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* Merge branch 'for-linus' of ↵Linus Torvalds2010-08-151-8/+15
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: sound/usb/format: silence uninitialized variable warnings MAINTAINERS: Add Ian Lartey as comaintaner for Wolfson devices MAINTAINERS: Make Wolfson entry also cover CODEC drivers ASoC: Only tweak WM8994 chip configuration on devices up to rev D ASoC: Optimise DSP performance for WM8994 ALSA: hda - Fix dynamic ADC change working again ALSA: hda - Restrict PCM parameters per ELD information over HDMI sound: oss: sh_dac_audio.c removed duplicated #include
| * Merge branch 'fix/asoc' into for-linusTakashi Iwai2010-08-151-8/+15
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| | * ASoC: Only tweak WM8994 chip configuration on devices up to rev DMark Brown2010-08-131-7/+13
| | | | | | | | | | | | | | | | | | | | | | | | Any subsequent revisions will have these configuration changes applied by default. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Optimise DSP performance for WM8994Mark Brown2010-08-131-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | Change the chip defaults to optimise performance of some of the DSP functionality. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | Merge branch 'for-linus' of ↵Linus Torvalds2010-08-127-9/+38
|\ \ \ | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ASoC: add AD1980 obsolete information ASoC: register cache should be 1 byte aligned for 1 byte long register ALSA: hda - Adding support for new IDT 92HD87XX codecs ASoC: Fix inverted mute controls for WM8580 ALSA: HDA: Use model=auto for LG R510 ALSA: hda - Update model entries in HD-Audio-Models.txt ALSA: hda: document VIA models ALSA: hda - patch_nvhdmi.c: Add missing codec IDs, unify names ALSA: hda - add support for Conexant CX20584 ALSA: hda - New snd-hda-intel model/pin config for hp dv7-4000 ALSA: hda - Fix missing stream for second ADC on Realtek ALC260 HDA codec ALSA: hda - Make converter setups sticky ALSA: hda - Add support for Acer ZGA ALC271 (1025:047c) sound/oss: Adjust confusing if indentation sound: oss: au1550_ac97.c removed duplicated #include ASoC: Fix for changed Eureka Kconfig symbol names
| * | Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-08-117-9/+38
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| | * ASoC: add AD1980 obsolete informationSonic Zhang2010-08-104-3/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This codec has been obsoleted by ADI, so add appropriate warnings to the source tree to dissuade people from using in new designs based on driver support. Signed-off-by: Sonic Zhang <sonic.zhang@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: register cache should be 1 byte aligned for 1 byte long registerCliff Cai2010-08-101-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * ASoC: Fix inverted mute controls for WM8580Mark Brown2010-08-101-3/+3
| | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
| | * ASoC: Fix for changed Eureka Kconfig symbol namesEric Bénard2010-08-051-1/+3
| | | | | | | | | | | | | | | Signed-off-by: Eric Bénard <eric@eukrea.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge branch 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6Linus Torvalds2010-08-125-10/+10
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | * 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: mmc_spi: Fix unterminated of_match_table of/sparc: fix build regression from of_device changes of/device: Replace struct of_device with struct platform_device
| * | | of/device: Replace struct of_device with struct platform_deviceGrant Likely2010-08-065-10/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | of_device is just an alias for platform_device, so remove it entirely. Also replace to_of_device() with to_platform_device() and update comment blocks. This patch was initially generated from the following semantic patch, and then edited by hand to pick up the bits that coccinelle didn't catch. @@ @@ -struct of_device +struct platform_device Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Reviewed-by: David S. Miller <davem@davemloft.net>
* | | | Merge remote branch 'gcl/next' into nextBenjamin Herrenschmidt2010-08-091-4/+18
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| * | | | sound/soc: mpc5200_psc_ac97: Use gpio pins for cold resetEric Millbrandt2010-08-061-4/+18
| |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Call the gpio reset platform function instead of using the flawed ac97 functionality of the MPC5200(b) From MPC5200B User's Manual: "Some AC97 devices goes to a test mode, if the Sync line is high during the Res line is low (reset phase). To avoid this behavior the Sync line must be also forced to zero during the reset phase. To do that, the pin muxing should switch to GPIO mode and the GPIO control register should be used to control the output lines." Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* | | | Merge branch 'for-linus' of ↵Linus Torvalds2010-08-07104-626/+8717
|\ \ \ \ | |/ / / |/| / / | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits) ALSA: hda - Add pin-fix for HP dc5750 ALSA: als4000: Fix potentially invalid DMA mode setup ALSA: als4000: enable burst mode ALSA: hda - Fix initial capsrc selection in patch_alc269() ASoC: TWL4030: Capture route runtime DAPM ordering fix ALSA: hda - Add PC-beep whitelist for an Intel board ALSA: hda - More relax for pending period handling ALSA: hda - Define AC_FMT_* constants ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs ALSA: hda - Add support for HDMI HBR passthrough ALSA: hda - Set Stream Type in Stream Format according to AES0 ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF ASoC: wm9081: fix resource reclaim in wm9081_register error path ASoC: wm8978: fix a memory leak if a wm8978_register fail ASoC: wm8974: fix a memory leak if another WM8974 is registered ASoC: wm8961: fix resource reclaim in wm8961_register error path ASoC: wm8955: fix resource reclaim in wm8955_register error path ASoC: wm8940: fix a memory leak if wm8940_register return error ASoC: wm8904: fix resource reclaim in wm8904_register error path ...
| * | Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-08-05104-626/+8717
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| | * ASoC: TWL4030: Capture route runtime DAPM ordering fixPeter Ujfalusi2010-08-041-36/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the ordering problem in DAPM domain, when the user changes between digital and analog sources during active capture (or loopback) scenario. Before this patch, when the user changed from analog source to digital there were a short time, when the codec enabled analog mic bias (2.2 volts) instead of the correct digital mic bias (1.8 volts) to the digital microphones. This behaviour caused by the former implementation of selecting the correct type of bias. This was done at the POST_REG event of the DAPM_MUX_E("TXx Capture Route") widget. By moving the bias type selection as DAPM_SUPPLY and connecting it to the corresponding digimic widget the problematic situation can be avoided. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: wm9081: fix resource reclaim in wm9081_register error pathAxel Lin2010-08-031-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | This patch fixes the error path in wm9081_register to properly free resources. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8978: fix a memory leak if a wm8978_register failAxel Lin2010-08-031-3/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is a memory leak found if wm8978_register() fail. This patch moves the buffer allocate and release at the same level to prevent the memory leak. Signed-off-by: Axel Lin <axel.lin@gmail.com> Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8974: fix a memory leak if another WM8974 is registeredAxel Lin2010-08-031-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register() return -EINVAL (if another WM8974 is registered). Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8961: fix resource reclaim in wm8961_register error pathAxel Lin2010-08-031-4/+5
| | | | | | | | | | | | | | | | | | | | | | | | This patch fixes the error path in wm8961_register to properly free resources. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8955: fix resource reclaim in wm8955_register error pathAxel Lin2010-08-031-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | This patch fixes the error path in wm8955_register to properly free resources. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8940: fix a memory leak if wm8940_register return errorAxel Lin2010-08-031-1/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds checking for wm8940_register return value, and does kfree(wm8940) if wm8940_register() fail. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8904: fix resource reclaim in wm8904_register error pathAxel Lin2010-08-031-5/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch includes below fixes: 1. wm8904 need to be kfreed in wm8904_register() error path before return. 2. fix the error path for snd_soc_register_codec() fail and snd_soc_register_dai() fail to properly free resources. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8711: fix a memory leak if another WM8711 is registeredAxel Lin2010-08-031-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is not freed if wm8711_register() return -EINVAL(if another ad1836 is registered). Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8523: fix resource reclaim in wm8523_register error pathAxel Lin2010-08-031-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch includes below fixes: 1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL. 2. If snd_soc_register_codec failed, goto error path to properly free resources. 3. Instead of using mixed in-line and goto style cleanup, use goto style error handling if snd_soc_register_dai failed. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: da7210: fix a memory leak if failed to initialise da7210 audio codecAxel Lin2010-08-031-2/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | da7210 should be kfreed if da7210_init() return error. This patch also fixes the error handing in the case of snd_soc_register_dai() fail by adding snd_soc_unregister_codec() in error path. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: ak4642: fix a memory leak if failed to initialise AK4642Axel Lin2010-08-031-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | ak4642 should be kfreed if ak4642_init() return error. Signed-off-by: Axel Lin <axel.lin@gmail.com> Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: ad1836: fix a memory leak if another ad1836 is registeredAxel Lin2010-08-031-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register() return -EINVAL (if another ad1836 is registered). Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Initial WM8741 CODEC driverIan Lartey2010-08-034-0/+799
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8741 is a very high performance stereo DAC designed for audio applications such as professional recording systems, A/V receivers and high specification CD, DVD and home theatre systems. The device supports PCM data input word lengths from 16 to 32-bits and sampling rates up to 192kHz. The WM8741 also supports DSD bit-stream data format, in both direct DSD and PCM-converted DSD modes. TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to allow for all supported sample rate / Master Clock frequency combinations. Fully enable control of supplies. Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Merge branch 'for-2.6.36' of ↵Takashi Iwai2010-08-023-108/+156
| | |\ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| | | * ASoC: omap-mcbsp: Remove period size constraint in THRESHOLD modePeter Ujfalusi2010-08-021-39/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The use of sDMA packet mode in THRESHOLD mode removes the restriction on the period size. With the extended THRESHOLD mode user space can ask for any period size it wishes, and the driver will configure the sDMA and McBSP FIFO accordingly. Replace the hw_rule for the period size with static constraint, which will make sure that the period size will be always even (to avoid prime period size, which could be possible in mono stream) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: omap-mcbsp: Support for sDMA packet modePeter Ujfalusi2010-08-021-6/+56
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Utilize the sDMA controller's packet syncronization mode, when the McBSP FIFO is in use (by extending the THRESHOLD mode). When the sDMA is configured for packet mode, the sDMA frame size does not need to match with the McBSP threshold configuration. Uppon DMA request the sDMA will transfer packet size number of words, and still trigger interrupt on frame boundary. The patch extends the original THRESHOLD mode by doing the following: if (period_words <= max_threshold) Current THRESHOLD mode configuration Otherwise (period_words > max_threshold) McBSP threshold = sDMA packet size sDMA frame size = period size With the extended THRESHOLD mode we can remove the constraint for the maximum period size, since if the period size is bigger than the maximum allowed threshold, than the driver will switch to packet mode, and picks the best (biggest) threshold value, which can divide evenly the period size. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: omap-mcbsp: Code cleanup in omap_mcbsp_dai_hw_paramsPeter Ujfalusi2010-08-021-9/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | To make the code a bit more readable, change the indexed references to the omap_mcbsp_dai_dma_params elements with pointer. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: omap-mcbsp: Restructure the code within omap_mcbsp_dai_hw_paramsPeter Ujfalusi2010-08-021-14/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In preparation for the extended threshold mode (sDMA packet mode support), the code need to be restructured. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: tlv320dac33: Add support for automatic FIFO configurationPeter Ujfalusi2010-07-291-26/+64
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Platform parameter to enable automatic FIFO configuration when the codec is in Mode1 or Mode7 FIFO mode. When this mode is selected, the controls for changing nSample (in Mode1), and UTHR (in Mode7) are not added. The driver configures the FIFO configuration based on the stream's period size in a way, that every burst will read period size of data from the host. In Mode7 we need to use a formula, which gives close enough aproximation for the burst length from the host point of view. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: tlv320dac33: Revisit the FIFO Mode1 handlingPeter Ujfalusi2010-07-291-37/+34
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Replace the hardwired latency definition with platform data parameter, and simplify the nSample parameter calculation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: TWL4030: Capture route DAPM event fixPeter Ujfalusi2010-07-271-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is no need to handle POST_PMU, POST_PMD event with the Capture Route widget. It is enough to handle POST_REG event, since that will come when the user changes the routing, and we will switch the needed bits in the registers. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: fsi: Add new funtion for SPDIFKuninori Morimoto2010-07-291-5/+55
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC: fsi: remove device id checkKuninori Morimoto2010-07-291-5/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current FSI driver id is not only 0 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC: fsi: remove unnecessary clock processingKuninori Morimoto2010-07-291-5/+0
| | |/ | | | | | | | | | | | | | | | Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: da7210: Add HeadPhone Playback Volume controlKuninori Morimoto2010-07-231-4/+27
| | | | | | | | | | | | | | | | | | | | | | | | | | | HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Merge branch 'for-2.6.36' of ↵Takashi Iwai2010-07-213-41/+63
| | |\ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| | | * ASoC: TWL4030: Add configurable delay after digimic enablePeter Ujfalusi2010-07-212-4/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: TWL6040: Correct widget handling for driversJorge Eduardo Candelaria2010-07-171-28/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to reduce pop-noise at powering up/down of the DACs and Drivers, these components have to be handled in a specific sequence. Headset, Handsfree, and Earphone drivers are now registered as PGA components to ensure DACs are enabled first. Also, add a delay to leave time for DACs to settle before continuing power up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: TWL4030: DAC power optimizationPeter Ujfalusi2010-07-131-6/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: TWL4030: Fix for digital loopback gain rangePeter Ujfalusi2010-07-131-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: Unconditionally enable WM8994 AIF1ADC TDM modeMark Brown2010-07-201-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | AIF1ADC TDM mode has no effect other than causing the ADCDAT line to be tristated rather than driven low on clock cycles where there is no data to be transmitted. If the clock cycle is idle then there should be no devices using the data so tristating should have no adverse effects. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: davinci: let platform data define edma queue numbersSekhar Nori2010-07-204-5/+19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently the EDMA queue to be used by for servicing ASP through internal RAM is fixed to EDMAQ_0 and that to service internal RAM from external RAM is fixed to EDMAQ_1. This may not be the desirable configuration on all platforms. For example, on DM365, queue 0 has large fifo size and is more suitable for video transfers. Having audio and video transfers on the same queue may lead to starvation on audio side. platform data as defined currently passes a queue number to the driver but that remains unused inside the driver. Fix this by defining one queue each for ASP and RAM transfers in the platform data and using it inside the driver. Since EDMAQ_0 maps to 0, thats the queue that will be used if the asp queue number is not initialized. None of the platforms currently utilize ping-pong transfers through internal RAM so that functionality remains unchanged too. This patch has been tested on DM644x and OMAP-L138 EVMs. Signed-off-by: Sekhar Nori <nsekhar@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC:Support Samsung SoC(S5P) in I2Sv2Chanwoo Choi2010-07-201-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch modify I2Sv2 driver to support Samsung SoC(S5PV210). Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
generated by cgit v1.1 at 2024-12-26 22:36:21 +0000