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* | | ASoC: Pass dai_link as argument to platform suspend and resumejassi brar2010-02-22
| |/ |/| | | | | | | | | | | | | | | | | Passing pointer to relevant dai_link provides easier reach to the ASoC tree in suspend/resume of snd_soc_platform. It also provides direct access to the dai at the other end of the dai_link. 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: Make pmdown_time a longMark Brown2010-02-17
| | | | | | | | | | | | | | 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 Brown2010-02-16
| | | | | | | | | | | | | | | | | | | | 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 Brown2010-02-12
| | | | | | | | | | | | | | | | | | | | 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 Brown2010-02-04
| | | | | | | | | | | | | | | | | | | | 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 Brown2010-02-03
| | | | | | | | | | | | | | | | | | | | | | 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 Liakhovetski2010-01-25
| | | | | | | | | | | | | | | | | | 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 Liakhovetski2010-01-25
| | | | | | | | | | | | | | | | | | | | | | 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 Brown2010-01-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Ujfalusi2010-01-20
| | | | | | | | | | | | | | | | | | | | | | | | | | 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 Liakhovetski2010-01-19
| | | | | | | | | | | | 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 Brown2010-01-12
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| * Merge branch 'for-linus' of ↵Linus Torvalds2009-12-09
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Rosa2009-12-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Delvare2009-12-04
| | | | | | | | | | | | | | | Signed-off-by: Jean Delvare <jdelvare@suse.de> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
| * | Merge branch 'topic/hda' into for-linusTakashi Iwai2009-12-04
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| | * \ Merge branch 'topic/ice1724-quartet' into topic/hdaTakashi Iwai2009-12-01
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| | | * | ALSA: ak4113 supportPavel Hofman2009-09-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 Hofman2009-09-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 Hofman2009-09-21
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 Iwai2009-12-04
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| * \ \ \ Merge branch 'topic/misc' into for-linusTakashi Iwai2009-12-04
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| | * | | | ALSA: opti-miro: expose ACI mixer to outside driversKrzysztof Helt2009-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The ACI mixer is used to control the radio FM module installed on the Miro PCM20 sound card. Expose ACI mixer outside the sound card driver. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | | ALSA: opti-miro: make miro.h header available outside the alsa directoryKrzysztof Helt2009-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Move the miro.h header to the include/sound directory. It can be used in the Miro PCM20 radio driver (v4l). Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | | ALSA: cs4236: update control namesKrzysztof Helt2009-11-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Update control names to be more closer to their meaning. Change the "Mono" name to the "Beep" as this line is usually used to forward the PC beeper signal to sound card's output. Update names for both cs423x and wss. Clean up cs4235 controls according to the cs4235 doc. Rename some of the cs4235 controls to be consistent with the cs4236's ones. Also, delete one misnamed cs4231 register define. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | | ALSA: cs4236: detect chip in one passKrzysztof Helt2009-11-05
| | | |/ / | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The cs4236 was two step detection with call to the snd_wss_free() between two steps. The snd_wss_free() did not free a sound device created in the snd_wss_create(). This caused an OOPS during module removal as the same sound device was released twice. The same OOPS happened if the cs4236 module loading failed. Fix this by adapting the snd_cs4236_create() to correctly work with chips less capable then cs4236. The snd_cs4236_create() behaves the same as the snd_wss_create() if the chip is less capable than the cs4236. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | Merge branch 'topic/beep-rename' into topic/core-changeTakashi Iwai2009-12-01
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| | * | | ALSA: sh: add SuperH DAC audio driver for ALSA V4Rafael Ignacio Zurita2009-11-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is a port of the sound/oss/sh_dac_audio.c driver. The driver uses an on-chip 8-bit D/A converter, which has a speaker connected to one of its channels, found in several ancient HP machines. For interrupts it uses a high-resolution timer (hrtimer). Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx). Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver would be obsolete soon, and it could be removed. Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com> Acked-by: Paul Mundt <lethal@linux-sh.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: sscape - Remove sscap_ioctl.h from include/sound/KbuildTakashi Iwai2009-10-02
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: sscape: convert to firmware loader frameworkKrzysztof Helt2009-10-01
| | | |/ | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | sound: rawmidi: record a substream's owner processClemens Ladisch2009-11-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Record the pid of the task that opened a RawMIDI substream. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | sound: pcm: record a substream's owner processClemens Ladisch2009-11-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Record the pid of the task that opened a PCM substream. For sound cards with hardware mixing, this allows determining which process is associated with a specific substream's volume control. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | control: use reference-counted pidClemens Ladisch2009-11-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead of storing the PID number, take a reference to the task's pid structure. This protects against duplicates due to PID overflows, and using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is correct as seen from the current namespace. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | control: remove snd_konctrol_volatile::owner_pid fieldClemens Ladisch2009-11-06
| |/ / | | | | | | | | | | | | | | | | | | | | | We do not need to save the ID of the process that locked a control because that information is already available in the owner's file data. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ASoC: tpa6130a2: Support for tpa6140's regulatorsIlkka Koskinen2010-01-11
| | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2009-12-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2009-12-04
| |/ |/| | | | | | | | | | | Allows custom controls to use it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Add BCLK calculation utility for TDM mode tooMark Brown2009-11-25
| | | | | | | | | | 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 Shim2009-11-12
| | | | | | | | | | | | | | | | | | | | | | 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 Brown2009-11-12
| | | | | | | | | | | | | | | | | | | | 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 Brown2009-11-03
| | | | | | | | | | | | | | | | | | 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 Ujfalusi2009-10-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2009-10-15
| | | | | | | | | | | | | | | | 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 Ujfalusi2009-10-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2009-10-06
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| * Merge branch 'topic/ymfpci' into for-linusTakashi Iwai2009-09-10
| |\ | | | | | | | | | | | | * topic/ymfpci: sound: ymfpci: increase timer resolution to 96 kHz
| | * sound: ymfpci: increase timer resolution to 96 kHzClemens Ladisch2009-08-10
| | | | | | | | | | | | | | | | | | | | | | | | Allow the interval timer to be programmed with its full 96 kHz precision. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'topic/tlv-minmax' into for-linusTakashi Iwai2009-09-10
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | * topic/tlv-minmax: ALSA: usb-audio - Correct bogus volume dB information ALSA: usb-audio - Use the new TLV_DB_MINMAX type ALSA: Add new TLV types for dBwith min/max
| | * | ALSA: Add new TLV types for dBwith min/maxTakashi Iwai2009-06-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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