This patch add voice digital loopback (sidetone) to the twl4030
driver. It mixes voice uplink attenuated (by sidetone gain) with
voice downlink when the codec is working in option2 (voice/audio
mode).
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds voice downlink analog bypass switch. It follows
the same approach as in other analog bypass switches.
DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer',
that will also allow voice DAC to be powered in digital voice
loopback (sidetone).
Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The gain control for earpiece amplifier uses 0dB ~ 12dB according to the
TRM, but the present code is implemented to -6dB ~ 6dB.
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NJoonyoung Shim <jy0922.shim@samsung.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NJoonyoung Shim <jy0922.shim@samsung.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix for compillation error introduced by the constrain patch.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds the needed code to be able to use 96KHz playback.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In case of duplex mode (capture and playback at the same time), the second
stream has to have the same parameters (rate, sample size) as the already
running stream.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

TWL4030 supports 96KHz sample playback, but only playback.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

CC      sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Enum type for selecting the desired ramp delay for the headset output.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds the digital loopback/bypass support for twl4030 codec.

The digital loopback will let the digimic0 (routed in the TX1 capture path
inside of TWL4030) data to be routed back to the RX2 playback path
(I2S stereo). It can also route the analog capture date routed through the
TX1 back to RX2.

Effectively the digital loopback is routing the audio from the TX1 capture path
to the RX2 playback path.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds the analog loopback/bypass support for twl4030 codec.

Details for the implementation:
It seams that the analog loopback needs the DAC powered on on the channel,
where the loopback is selected. The switch for the DACs has been moved from
the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be
powered while the audio playback is used or/and the loopback is enabled for
the channel.

The twl4030 codec powering has been reworked. Now the codec will be powered as
long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are
when the given change in the registers needs the codec power down/up cycle in
order to take effect. Otherwise the codec is on.

When the codec enters to STANDBY state and none of the loopback paths are
enabled, than the amplifiers, which are no in the DAPM path are forced to turn
off and the PLL is disabled. When playback/capture starts the disabled gains
are restored and the PLL is enabled.

When one of the loopback enabled in STANDBY mode, the disabled gains are
restored and the PLL is enabled also.

In short: the codec always goes to the lowest power state based on the
bias_level and the bypass_state.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Move the twl4030_power_up and twl4030_power_down function
earlier to facilitate the analog bypass implementation.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Change the power switches for the physical ADC and for the
amplifiers for the analog capture path to map more closely
the actual path inside of the codec.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The Headset Left anti-pop and bias ramp does not need to be
enabled, if the headset is not in use.
Move the code to DAPM event handler for HeadsetL.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge the codec up and down functions to a simple one.
Codec is powered down by default (reg_cache change).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The offset cancelation bit in ANAMICL register is self cleanig.
Make sure that the reg_cache holds the same value as the HW
register.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This is a further stage on the road to refactoring away the ASoC
platform device.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch takes fixes a number of bugs in the caching code used by
several ASoC codec drivers. Mostly off-by-one fixes.
Signed-off-by: NIan Molton <ian@mnementh.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Call the snd_soc_free_pcm and snd_soc_dapm_free when the
codec driver is unloaded.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.

This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.

[Edited to raise priority of error log message and document parameters.
 -- broonie]
Signed-off-by: NIan Molton <ian@mnementh.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The soc_value_enum has been merged to soc_enum.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Convert the bitfield coded enums to the new VALUE_ENUM type.
Remove the enum check, since the VALUE_ENUM type can handle
the bitfield coding and also handles the 'holes' in the bitfield.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds DAPM implementaion for the capture path
on twlx030.

TWL has two physical ADC and two digital microphone (stereo) connections.
The CPU interface has four microphone channels.
For simplicity the microphone channel paths are named as:
TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid
TX2 (Left/Right)

Input routing (simplified version):
There is two levels of mux settings for TWL in input path:
Analog input mux:
 ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic}
 ADCR <- {Off, Sub mic, AUXR}

Analog/Digital mux:
TX1 Analog mode:
 TX1L <- ADCL
 TX1R <- ADCR
TX1 Digital mode:
 TX1L <- Digimic0 (Left)
 TX1R <- Digimic0 (Right)

TX2 Analog mode:
 TX2L <- ADCL
 TX2R <- ADCR
TX2 Digital mode:
 TX2L <- Digimic1 (Left)
 TX2R <- Digimic1 (Right)

The patch provides the following user controls for the capture path:
Mux settings:
"TX1 Capture Route": {Analog, Digimic0}
"TX2 Capture Route": {Analog, Digimic1}
"Analog Left Capture Route":  {Off, Main Mic, Headset Mic, AUXL, Carkit Mic}
"Analog Right Capture Route": {Off, Sub Mic, AUXR}

Volume/Gain controls:
"TX1 Digital Capture Volume": Stereo gain control for TX1 path
"TX2 Digital Capture Volume": Stereo gain control for TX2 path
"Analog Capture Volume":      Stereo gain control for the analog path only

Important things for the board files:
Microphone bias:
"Mic Bias 1":       Bias for Main mic or for digimic0 (analog or digital path)
"Mic Bias 2":       Bias for Sub mic or for digimic1 (analog or digital path)
"Headset Mic Bias": Bias for Headset mic

When the routing configured correctly only the needed components will be
powered/enabled.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Modify the enum filter to more generic that it will filter
out the enums with text "Invalid".
The enum filter also required for the capture path.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

A special start-up sequence is required to reduce the pop-noise of Class D
amplifier when enable hands-free on TWL4030.
Signed-off-by: NStanley.Miao <stanley.miao@windriver.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

To avoid confusion the names for the DACs changed:
DACL1 -> DAC Left1
...
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The mux switch related texts fits to on line, no need to wrap
them.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fixed the function name of module init entry for twl4030.c, which
conflicted with the existing hardware init function:
  sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init'
  sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here

Also fixed the section type of init function.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Currently this is done at module probe time since ASoC ties in codec
device probe to the instantiation of the entire ASoC device. Subsequent
patches will refactor the codec drivers to handle probing separately.
Note that the core does not yet use this information.

AC97 is special since the codec is controlled over the AC97 link but
we want to give the machine driver a chance to set up the system before
trying to instantiate since it may need to do configuration before the
AC97 link will operate
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There is a separate gain control for the Headset output already.
Do not reset the gain to 0 dB at power up.
In power-down, there is no need to set the Headset output gain
to power-down mode, since if the CODECPDZ is in powered off this
setting has no effect.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds DAPM muxing, routing for the Handsfree outputs.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds DAPM muxing, routing for the Carkit outputs.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds DAPM muxing, routing for the Headset outputs.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds DAPM muxing, routing for the PreDrive outputs.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Adds DAPM muxing, routing for the Earpiece output.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>