- 19 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 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mike Arthur 提交于
The WM8711 or WM8711L (WM8711/L) is a low power stereo DAC with an integrated headphone driver. The WM8711/L is designed specifically for portable MP3 audio and speech players. The WM8711/L is also ideal for MD, CD machines and DAT players. Signed-off-by: NMike Arthur <Mike.Arthur@wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 8月, 2009 2 次提交
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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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 1 次提交
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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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- 15 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 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 1 次提交
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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 1 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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- 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 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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- 06 8月, 2009 5 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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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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- 03 8月, 2009 3 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
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 1 次提交
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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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- 30 7月, 2009 1 次提交
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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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- 29 7月, 2009 1 次提交
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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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- 23 7月, 2009 3 次提交
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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 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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- 22 7月, 2009 2 次提交
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由 Joonyoung Shim 提交于
The callback function to control register was used by whole controls in MAX9877 driver, but this causes using many if statement for double register control or invert. So, the callback function for double register control is separate differently, and the code for invert is added in the callback function. Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 javier Martin 提交于
This corrects a bug with ADC Inversion Switch in wm8974 codec. Signed-off-by: NJavier Martin <javier.martin@vista-silicon.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 7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 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL configuration if the output frequency is zero. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 7月, 2009 1 次提交
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由 Takashi Iwai 提交于
Replaced with dev_{get|set}_drvdata(). Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 16 7月, 2009 1 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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- 15 7月, 2009 1 次提交
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由 Joonyoung Shim 提交于
The MAX9877 combines a high-efficiency Class D audio power amplifier with a stereo Class AB capacitor-less DirectDrive headphone amplifier. The max9877_add_controls() is called to register the MAX9877 specific controls on machine specific init() of the machine driver. The datasheet for the MAX9877 can find at the following url: http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf [Slight edit to sort the ALL_CODECS entries -- broonie.] Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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