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* ASoC: Allow DAI links to be kept active over suspendMark Brown2010-05-101-1/+37
| | | | | | | | | | | | | As well as allowing DAPM pins to be marked as ignoring suspend allow DAI links to be similarly marked. This is primarily intended for digital links between CODECs and non-CPU devices such as basebands in mobile phones and will suppress all suspend calls for the DAI link. It is likely that this will need to be revisited if used with devices which are part of the SoC CPU. Tested-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 active paths from the GSM modem while the GTA02 is suspendedMark Brown2010-05-101-0/+8
| | | | | Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Support leaving paths enabled over system suspendMark Brown2010-05-102-5/+58
| | | | | | | | | | | | | | | | | | | | | | | Some devices can usefully run audio while the Linux system is suspended. One of the most common examples is smartphone systems, which are normally designed to allow audio to be run between the baseband and the CODEC without passing through the CPU and so can suspend the CPU when on a voice call for additional power savings. Support such systems by providing an API snd_soc_dapm_ignore_suspend(). This can be used to mark DAPM endpoints as not being sensitive to system suspend. When the system is being suspended paths between endpoints which are marked as ignoring suspend will be kept active. Both source and sink must be marked, and there must already be an active path between the two endpoints prior to suspend. When paths are active over suspend the bias management will hold the device bias in the ON state. This is used to avoid suspending the CODEC while it is still in use. Tested-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: Refactor DAPM suspend handlingMark Brown2010-05-102-23/+33
| | | | | | | | | | | | | | | | | | Instead of using stream events to handle power down during suspend integrate the handling with the normal widget path checking by replacing all cases where we report a connected endpoint in a path with a function snd_soc_dapm_suspend_check() which looks at the ALSA power state for the card and reports false if we are in a D3 state. Since the core moves us into D3 prior to initating the suspend all power checks during suspend will cause the widgets to be powered down. In order to ensure that widgets are powered up on resume set the card to D2 at the start of resume handling (ALSA API calls require D0 so we are still protected against userspace access). Tested-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: Remove unused DAPM suspend flagMark Brown2010-05-101-10/+0
| | | | | | | | | We now manage suspend within the main power analysis rather than by flipping the state of widgets. Tested-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: Remove unneeded suspend bias managment from CODEC driversMark Brown2010-05-1021-31/+7
| | | | | | | | | | | The core will ensure that the device is in either STANDBY or OFF bias before suspending, restoring the bias in the driver is unneeded. Some drivers doing slightly more roundabout things have been left alone for now. Tested-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: SMDK64XX: Switch to IISv4 CPU driverJassi Brar2010-05-072-8/+5
| | | | | | | | | | Switch the MACHINE driver to use IISv4 CPU dai. Remove BROKEN dependency now that we have proper CPU driver available. Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4 controller. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: S3C64XX: IISv4: Add CPU driverJassi Brar2010-05-074-0/+217
| | | | | | | | | | Add the CPU driver for the IISv4 block found on S3C6410. For now, the driver is almost a copy of s3c64xx-i2s.c but it should diverge as more IISv4 specific stuff is added. 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: tpa6130a2: Fix for the custom kcontrol functionsPeter Ujfalusi2010-05-071-8/+10
| | | | | | | | | | | Since the functions arre only used for volume register, change their name, and also fix them to properly handle the cases, when via soc core the volume is limited. 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>
* Revert "ASoC: tpa6130a2: Support for limiting gain"Peter Ujfalusi2010-05-071-68/+8
| | | | | | | | | | | | 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 Ujfalusi2010-05-071-0/+39
| | | | | | | | | | | | | | | | | | | | | 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 'topic/asoc' of ↵Mark Brown2010-05-079-143/+601
|\ | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35
| * Merge branch 'for-2.6.35' of ↵Takashi Iwai2010-05-069-143/+601
| |\ | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| | * ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC powerPeter Ujfalusi2010-05-061-11/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Do not change the codec defaults for the following registers: 0x40, 0x41: Line output gains, do not use amplification 0x42: LOM/LOP Voltage hold, and selection 0x44: LOM inversion control It has been found, that the values configured to these registers can cause amplification, which can make the output of DAC33 distorted. The codec reset values are considered safe in all environmnts. 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: tpa6130a2: Support for limiting gainPeter Ujfalusi2010-05-061-8/+68
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Nikula2010-05-061-0/+25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: omap: Add basic audio support for Nokia RX-51/N900Jarkko Nikula2010-05-063-0/+306
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds support for integrated stereo speakers and digital microphone found on Nokia RX-51 hardware. This is a cut down version based on Maemo kernel sources and earlier patchset by Eduardo Valentin et al. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Eduardo Valentin <eduardo.valentin@nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Eduardo Valentin <eduardo.valentin@nokia.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: tpa6130a2: TLV mapping for tpa6140a2Peter Ujfalusi2010-05-041-3/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Both tpa6130a2, and tpa6140a2 is supported by the same driver, but the gain dB scaling is different on the amplifiers. Provide different mixer control for the chips with correct TLV mapping. User space will see: "TPA6130A2 Headphone Playback Volume" in case of 6130 "TPA6140A2 Headphone Playback Volume" in case of 6140 The way machine drivers are using this amplifier remained the same. 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: Support for turning off the codecPeter Ujfalusi2010-05-031-21/+45
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Let the codec to hit OFF instead of STANDBY, when there is no activity. When the codec is off, than the associated regulator can be also turned off (if the number of users on the regulator is 0). After initialization, the codec remains in power off, it is only turned on for reading the ID registers (also testing the regulators). The codec power is enabled, when the codec is moving from BIAS_OFF to BIAS_STANDBY. The codec is turned off, when it hits BIAS_OFF. There are few scenarios, which has to be taken care:: 1. Analog bypass caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, but we does not need to execute the playback related configuration 2. Playback caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, and also we need to execute the playback related configuration. 3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is already on. 4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON) Nothing need to be done. 5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is still on. Since the power up, and the codec init is optimized, the added overhead in stream start is minimal. Withing this patch, the hard_power function is now only doing what it supposed to: only handle the powers, and GPIO reset line. The codec initialization and state restore has been moved out. 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: Manage a pointer for snd_pcm_substream in private structurePeter Ujfalusi2010-05-031-0/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As a preparation for supporting codec to be turned off, when we are in BIAS_STANDBY. The substream must be easily available in other places than pcm_* callbacks. Manage a pointer in _startup, and _shutdown for this. 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: Revised module loading, and DAC33 ID readPeter Ujfalusi2010-05-031-19/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Optimize the way how tlv320dac33 is powered uppon module and soc initialization. Also read the DAC33 ID registers, and update the reg_cache to reflect it. 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: Optimize power up, and restorePeter Ujfalusi2010-05-031-67/+39
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On power up we only need to initialize the codec, and restore only registers, which are not in either in DAPM nor in the playback start sequence. These are mostly gain related 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: TWL4030: Remove OUTL/R outputsPeter Ujfalusi2010-05-031-4/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | OUTL/R are leftovers from the original driver, and they are no longer needed. 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: AIF/APLL fix in DAPM domainPeter Ujfalusi2010-05-031-22/+60
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. 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 - disable regulators at i2c remove()Liam Girdwood2010-04-281-0/+1
| | | | | | | | | | | | | | | | | | 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: zoom2 - update DAPM pinsLiam Girdwood2010-04-281-3/+0
| | | | | | | | | | | | | | | | | | | | | | | | Remove bogus twl4030 pins 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: pandora - update DAPM pinsLiam Girdwood2010-04-281-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | Remove bogus TWL4030 pins. 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: S3C: I2S: Move set_sysclk to common codeJassi Brar2010-05-053-67/+50
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Now that we can specify feature of a particular controller, we can avoid multiple copies of same code by defining the CDCLKCON bit feature in controller specific code and detecting that flag in the code common to all controllers. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C: I2Sv2: New field for controller featureJassi Brar2010-05-051-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to make s3c-i2s-v2.c manage controllers with minor quirks and variation in features, we define a per-block flag that indicates the availability/lack of a particular feature to the s3c-i2s-v2.c While adding support for new SoCs' I2S, check for the blocks of older SoCs that have similar feature and set the flag for that feature. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C64XX: I2S: Use s3c2412 definesJassi Brar2010-05-052-5/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Now that the fields are defined for s3c2412, use them and avoid having multiple copies of same defines. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C: I2Sv2: Unify i2s_get_clock callbackJassi Brar2010-05-057-30/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock and discard other two copies. Also, switch the users to make calls to the newly defined and generic s3c_i2sv2_get_clock Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C: I2Sv2: Discard redundant field iis_clkJassi Brar2010-05-051-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | No need to keep redundant field iis_clk in s3c_i2sv2_info. iis_cclk and iis_pclk is all we need. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C2412: I2S: Return correct source clockJassi Brar2010-05-051-1/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Until now, s3c2412_get_iisclk would return NULL since iis_clk was never initialized. Return appropriate pointer as per the selection made for source clock. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C2412: I2S: Debug IMS fieldJassi Brar2010-05-053-33/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The IMS field of s3c2412/13 is essentially the same as that of s3c64xx. That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit selects source clock for signal generation. For that reason, remove improper defines for IISMOD[11:10] field mask and define two 1bit fields that can be set independent of each other. As a consequence, corresponding fields for PLAT_S3C64XX too get to use these new defines. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: SAMSUNG: I2S: Add bit definitionsJassi Brar2010-05-051-2/+40
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Define more bit definitions in the order of mainline support for the SoC. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C: I2Sv2: Move defines closer to driverJassi Brar2010-05-054-5/+85
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The header for I2Sv2 linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h contains only controller specific definitions and nothing SoC specific. So, it could be moved to sound/soc/s3c24xx/ Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Ben Dooks <ben-linux@fluff.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Add debug output tracing all cache register writesMark Brown2010-05-051-0/+12
|/ / | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Add WM9090 amplifier driverMark Brown2010-04-304-0/+1494
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Remove redundant WM8960 SYSCLKSEL clkdiv optionMark Brown2010-04-282-5/+0
|/ | | | | | | The SYSCLK source is automatically managed when configuring the PLL. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Merge branch 'for-2.6.35' of ↵Takashi Iwai2010-04-272-54/+277
|\ | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| * ASoC: tlv320aic3x: Add basic regulator supportJarkko Nikula2010-04-271-0/+37
| | | | | | | | | | | | | | | | | | This patch adds the TLV320AIC3x supplies and enables all of them for the entire lifetime of the device. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: tlv320aic3x: Change bias management semanticsJarkko Nikula2010-04-271-7/+3
| | | | | | | | | | | | | | | | | | | | Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with BIAS_STANDBY where PLL is disabled. Remove also old comments about power control. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: tlv320aic3x: Remove needless power off from aic3x_set_bias_levelJarkko Nikula2010-04-271-34/+0
| | | | | | | | | | | | | | | | | | | | | | These ADC, DAC and output pin power off commands are needless in aic3x_set_bias_level since they are not enabled in aic3x_init and they are defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them anyway. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: tlv320aic3x: Remove unused version stringJarkko Nikula2010-04-271-2/+0
| | | | | | | | | | | | Signed-off-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: FIFO caused delay reportingPeter Ujfalusi2010-04-261-5/+217
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Delay reporting for the three implemented DAC33 FIFO modes. DAC33 has FIFO depth status register(s), but it can not be used, since inside of pcm_pointer we can not send I2C commands. Timestamp based estimation need to be used. The method of calculating the delay depends on the active FIFO mode. Bypass mode: FIFO is bypassed, report 0 as delay Mode1: nSample fill mode. In this mode I need to use two timestamp ts1: taken when the interrupt has been received ts2: taken before writing to nSample register. Interrupts are coming when DAC33 FIFO depth goes under alarm threshold. Phase1: when we received the alarm threshold, but our workqueue has not been executed (safeguard phase). Just count the played out samples since ts1 and subtract it from the alarm threshold value. Phase2: During nSample burst (after writing to nSample register), count the played out samples since ts1, count the samples received since ts2 (in a burst). Estimate the FIFO depth using these and alarm threshold value. Phase3: Draining phase (after the burst read), count the played out samples since ts1. Estimate the FIFO depth using the nSample configuration and the alarm threshold value. Mode7: Threshold based fill mode. In this mode one timestamp is enough. ts1: taken when the interrupt has been received Interrupts are coming when DAC33 FIFO depth reaches upper threshold. Phase1: Draining phase (after the burst), counting the played out samples since ts1, and subtract it from the upper threshold value. Phase2: During burst operation. Using the pre calculated time needed to play out samples from the buffer during the drain period (from upper to lower threshold), move the time window to cover the estimated time from the burst start to the current time. Calculate the samples played out since lower threshold and also the samples received during the same time. 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: Calculate the interface speed during burstsPeter Ujfalusi2010-04-261-0/+5
| | | | | | | | | | | | | | | | | | | | | | When the DAC33 FIFO is in use the dai interface is running in much higher speed than the sampling frequency. Calculate the rate based on the internal base frequency and the bclk divider. 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: Change magic numbers used in Mode7Peter Ujfalusi2010-04-261-5/+6
| | | | | | | | | | | | | | | | | | Upper and Lower threshold values are used as magic numbers. Replace them with defines for later use. 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: Skip calculations in FIFO Bypass modePeter Ujfalusi2010-04-261-0/+5
| | | | | | | | | | | | | | | | | | | | There is no need for calculations for FIFO bypass mode. Just in case set the nsample maximum limit, which has been done in the calculation phase. 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: Fix for early interrupt in FIFO Mode1Peter Ujfalusi2010-04-261-2/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Alarm threshold interrupt is triggered right after the playback start. This interrupt is recieved during the first burst period, and caused the state machine to write additional nSample command, which has to be avoided. To fix this issue move the DAC33 interrupt unmasking after we configured the PREFILL register with a small delay. 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: UDA134X: Add UDA1345 CODEC supportVladimir Zapolskiy2010-04-261-0/+13
| | | | | | | | | | | | | | | | | | 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>
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