- 07 May, 2010 4 commits
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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:
Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by:
Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by:
Mark Brown <broonie@opensource.wolfsonmicro.com>
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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:
Jassi Brar <jassi.brar@samsung.com> Acked-by:
Ben Dooks <ben-linux@fluff.org> Signed-off-by:
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- 06 May, 2010 5 commits
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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.htmlSigned-off-by:
Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by:
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- 05 May, 2010 10 commits
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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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- 04 May, 2010 1 commit
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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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- 03 May, 2010 6 commits
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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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Peter Ujfalusi authored
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:
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Peter Ujfalusi authored
OUTL/R are leftovers from the original driver, and they are no longer needed. Signed-off-by:
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Peter Ujfalusi authored
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:
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- 30 Apr, 2010 1 commit
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Mark Brown authored
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:
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- 28 Apr, 2010 4 commits
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Liam Girdwood authored
Acked-by:
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Liam Girdwood authored
Remove bogus twl4030 pins Acked-by:
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Liam Girdwood authored
Remove bogus TWL4030 pins. Acked-by:
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Mark Brown authored
The SYSCLK source is automatically managed when configuring the PLL. Signed-off-by:
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- 27 Apr, 2010 5 commits
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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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Jarkko Nikula authored
This patch adds the TLV320AIC3x supplies and enables all of them for the entire lifetime of the device. Signed-off-by:
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Jarkko Nikula authored
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:
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Jarkko Nikula authored
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:
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Jarkko Nikula authored
Signed-off-by:
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- 26 Apr, 2010 4 commits
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Vladimir Zapolskiy authored
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:
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Mark Brown authored
Signed-off-by:
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Mark Brown authored
Reported-by:
Anti Sullin <anti.sullin@artecdesign.ee> Signed-off-by:
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Peter Ujfalusi authored
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:
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