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* | | | Revert "ASoC: tpa6130a2: Support for limiting gain"Peter Ujfalusi2010-05-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578. Since core has now support for limiting the volume on controls this patch is not needed. Furthermore, this patch actually prevents the core to set new volume on the TPA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: core: Support for limiting the volumePeter Ujfalusi2010-05-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | Merge branch 'for-2.6.35' of ↵Takashi Iwai2010-05-06
|\ \ \ \ | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| * | | | ASoC: tpa6130a2: Support for limiting gainPeter Ujfalusi2010-05-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for platform dependent gain limiting on the tpa6130a2 (and tpa6140a2) Headset amplifier. 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: tlv320aic3x: Add platform data and reset gpio handlingJarkko Nikula2010-05-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Handle the reset GPIO within the codec driver in order to follow the startup protocol for the tlv320aic3x codecs. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-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: Add WM9090 amplifier driverMark Brown2010-04-30
|/ / / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM9090 is a high performance low power audio subsystem, including headphone and class D speaker drivers. Note that this driver is a standalone CODEC driver and so is only immediately suitable for use with the WM9090 as a standalone sound card taking line inputs, or with a DAC with no software control. The pending ASoC multi-CODEC support will expand the range of systems that can use the driver, or system-specific adaptations can be made. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: UDA134X: Add UDA1345 CODEC supportVladimir Zapolskiy2010-04-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds support for Philips UDA1345 CODEC. The CODEC has only volume control, de-emphasis, mute, DC filtering and power control features. Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Add indirection for CODEC private dataMark Brown2010-04-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | Merge branch 'for-2.6.34' into for-2.6.35Mark Brown2010-04-05
|\ \ \ \ | | |/ / | |/| | | | | | | | | | Conflicts due to context changes next to the backported DMA data change: include/sound/soc.h
| * | | ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_streamDaniel Mack2010-04-05
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This fixes a memory corruption when ASoC devices are used in full-duplex mode. Specifically for pxa-ssp code, where this pointer is dynamically allocated for each direction and destroyed upon each stream start. All other platforms are fixed blindly, I couldn't even compile-test them. Sorry for any breakage I may have caused. [Note that this is a backported version for 2.6.34. Upstream commit is fd23b7dee] Signed-off-by: Daniel Mack <daniel@caiaq.de> Reported-by: Sven Neumann <s.neumann@raumfeld.com> Reported-by: Michael Hirsch <m.hirsch@raumfeld.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Add a notifier for jack status changesMark Brown2010-03-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some systems provide both mechanical and electrical detection of jack status changes. On such systems power savings can be achieved by only enabling the electrical detection methods when physical insertion has been detected. Begin supporting such systems by providing a notifier for jack status changes which can be used to trigger any reconfiguration. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_streamDaniel Mack2010-03-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This fixes a memory corruption when ASoC devices are used in full-duplex mode. Specifically for pxa-ssp code, where this pointer is dynamically allocated for each direction and destroyed upon each stream start. All other platforms are fixed blindly, I couldn't even compile-test them. Sorry for any breakage I may have caused. Reported-by: Sven Neumann <s.neumann@raumfeld.com> Reported-by: Michael Hirsch <m.hirsch@raumfeld.com> Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge branch 'topic/jack' into for-2.6.35Mark Brown2010-03-19
|\ \ \ | | |/ | |/|
| * | ALSA: Add support for key reporting via the jack interfaceMark Brown2010-03-17
| |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some devices provide support for detection of a small number of buttons on their jacks. One common implementation provides a single button, implemented by shorting the microphone to ground and detected along with microphone presence detection by detecting varying current draws on the microphone bias signal. Provide support for up to three buttons via the jack interface. These default to reporting BTN_n but an API is provided to allow these to be remapped to other keys by the machine driver where it knows what the keys are. More keys can be added with ease if required. This is only intended to support simple accessory button designs. If the interface is limiting then either creating a child device for the accessory or accessing the input device in the jack directly is recommended. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Support GPIO based microphone detection for WM8904Mark Brown2010-03-16
| | | | | | | | | | | | | | | | | | | | The WM8904 allows microphone detection signals to be brought out as alternate functions of the GPIO signals which can be detected using interrupt inputs on the CPU. Allow this to be configured using platform data. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Allow configuration of WM8904 GPIO pin functionsMark Brown2010-03-16
| | | | | | | | | | | | | | | | | | Provide platform data allowing the configuration of the GPIO pins on the WM8904 to be selected, allowing alternate functions to be enabled. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Implement interrupt driven microphone detection for WM8903Mark Brown2010-03-16
| | | | | | | | | | | | | | | | Support use of the WM8903 IRQ for reporting of microphone presence and short detection. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Add WM8903 interrupt supportMark Brown2010-03-16
| | | | | | | | | | | | | | Currently used to detect completion of the write sequencer. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Initial WM8903 microphone bias and short detectionMark Brown2010-03-16
| | | | | | | | | | | | | | | | | | Provide support for WM8903 microphone presence and short detection using the GPIOs to route out a logic signal suitable for handling using snd_soc_jack_add_gpios() on the processor GPIOs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Add GPIO configuration support for WM8903Mark Brown2010-03-16
| | | | | | | | | | | | | | | | | | Allow users to pass in a default configuration for the GPIOs of the WM8903 as platform data. This allows configuration of the pin muxing of the device. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Allow pins to be force enabledMark Brown2010-03-16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Allow pins to be forced on regardless of their power state. This is intended for use with microphone bias supplies which need to be enabled in order to support microphone detection - in systems without appropriate hardware leaving the microphone unbiased when not in use saves power. The force done at power check time in order to avoid disrupting other power detection logic. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Remove unused 'muted' flag from DAPM widgetsMark Brown2010-03-16
| | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: tlv320dac33: Add option for keeping the BCLK runningPeter Ujfalusi2010-03-12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Platform data option for the codec to keep the BCLK clock continuously running in FIFO modes (codec master). OMAP3 McBSP when in slave mode needs continuous BCLK running on the serial bus in order to operate correctly. Since in FIFO mode the DAC33 can also shut down the BCLK clock and enable it only when it is needed, let the platforms decide if the CPU side needs the BCLK running or not. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge commit 'v2.6.34-rc1' into for-2.6.35Mark Brown2010-03-10
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| * Merge branch 'topic/misc' into for-linusTakashi Iwai2010-03-08
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| | * ALSA: timer - pass real event in snd_timer_notify1() to instance callbackJaroslav Kysela2010-03-03
| | | | | | | | | | | | | | | | | | | | | Do not use hardcoded SNDRV_TIMER_EVENT_START value. Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-03-01
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| * | Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai2010-01-26
| |\ \
| | * | ALSA: pcm_lib - return back hw_ptr_interruptJaroslav Kysela2010-01-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * | | Merge branch 'topic/noncached-mmap' into topic/miscTakashi Iwai2010-01-21
| |\ \ \
| | * | | ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffersTakashi Iwai2010-01-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai2010-01-21
| |\ \ \ \ | | |/ / / | |/| / / | | |/ /
| | * | ALSA: pcm_core: Fix wake_up() optimizationJaroslav Kysela2010-01-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * | | ALSA: Add snd_pci_quirk_lookup_id()Takashi Iwai2010-01-14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Added a new function to look up a quirk entry with the given PCI SSID instead of a pci device pointer. This can be used when the searched ID is overridden for debugging or such a purpose. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge commit alsa/devel into topic/miscTakashi Iwai2010-01-12
| |\| | | | | | | | | | | | | | | | | | Conflicts: include/sound/version.h
| | * | ALSA: pcm_lib - optimize wake_up() calls for PCM I/OJaroslav Kysela2010-01-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As noted by pl bossart <bossart.nospam@gmail.com>, the PCM I/O routines (snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls until all samples are not processed. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| | * | ALSA: pcm_lib - cleanup & merge hw_ptr update functionsJaroslav Kysela2010-01-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them. The main change is hw_ptr_interrupt variable removal to simplify code logic. This variable can be computed directly from hw_ptr. Ensure that updated hw_ptr is not lower than previous one (it was possible with old code in some obscure situations when interrupt was delayed or the lowlevel driver returns wrong ring buffer position value). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| | * | ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positionsJaroslav Kysela2010-01-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | In some debug cases, it might be usefull to see previous ring buffer positions to determine position problems from the lowlevel drivers. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| | * | ALSA: Release v1.0.22.1Jaroslav Kysela2009-12-28
| | | | | | | | | | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| | * | ALSA: Release v1.0.22Jaroslav Kysela2009-12-16
| | | | | | | | | | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * | | ALSA: cs46xx - Fix suspend/resume with new DSPTakashi Iwai2009-12-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the basic suspend/resume of snd-cs46xx drivers with new DSP. References: https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Tested-by: Florian Zumbiehl <florz@florz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: jazz16: Add support for Media Vision Jazz16 chipsetKrzysztof Helt2009-12-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | sound: pcm: add vmalloc buffer helper functionsClemens Ladisch2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are now five copies of the code to allocate a PCM buffer using vmalloc(). Add a sixth in the core so that the others can be removed. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: Release v1.0.22Jaroslav Kysela2009-12-17
| |/ / | | | | | | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ASoC: Remove unused pmdown_time flagMark Brown2010-03-05
| | | | | | | | | | | | | | | | | | | | | The flag is no longer used in the code so it just wastes a bit. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | ASoC: Add support for WM8960 capless modeMark Brown2010-03-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8960 headphone outputs can be run in capless mode with OUT3 used to drive a pseudo ground for the headphone drivers. In this mode the mono mixer is not used, the mixer should be turned on in concert with the headphone output drivers and the device bias levels are managed differently. Also tweak the existing bias management to remove the use of active discharge while we're at it since that's often audible. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | ASoC: Move WM8960 platform data into include/soundMark Brown2010-03-03
| | | | | | | | | | | | | | | | | | | | | | | | Avoids machine files having to peer into sound/soc which is a bit rude and icky. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | ASoC: core: Add delay operation to snd_soc_dai_opsPeter Ujfalusi2010-03-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The delay callback can be used by the core to query the delay on the dai caused by FIFO or delay in the platform side. In case if both CPU and CODEC dai has FIFO the delay reported by each will be added to form the full delay on the chain. If none of the dai has FIFO, than the delay will be kept as zero. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Allow mulitple usage count of codec and cpu daiJassi Brar2010-02-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two or more dai_links we need to log the number of active users of the dai. For that, we change semantics of the snd_soc_dai.active flag from indicator to reference counter. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Remove runtime field from DAIjassi brar2010-02-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | In order for having snd_soc_dais shared among two or more dai_links, remove the relatively global runtime field from the struct snd_soc_dai Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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