- 10 8月, 2009 1 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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- 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 6 次提交
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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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由 Janusz Krzysztofik 提交于
This patch adds debugging statement that can help in tracing how the driver is trying to control the codec device. 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
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Daniel Mack 提交于
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: NDaniel Mack <daniel@caiaq.de> Cc: Timur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 John Bonesio 提交于
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: NJohn Bonesio <bones@secretlab.ca> Acked-by: NGrant Likely <grant.likely@secretlab.ca> 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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- 03 8月, 2009 4 次提交
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由 Mark Brown 提交于
This converts all the Wolfson drivers using this format (the only devices that do) except WM8753 to use it. 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
As part of this refactoring the type of the CODEC hw_read operation is changed to match the regular read operation. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 01 8月, 2009 5 次提交
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由 Janusz Krzysztofik 提交于
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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
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: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: NAlan Cox <alan@lxorguk.ukuu.org.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
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: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
This helps CODECs with sparse register maps work better with the register cache display interface. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 30 7月, 2009 4 次提交
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由 Mark Brown 提交于
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由 Barry Song 提交于
1. fix "line over 80 characters" checkpatch warnings 2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead 3. fix typos Signed-off-by: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
With the s3c platform has implementing gpiolib support the s3c_gpio api has been deprecated. This patch gets rid of all s3c_gpio calls and replaces them by using gpiolib. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 29 7月, 2009 3 次提交
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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 Barry Song 提交于
The I2S DAI driver for blackfin SPORT, but works in TDM mode. I2S is not a special case of TDM with only left and right two slots for SPORT interface. I2S coordinates with TDM in SPORT, but not a part of TDM. TDM require different hardware configuration with I2S, not only different slot number. One is "Stereo Serial Operation" mode of SPORT, the other one is "Multichannel Operation" mode. They are incompatible at the same time. Hardware and DMA description and data transfer flow are much different for I2S and TDM. Merging them as a whole will be very ugly and difficult to maintain. So we don't define a new DAI type, but give two DAI instances for standard I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the I2S-family DAI type. Signed-off-by: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
The patch fixes some checkpatch identified issues and adds a comment about line discipline interaction to my driver code, as requested by Mark on my inital submission (thank you Mark for applying my imperfect patch anyway). It also fixes MODULE_ALIAS mismatch as used in my machine driver. Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 7月, 2009 2 次提交
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由 Marek Vasut 提交于
This patch removes the old method of jack detection from palm27x-asoc driver and adds jack detection api. It also removes some other (now) useless stuff from the driver and corrects pin configuration for the codec. Signed-off-by: NMarek Vasut <marek.vasut@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
The patch adds a few small enhancements to the ASoC jack handling, as suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes for related bugs found while learning Mark's code and testing results. Enhancements: 1. Update status of an ASoC jack while associating it with new gpios. 2. Really update DAPM pins while associating them with an ASoC jack. 3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes. Fixes: 1. Apply mask on jack status report before using it, just for case. 2. While updating jack associated DAPM pins, use full resulting jack status, not the status report passed as an argument. Created and tested on linux-2.6.31-rc3 Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 23 7月, 2009 6 次提交
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由 Marek Vasut 提交于
This patch allows passing platform_data to devices attached to AC97 bus (like touchscreens, battery measurement chips ...). Signed-off-by: NMarek Vasut <marek.vasut@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Joonyoung Shim 提交于
The MAX9877 needs an address of start register when we write values to registers through i2c_master_send(), but the code for this was missed in max9877_write_regs(). If the value of control is 0 in the max9877_set_out_mode(), the value is not increased to 1, but actually the value to write to the register should be 1. And the register bits for out_mode and osc_mode should be cleared before writing. Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
This patch adds support for Conexant CX20442-11 voice modem codec, suitable for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related sound card driver will follow. This codec is an optional part of the Conexant SmartV three chip modem design. As such, documentation for its proprietary digital audio interface is not available. However, on Amstrad Delta board, thanks to Mark Underwood who created an initial, omap-alsa based sound driver a few years ago[1], the codec has been discovered to be accessible not only from the modem side, but also over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound card that can access the codec DAI directly. The DAI configuration parameters (sample rate and format, number of channels) has been selected out empirically for best user experience. The codec analogue interface consists of two pairs of analogue I/O pins: speakerphone interface or telephone handset/headset interface. Furthermore, it seams to provide two operation modes for speakerphone I/O: standard and advanced, with automatic gain control and echo cancelation. Even if the codec control interface is unknown and not available, all those interfaces and modes can be selected over the modem chip using V.253 commands. The driver is able to issue necessary commands over a suitable hw_write function if provided by a sound card driver. Otherwise, the codec can be controlled over the modem from userspace while inactive. Even if nothig is known about the codec internal power management capabilities, DAPM widgets has been used to model the codec audio map. Automatically performed powering up/down of those virtual widgets results in corresponding V.253 commands being issued. Some driver features/oddities may be board specific, but I have no way to verify that with any board other than Amstrad Delta. [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. 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 Lopez Cruz, Misael 提交于
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Chaithrika U S 提交于
PLL was not being enabled when it was not bypassed. This patch enables the PLL when it is used. Additionally, it disables the PLL when it is bypassed. Without this patch, the audio on TI DM646x EVM and DM355 EVM does not work properly. The bit clocks and the frame sync signals from the codec are not correct and hence the playback/record are faster than usual for most sample rates. The reason for this was that the PLL was not enabled when it was not bypassed. Tested on DM6467 EVM, playback tested on DM355 EVM. Signed-off-by: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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