- May 26, 2010
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Guennadi Liakhovetski authored
Signed-off-by:
Guennadi Liakhovetski <g.liakhovetski@gmx.de> Signed-off-by:
Mark Brown <broonie@opensource.wolfsonmicro.com>
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- May 11, 2010
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Mark Brown authored
If the FLL is not configured attempting to resume it will produce a warning message so skip the resume. Signed-off-by:
Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Disable the output stage prior to the delay stage rather than the other way around. Fixes merge issue with previous headphone output path corrections. Signed-off-by:
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Mark Brown authored
Log the values we're getting back from the DC servo and the values we write to it. Signed-off-by:
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Mark Brown authored
Signed-off-by:
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Peter Ujfalusi authored
If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by:
Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by:
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- May 10, 2010
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Mark Brown authored
Make dev_() prints much prettier. Signed-off-by:
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Mark Brown authored
This allows more flexible integration with subsystem features. Signed-off-by:
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Mark Brown authored
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:
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Mark Brown authored
Acked-by:
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Mark Brown authored
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:
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Mark Brown authored
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:
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Mark Brown authored
We now manage suspend within the main power analysis rather than by flipping the state of widgets. Tested-by:
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Mark Brown authored
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:
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- May 07, 2010
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Jassi Brar authored
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:
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Jassi Brar authored
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:
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Peter Ujfalusi authored
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:
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Peter Ujfalusi authored
This reverts commit 6f399115 . 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:
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Peter Ujfalusi authored
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:
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Mark Brown authored
Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35
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Jassi Brar authored
The S3C DMA API doesn't make use of hw_addr.to/from and also the FIFO addresses are provided from the I2S drivers. So these fields are redundant. This patch removes the hw_addr.to/from fields for I2S and the inclusion of header, paving way for the header to be moved closer to the I2S controller drivers. Signed-off-by:
Ben Dooks <ben-linux@fluff.org> Signed-off-by:
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- May 06, 2010
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Takashi Iwai authored
Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
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Peter Ujfalusi authored
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:
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Peter Ujfalusi authored
Add support for platform dependent gain limiting on the tpa6130a2 (and tpa6140a2) Headset amplifier. Signed-off-by:
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Jarkko Nikula authored
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:
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Jarkko Nikula authored
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:
Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by:
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- May 05, 2010
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Jassi Brar authored
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:
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Jassi Brar authored
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:
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Jassi Brar authored
Now that the fields are defined for s3c2412, use them and avoid having multiple copies of same defines. Signed-off-by:
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Jassi Brar authored
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:
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Jassi Brar authored
No need to keep redundant field iis_clk in s3c_i2sv2_info. iis_cclk and iis_pclk is all we need. Signed-off-by:
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Jassi Brar authored
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:
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Jassi Brar authored
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:
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Jassi Brar authored
Define more bit definitions in the order of mainline support for the SoC. Signed-off-by:
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Jassi Brar authored
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:
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Mark Brown authored
Signed-off-by:
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Peter Ujfalusi authored
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:
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- May 03, 2010
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Peter Ujfalusi authored
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:
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Peter Ujfalusi authored
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:
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Peter Ujfalusi authored
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:
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