| Commit message (Collapse) | Author | Age |
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The data format configuration for S3C64xx IISv2 is completely different
to that for S3C24xx. Instead of a single bit configuration in bit 0 of
IISMOD we have format selection in bits 13 and 14 and bit clock rate
selection in bits 1 and 2. While we're here add support for 24 bit
samples in S3C64xx.
At some point it may be desirable to expose the bit clock rate selection
to users but given the limited configuration options that may not be
required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This makes the interface usable with the s3c-iis-v2 rate calculator
and consistent with S3C2412.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This adds a new control named 'Master Playback Switch' for cs4270
codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch
the put function and store the information about manually set mute
controls from userspace. When a manual mute is set, we don't want the
soc core to un-mute the outputs.
Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The control modifies the MUTE register, hence the polarity must be
inverted.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-By: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Make sure we get the DAI operations initialised.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Say what invalid values we're seeing when we see an invalid value and
ensure that errors are displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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It's expected behaviour for the CODEC header to provide them but the
WM8350 doesn't due to having all the registers together under drivers/mfd.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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There is only one LRCLK pin on each interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The S3C64xx IIS code had a number of problems with device registration.
The hardware has two IIS ports of which the driver supported only one
at once via a single exported DAI, attempting to identify the DAI to
use based on the dev->id of the ASoC platform device. As well as
limiting the driver to only supporting one IIS port at once this also
meant that the ID of the soc-audio device (or in future the card device)
had to match the IIS ID.
Fix both problems by converting the driver to register the DAIs based on
probing of platform devices registered by the arch/arm code, using those
platform devices to interact with the clock API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch adds support for the four channel TDM mode
on Beagle board.
Depending on the channel count, the interface needs to be
configured differently (I2S for stereo DSP_A for four channels)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030
codec.
The channel allocations are:
Playback:
TDM i2s TWL RX
Channel 1 Left SDRL2
Channel 3 Right SDRR2
Channel 2 -- SDRL1
Channel 4 -- SDRR1
Capture:
TDM i2s TWL TX
Channel 1 Left TXL1
Channel 3 Right TXR1
Channel 2 -- TXL2
Channel 4 -- TXR2
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add 4 channel support to omap-mcbsp.
This mode is going to be used by the twl4030 codec, when it
is configured in Option1 mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The original idea came from pHilipp, and this makes the code looks
more consistent.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The SSP DMA parameters can actually be easily generated at run-time since
they are almost similar except for the FIFO width and direction. Another
benefit is the re-use of information from 'struct ssp_device', like SSDR
physical FIFO address and DRCMR register index for both directions.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: pHilipp Zabel <philipp.zabel@gmail.com>
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Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have
to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset
Left/Right, Carkit Left/Right) from mux to mixer.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This is now handled by symmetric_rates.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Save a little extra power by enabling the DC servo offset correction
for the output channels only when the relevant channels are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Modify the default startup sequence in the chip to set the DC servo
dither level for optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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CLK_DSP provides a master clock for the DAC and ADC related functionality
on the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.
Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.
Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add checking in hw_params and prepare to detect bufferless pcms(i.e. BT
<--> codec).
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This also switches us to using a switch statement for the widget type
in dapm_power_widget().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This will form a basis for further power check refactoring: the overall
goal of these changes is to allow us to check power separately to
applying it, allowing improvements in the power sequencing algorithms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or
16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono
TX or stereo TX.
The PCM voice interface has two modes
- PCM mode1 : This uses the normal FS polarity and the rising edge of
the clock signal.
- PCM mode2 : This uses the FS polarity inverted and the falling edge
of the clock signal.
If the system master clock is not 26MHz or the twl4030 codec mode is not
option2, the voice PCM interface is not available.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The original implementation of the constraints were good against sane
applications.
If the opening sequence is:
stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the
constraints are set correctly for stream2.
But if the sequence is:
stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2
would receive constraint rate = 0, sample_bits = 0, since the stream1 has not
yet called hw_params...
The command to trigger this event:
gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false
This patch does some 'black magic' in order to always set the correct
constraints and sets it only when it is needed for the other stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.
The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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It has a shared LRCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Some limited volume controls (mostly simple attenuations) have only two
settings so the ASoC info functions misreport them as booleans. Since
we currently have no better information check for " Volume" in the
control name and always report any controls matching as being integer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Also make sure we're checking for the right operation while we're here.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.
Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.
The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.
Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.
The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.
This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)
SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).
This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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