31 Jul, 2009 5 commits
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Janusz Krzysztofik authored
This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Depends on: 1) latest version of the CX20442 codec driver that exposes v253_ops structure[1], 2) patch 2/3 form this series: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver[2]. CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience. Board specific audio function control (with related DAPM widgets) has been modeled after empirically discovered codec capabilities. Unlike other ASoC machine drivers, this one makes use of a codec provided line discipline that is required for talking to a modem chip that can control the codec behavoiur. As the line discipline operations must call board specific bits as well, the machine driver registers its own line discipline ops, not the codec provided, and then calls those codec provided from inside its own callbacks. If some kind of a glue, like a bus over a tty, exsited that could help in runtime detection of a modem (bus adapter) over a more generic line discipline (bus driver)[3], the line discipline code could be probably designed in a more generic way. In order to work at all, this driver requires a working McBSP1. On OMAP1510 based machines (not sure if other OMAP1 variants as well), where McBSP1 is a DSP public peripheral, that means the kernel must provide basic DSP support, ie. omap_dsp_init(), in order to power up the DSP. This used to be included in linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c. Without that, the driver would not work, ie. not shift in/out any bits over the CPU DAI[4]. This limitation is not board, but CPU specific, and may apply to other code that makes use of McBSP1/McBSP3 on affected machines. I provide an extra patch (4/3) as a temporary solution. To work correctly in playback mode, this driver requires my prevoiusly submitted patch that corrects pcm pointer calculation for OMAP1510 based machines[5] (already included in linux-2.6.31-rc3). To support codec controls, this driver requires my previously submitted patch that adds support for modem found on Amstrad Delta[6]. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.html [2] http://www.spinics.net/lists/linux-serial/msg01862.html [3] http://www.spinics.net/lists/linux-serial/msg01856.html [4] http://www.spinics.net/lists/linux-omap/msg15114.html [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html [6] http://www.spinics.net/lists/linux-omap/msg15432.html Credits to: Mark Underwood - for his initial, omap-alsa based sound driver for this machine, Mark Brown - for his help, patience and excellent subsytem maintainer support. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Janusz Krzysztofik authored
This corrected patch adds machine independent line discipline code, prevoiusly exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442 codec driver. The code can be used as a standalone line discipline, or as a set of codec specific functions called from machine's line discipline callbacks. Anyway, the line discipline itself must be registered by a machine driver. Applies on top of the followup to my initial driver version: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Janusz Krzysztofik authored
This patch adds new line discipline name an number to include/linux/tty.h. The line discipline will be used by the Amstrad E3 (Delta) sound driver that will come next in this series of patches. Created against linux-2.6.31-rc3. Applies to linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Alan Cox <alan@lxorguk.ukuu.org.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Lars-Peter Clausen authored
The irq can fire as soon as it has been requested, thus all fields accessed from within the irq handler must be initialized prior to requesting the irq. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
This helps CODECs with sparse register maps work better with the register cache display interface. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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