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| * ASoC: Make pmdown_time a longMark Brown2010-02-171-1/+1
| | | | | | | | | | | | | | Fixes a warning. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: Make pmdown_time a per-card settingMark Brown2010-02-161-0/+2
| | | | | | | | | | | | | | | | | | | | Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: Add WM2000 driverMark Brown2010-02-121-0/+26
| | | | | | | | | | | | | | | | | | | | The WM2000 is a low power, high quality handset receiver speaker driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It provides enhanced voice communication quality in a noisy environment if the handset acoustics are designed appropriately. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: Add a cache_sync bit to the CODEC structureMark Brown2010-02-041-0/+1
| | | | | | | | | | | | | | | | | | | | Add a bit to the CODEC structure indicating if a cache sync is required. By default this will be set if a cache only write is done to a soc-cache register cache. This allows us to avoid syncing the cache back after using cache only writes if there were no changes. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: Allow CODECs to ask soc-cache to suppress physical writesMark Brown2010-02-031-0/+1
| | | | | | | | | | | | | | | | | | | | | | Currently the soc-cache code will always write to the device, meaning that we need the device to be powered and active at pretty much all times the system is active. Allowing cache only writes lays some groundwork for future enhancements to allow devices to be put into a full off state when the audio subsystem is idle. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: add helper macros to declare struct soc_enum instancesGuennadi Liakhovetski2010-01-251-0/+17
| | | | | | | | | | | | | | | | | | Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-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: add simplified versions of widget macrosGuennadi Liakhovetski2010-01-251-0/+32
| | | | | | | | | | | | | | | | | | | | | | Many macros from include/sound/soc-dapm.h take an array and a number of elements in it as arguments, whereas most users use static arrays and use "x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of those macros, calling ARRAY_SIZE() internally. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: Support turning off bias when the CODEC is idleMark Brown2010-01-211-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: tlv320dac33: Burst mode BCLK divider configurationPeter Ujfalusi2010-01-201-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | Add possibility to configure the burst mode BCLK divider through platform data structure. The BCLK divider changes the actual speed of the serial bus in burst mode, which is faster than the sampling frequency of the running stream. In this way platforms can experiment with the optimal burst speed without the need to modify the codec driver itself. 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: fix compile breakage - add a missing header includeGuennadi Liakhovetski2010-01-191-0/+2
| | | | | | | | | | | | Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * Merge branch 'for-2.6.33' into for-2.6.34Mark Brown2010-01-1214-35/+452
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| * | ASoC: tpa6130a2: Support for tpa6140's regulatorsIlkka Koskinen2010-01-111-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | tpa6140a2 uses different names for the regulators. Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: Add initial WM8955 CODEC driverMark Brown2009-12-181-0/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8955 is a low power, high quality stereo DAC with integrated headphone and loudspeaker amplifiers, designed to reduce external component requirements in portable digital audio applications. This is an initial driver implementing support for the majority of the functionality in the device, currently OUT3 is not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * | ASoC: Initial WM8904 CODEC driverMark Brown2009-12-041-0/+57
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8904 is a high performance ultra-low power stereo CODEC optimised for portable audio applications, with features including a class W amplifier, FLL with free running mode, Mobile ReTune and ground referenced headphone and line outputs. Support for some features, most particularly the digital microphone interface, is not yet present. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * | ASoC: Export snd_soc_update_bits_unlocked()Mark Brown2009-12-041-0/+3
| | | | | | | | | | | | | | | | | | | | | Allows custom controls to use it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai2010-01-261-0/+1
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| * | | ALSA: pcm_lib - return back hw_ptr_interruptJaroslav Kysela2010-01-261-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Iwai2010-01-211-0/+4
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| * | | | ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffersTakashi Iwai2010-01-181-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Iwai2010-01-211-2/+3
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| * | | ALSA: pcm_core: Fix wake_up() optimizationJaroslav Kysela2010-01-211-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Iwai2010-01-141-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Iwai2010-01-123-3/+11
|\ \ \ \ | |/ / / | | | | | | | | | | | | Conflicts: include/sound/version.h
| * | | ALSA: pcm_lib - optimize wake_up() calls for PCM I/OJaroslav Kysela2010-01-071-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Kysela2010-01-072-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Kysela2010-01-071-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Kysela2009-12-281-1/+1
| | | | | | | | | | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * | | ALSA: Release v1.0.22Jaroslav Kysela2009-12-161-1/+1
| | |/ | |/| | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* | | ALSA: cs46xx - Fix suspend/resume with new DSPTakashi Iwai2009-12-221-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Helt2009-12-211-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Ladisch2009-12-181-0/+38
| | | | | | | | | | | | | | | | | | | | | | | | 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 Kysela2009-12-171-1/+1
|/ / | | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' of ↵Linus Torvalds2009-12-092-2/+2
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (42 commits) tree-wide: fix misspelling of "definition" in comments reiserfs: fix misspelling of "journaled" doc: Fix a typo in slub.txt. inotify: remove superfluous return code check hdlc: spelling fix in find_pvc() comment doc: fix regulator docs cut-and-pasteism mtd: Fix comment in Kconfig doc: Fix IRQ chip docs tree-wide: fix assorted typos all over the place drivers/ata/libata-sff.c: comment spelling fixes fix typos/grammos in Documentation/edac.txt sysctl: add missing comments fs/debugfs/inode.c: fix comment typos sgivwfb: Make use of ARRAY_SIZE. sky2: fix sky2_link_down copy/paste comment error tree-wide: fix typos "couter" -> "counter" tree-wide: fix typos "offest" -> "offset" fix kerneldoc for set_irq_msi() spidev: fix double "of of" in comment comment typo fix: sybsystem -> subsystem ...
| * | tree-wide: fix assorted typos all over the placeAndré Goddard Rosa2009-12-041-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | That is "success", "unknown", "through", "performance", "[re|un]mapping" , "access", "default", "reasonable", "[con]currently", "temperature" , "channel", "[un]used", "application", "example","hierarchy", "therefore" , "[over|under]flow", "contiguous", "threshold", "enough" and others. Signed-off-by: André Goddard Rosa <andre.goddard@gmail.com> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
| * | comment typo fix: sybsystem -> subsystemJean Delvare2009-12-041-1/+1
| | | | | | | | | | | | | | | Signed-off-by: Jean Delvare <jdelvare@suse.de> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* | | Merge branch 'topic/hda' into for-linusTakashi Iwai2009-12-043-7/+331
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| * \ \ Merge branch 'topic/ice1724-quartet' into topic/hdaTakashi Iwai2009-12-013-7/+331
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| | * | | ALSA: ak4113 supportPavel Hofman2009-09-211-0/+321
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * complete support for ak4113 * based on code for ak4114 and ak4117 Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: ak4620 support, codec regs listed in procPavel Hofman2009-09-211-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * complete support for ak4620 * codec regs listed in proc for all codecs/chips * adding total regs for each codec * fixing nb. of steps in input attenuation controls Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: ak4114 - fix errors in output selector bitsPavel Hofman2009-09-211-6/+6
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * the previous version had a typo - values of AK4114_OPS10-12 were identical with AK4114_OPS00-02 * Since no cards actually use this feature, the bug was not identified earlier Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge branch 'topic/asoc' into for-linusTakashi Iwai2009-12-045-9/+87
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| * | | ASoC: Add BCLK calculation utility for TDM mode tooMark Brown2009-11-251-0/+1
| | | | | | | | | | | | | | | | | | | | Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Add jack_status_check callback function for GPIO jacksJoonyoung Shim2009-11-121-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The jack_status_check callback function is the interface to check the status of the jack. Some target provides the method to distinguish what is the jack inserted - headphone jack, microphone jack, tvout jack, etc, so we can implement it using the jack_status_check function. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Add bit clock rate calculator utility functionsMark Brown2009-11-121-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Many devices need to calculate the bit clock rate desired to work out the clock configuration required for the device. Provide utility functions to do this using both hw_params structures and raw numbers. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * | | ASoC: Factor out snd_soc_init_card()Mark Brown2009-11-031-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_init_card() is always called as the last part of the CODEC probe function so we can factor it out into the core card setup rather than have each CODEC replicate the code to do the initialiastation. This will be required to support multiple CODECs per card. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Codec driver for Texas Instruments tlv320dac33 codecPeter Ujfalusi2009-10-151-0/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Remove snd_soc_suspend_device()Mark Brown2009-10-151-5/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: TPA6130A2 amplifier driverPeter Ujfalusi2009-10-091-0/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Driver for Texas Instruments TPA6130A2 stereo headphone amplifier. The driver provides playback gain control and also pre-defined DAPM_HP widgets and DAPM routings for power management. The DAPM_HP widget names are: "TPA6130A2 Headphone Left" "TPA6130A2 Headphone Right" From soc machine drivers to use with the tpa6130a2 amplifier, the tpa6130a2_add_controls has to be called, which adds the alsa controls and the DAPM routing needed for the tpa6130a2. After that the machine driver can connect the codec's output with 'TPA6130A2 Left' and 'TPA6130A2 Right': {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"}, {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"}, Internally the left and right channels are powered separately. When none of the channels are needed the amplifier is powered down: hard power: valid GPIO number is passed within platform data soft power: Using the software shutdown of the amplifier Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-068-41/+64
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| * | | ASoC: Add virtual enumeration support for DAPM muxesMark Brown2009-10-061-0/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sometimes it is desirable to have a mux which does not reflect any direct register configuration but which will instead only have an effect implicitly (for example, as a result of changing which parts of the device are powered up). Provide a virtual mux for this purpose. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
generated by cgit v1.1 at 2024-12-28 05:41:13 +0000