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* Merge remote-tracking branch 'asoc/fix/cs4271' into tmpMark Brown2013-01-10
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| * ASoC: cs4271: fix sparse warningDaniel Mack2012-12-01
| | | | | | | | | | | | | | | | Make the flag in the pdata of type bool to fix a sparse warning. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge remote-tracking branch 'asoc/fix/core' into tmpMark Brown2013-01-10
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| * | ASoC: core: Fix SOC_DOUBLE_RANGE() macrosMark Brown2012-12-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* | | UAPI: Remove empty Kbuild filesDavid Howells2013-01-02
| | | | | | | | | | | | | | | | | | | | | | | | Empty files can get deleted by the patch program, so remove empty Kbuild files and their links from the parent Kbuilds. Signed-off-by: David Howells <dhowells@redhat.com> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | | Merge tag 'asoc-3.8p1' of ↵Takashi Iwai2012-12-17
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: More updates for v3.8 Nothing terribly exciting here, just small localised changes. As well as fixes there are a couple of Cirrus changes and one devm_ change which were in prior to the merge window but got missed from the original pull to Takashi.
| * | Merge remote-tracking branch 'asoc/topic/core' into asoc-nextMark Brown2012-12-15
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| | * | ASoC: Prevent pop_wait overwriteMisael Lopez Cruz2012-12-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-nextMark Brown2012-12-09
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| * \ \ \ Merge remote-tracking branch 'asoc/topic/fsi' into asoc-nextMark Brown2012-12-09
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| * \ \ \ \ Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-nextMark Brown2012-12-09
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* | | | | Merge tag 'asoc-3.8' of ↵Takashi Iwai2012-12-03
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.8 Very quiet release for ASoC really: - Standardisation of the logging. - DT and dmaengine support for Atmel. - Support for Wolfson ADSP cores. - New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
| * \ \ \ \ Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-nextMark Brown2012-12-01
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| | * | | | ASoC: tlv320aic32x4: Add rstn gpio to platform data.Javier Martin2012-11-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add the possibility to specify a gpio through platform data so that a HW reset can be issued to the codec. Signed-off-by: Javier Martin <javier.martin@vista-silicon.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | | Merge remote-tracking branch 'asoc/topic/fsi' into asoc-nextMark Brown2012-12-01
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| | * | | | ASoC: fsi: master clock selection become independent from platform flagsKuninori Morimoto2012-11-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current FSI driver is using platform information pointer, but it is not good design for DT support. This patch makes master clock selection independent from platform information pointer. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: fsi: add master clock control functionsKuninori Morimoto2012-11-06
| | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current FSI driver required set_rate() platform callback function to set audio clock if it was master mode, because it seemed that CPG/FSI-DIV clocks calculation depend on platform/board/cpu. But it was calculable regardless of platform. This patch supports audio clock calculation method, but the sampling rate under 32kHz is not supported at this point. Old type set_rate() is still supported now, but it will be deleted on next version Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-nextMark Brown2012-12-01
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| | * | ASoC: cs4271: add support for AMUTEB=BMUTEC featureDaniel Mack2012-10-14
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The CS4271 has a feature to sync its analog mute flags, so one mute circuitry can be used for both channels. Give users access to this feature with a new DT property and a flag in the platform data. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ALSA: emu10k1: don't update firmware during suspend/resumeTakashi Iwai2012-11-22
| | | | | | | | | | | | | | | | | | | | | Add a flag to suppress the update in emu1010_firmware_thread() during suspend/resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: emu10k1: cache emu1010 firmwareTakashi Iwai2012-11-22
| | | | | | | | | | | | | | | | | | | | | Instead of calling request_firmware() at each time, keep the obtained firmware internally and reuse it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: vx: hard dependency on the standard fw loaderTakashi Iwai2012-11-22
| | | | | | | | | | | | | | | | | | | | | Yet again like previous two commits, drop the old hwdep user-space firmware code from vx driver (snd-vxpocket and snd-vx222). Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'for-linus' into for-nextTakashi Iwai2012-10-30
|\| | | | | | | | | | | ... for migrating the core changes for USB-audio disconnection fixes
| * | ALSA: Add a reference counter to card instanceTakashi Iwai2012-10-30
| |/ | | | | | | | | | | | | | | | | | | | | | | | | For more strict protection for wild disconnections, a refcount is introduced to the card instance, and let it up/down when an object is referred via snd_lookup_*() in the open ops. The free-after-last-close check is also changed to check this refcount instead of the empty list, too. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: core: add hooks for audio timestampsPierre-Louis Bossart2012-10-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ALSA did not provide any direct means to infer the audio time for A/V sync and system/audio time correlations (eg. PulseAudio). Applications had to track the number of samples read/written and add/subtract the number of samples queued in the ring buffer. This accounting led to small errors, typically several samples, due to the two-step process. Computing the audio time in the kernel is more direct, as all the information is available in the same routines. Also add new .audio_wallclock routine to enable fine-grain synchronization between monotonic system time and audio hardware time. Using the wallclock, if supported in hardware, allows for a much better sub-microsecond precision and a common drift tracking for all devices sharing the same wall clock (master clock). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: core: keep track of boundary wrap-aroundPierre-Louis Bossart2012-10-23
| | | | | | | | | | | | | | | | | | | | Keep track of boundary crossing when hw_ptr exceeds boundary limit and wraps-around. This will help keep track of total number of frames played/received at the kernel level Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | UAPI: (Scripted) Disintegrate include/soundDavid Howells2012-10-09
|/ | | | | | | | | Signed-off-by: David Howells <dhowells@redhat.com> Acked-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Thomas Gleixner <tglx@linutronix.de> Acked-by: Michael Kerrisk <mtk.manpages@gmail.com> Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com> Acked-by: Dave Jones <davej@redhat.com>
* Merge tag 'sound-3.7' of ↵Linus Torvalds2012-10-08
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This contains pretty many small commits covering fairly large range of files in sound/ directory. Partly because of additional API support and partly because of constantly developed ASoC and ARM stuff. Some highlights: - Introduced the helper function and documentation for exposing the channel map via control API, as discussed in Plumbers; most of PCI drivers are covered, will follow more drivers later - Most of drivers have been replaced with the new PM callbacks (if the bus is supported) - HD-audio controller got the support of runtime PM and the support of D3 clock-stop. Also changing the power_save option in sysfs kicks off immediately to enable / disable the power-save mode. - Another significant code change in HD-audio is the rewrite of firmware loading code. Other than that, most of changes in HD-audio are continued cleanups and standardization for the generic auto parser and bug fixes (HBR, device-specific fixups), in addition to the support of channel-map API. - Addition of ASoC bindings for the compressed API, used by the mid-x86 drivers. - Lots of cleanups and API refreshes for ASoC codec drivers and DaVinci. - Conversion of OMAP to dmaengine. - New machine driver for Wolfson Microelectronics Bells. - New CODEC driver for Wolfson Microelectronics WM0010. - Enhancements to the ux500 and wm2000 drivers - A new driver for DA9055 and the support for regulator bypass mode." Fix up various arm soc header file reorg conflicts. * tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits) ALSA: hda - Add new codec ALC283 ALC290 support ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls ALSA: hda - fix indices on boost volume on Conexant ALSA: aloop - add locking to timer access ALSA: hda - Fix hang caused by race during suspend. sound: Remove unnecessary semicolon ALSA: hda/realtek - Fix detection of ALC271X codec ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310 ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event ALSA: hda - make a generic unsol event handler ASoC: codecs: Add DA9055 codec driver ASoC: eukrea-tlv320: Convert it to platform driver ALSA: ASoC: add DT bindings for CS4271 ASoC: wm_hubs: Ensure volume updates are handled during class W startup ASoC: wm5110: Adding missing volume update bits ASoC: wm5110: Add OUT3R support ASoC: wm5110: Add AEC loopback support ASoC: wm5110: Rename EPOUT to HPOUT3 ASoC: arizona: Add more clock rates ASoC: arizona: Add more DSP options for mixer input muxes ...
| * Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-10-06
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Additional updates for v3.7 A couple more updates for 3.7, enhancements to the ux500 and wm2000 drivers, a new driver for DA9055 and the support for regulator bypass mode. With the exception of the DA9055 this has all had a chance to soak in -next (the driver was added on Friday so should be in -next today).
| | * ASoC: codecs: Add DA9055 codec driverAshish Chavan2012-09-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: dapm: Allow regulators to bypass as well as disable when idleMark Brown2012-09-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allow regulators managed via DAPM to make use of the bypass support that has recently been added to the regulator API by setting a flag SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will be put into bypass mode before being disabled, allowing the regulator to fall into bypass mode if it can't be disabled due to other users. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ALSA: Make snd_sgbuf_get_{ptr|addr}() available for non-SG casesTakashi Iwai2012-09-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. This is a preliminary work for the HD-audio DSP loader code. Signed-off-by: Ian Minett <ian_minett@creativelabs.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-09-22
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.7 Lots and lots of driver specific cleanups and enhancements but the only substantial framework feature this time round is the compressed API binding: - Addition of ASoC bindings for the compressed API, used by the mid-x86 drivers. - Lots of cleanups and API refreshes for CODEC drivers and DaVinci. - Conversion of OMAP to dmaengine. - New machine driver for Wolfson Microelectronics Bells. - New CODEC driver for Wolfson Microelectronics WM0010.
| | * ASoC: wm8960: remove 'dres' field from platform data structureTimur Tabi2012-09-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The 'dres' field (discharge resistance for headphone outputs) is no longer used in the driver, so remove it. It was used in the original version of the driver when entering standby from off, but we stopped using it when we switched from having a single startup sequence to having separate cap and capless sequences. Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: wm8960: Support shared LRCLKMark Brown2012-09-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | If the LRCLK is shared and the WM8960 is clock master then we should enable the LRCM bit to tell the device that it should drive LRCLK when either ADC or DAC is enabled rather than separately driving the two LRCLKs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Avoid recalculating the bitmask for SOC_ENUM controlsLars-Peter Clausen2012-09-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For ENUM controls the bitmask is calculated based on the number of items. Currently this is done each time the control is accessed. And while the performance impact of this should be negligible we can easily do better. The roundup_pow_of_two macro performs the same calculation which is currently done manually, but it is also possible to use this macro with compile time constants and so it can be used to initialize static data. So we can use it to initialize the mask field of a ENUM control during its declaration. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: dapm: Add flags to regulator suppliesMark Brown2012-09-07
| | | | | | | | | | | | | | | | | | | | | This will be used to enable additional control of the regulators. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
| | * ASoC: dapm: Ensure bypass paths are suspended and resumedMark Brown2012-09-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since bypass paths aren't part of DAPM streams and we may not have any DAPM streams there may not be anything that triggers a DAPM sync for them. Mark all input and output widgets as dirty and then sync to do so at the end of suspend and resume. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
| | * ASoC: Remove unused 'saved_value' field from snd_soc_dapm_widget structLars-Peter Clausen2012-09-05
| | | | | | | | | | | | | | | | | | | | | | | | The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove current PGA control handling"). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: tegra: move platform data headerStephen Warren2012-09-05
| | | | | | | | | | | | | | | | | | | | | | | | Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Merge branch 'asoc-omap' into for-3.7Mark Brown2012-09-05
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| | * | ASoC: wm0010: Add initial wm0010 DSP driverDimitris Papastamos2012-08-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM0010 is a compact digital signal processor that has been highly optimised for low-power audio applications. Extensive memory resources and core optimisation allow the device to manage all audio processing algorithms efficiently and autonomously, while the host processor sleeps or performs other tasks. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC: wm_hubs: Allow configuration of MICBIAS power up delay via pdataMark Brown2012-08-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sometimes the analogue circuitry connected to the microphone needs some time to settle after power up. Allow systems to configure this delay in the platform data, the driver will then insert the required delay during power up of paths that involve the microphone. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC: add definations for compressed operationsVinod Koul2012-08-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Here we update the asoc structures to add compress stream definations First the struct snd_soc_dai_driver adds a new member to indicate if the dai is compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in the struct snd_soc_dai_link. This is to be used for machine driver to perform any opertaions required for setting up compressed audio streams next is the compressed data operations, they are added using struct snd_compr_ops in the struct snd_soc_platform_driver. Signed-off-by: Namarta Kohli <namartax.kohli@intel.com> Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ALSA: Compress - add codec parameter checksVinod Koul2012-09-17
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: Define more channel map positionsTakashi Iwai2012-09-12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For following the standard, define more channel map positions and shuffle the items a bit: - As both PulseAudio and gstreamer define MONO channel position explicitly, we should follow that, too. The mono streams point to this channel position unless they are explicitly assigned to certain channel positions. - Top-front-* and Top-rear-* positions are added, carried from PulseAudio's definitions. - Move NA and MONO definitions at the top of table right after UNKNOWN, since these are more abstract in comparison with other practical positions. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: Follow channel position definitions to alsa-lib mixerTakashi Iwai2012-09-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is already a set of channel position definitions in alsa-lib mixer.h, and it'd be more practical to keep the same order for the PCM channel map, too. The value is shifted with 1 to keep zero for UNKNOWN. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: ac97: Implement channel map workaround for ALC650Takashi Iwai2012-09-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ALC650 has a channel swap option between surround and CLFE channels, so we need to tweak the channel maps dynamically depending on the register bit. Now struct snd_ac97 can contain chmap pointers for playback and capture. The driver may store these and let ac97 driver changing the channel mapping dynamically. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: PCM: channel mapping API implementationTakashi Iwai2012-09-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch implements the basic data types for the standard channel mapping API handling. - The definitions of the channel positions and the new TLV types are added in sound/asound.h and sound/tlv.h, so that they can be referred from user-space. - Introduced a new helper function snd_pcm_add_chmap_ctls() to create control elements representing the channel maps for each PCM (sub)stream. - Some standard pre-defined channel maps are provided for convenience. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge branch 'fixes' of git://git.alsa-project.org/alsa-kernel into for-nextTakashi Iwai2012-09-05
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