Ensures that the DAI and socdev exported by the codec match up with
their exported prototype.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kbuild ignores dependency from things that are themselves selected so
ASoC machine drivers need to ensure that the control bus is being built.
This also avoids issues where multiple buses are supported by a given
codec.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4270 supports stand-alone mode, where the codec is not connect to the
I2C or SPI buses. Instead, input voltages configure the codec at power-on.
The CS4270 ASoC device driver has partial support for this mode, but the
code was never tested, and partial support doesn't help anyone. It also made
the rest of the code more complicated than necessary.
[Removed redundant CS4270 dependency on I2C -- broonie]
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the acpture switch name so that it better reflects its
purpose.
Signed-off-by: Ian Molton <iann@mnementh.co.uk>
Signed-off-by: Mark 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: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM9713 driver does not support configuring the PLL output frequency
so the output frequency parameter is irrelevant. Allow users to set it
to zero by ignoring it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver adds support for the wm9705 ac97 codec. The driver supports
audio input and output.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark 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: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The soc_value_enum has been merged to soc_enum.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For codecs that have both SPI and I2C support we need to ensure that we
don't try to make the codec driver built in when I2C is modular since
that won't link. Do this by creating a helper variable which uses
conditional defaults to pick up the correct value for all combinations.
We don't need to do anything special for I2C-only codecs since a
conditional select passes on the full value for a tristate.
Reported-by: Ingo Molnar <mingo@elte.hu>
Tested-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Thanks to Troy Kisky <troy.kisky@boundarydevices.com> for noticing.
- DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2
- DSP_B has 0-bit data delay which corresponds to submode 1
- Currently driver sets them opposite so swap the submode setting
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data
delay but configures link for 0-bit data delay which is in fact DSP_B
- Fix this by changing format from DSP_A to DSP_B
- Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same
error is populated also there
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the missing __devexit annotation to wm8350_codec_remove():
sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8350 is an integrated audio and power management subsystem which
provides a single-chip solution for portable audio and multimedia systems.
The integrated audio CODEC provides all the necessary functions for
high-quality stereo recording and playback. Programmable on-chip
amplifiers allow for the direct connection of headphones and microphones
with a minimum of external components. A programmable low-noise bias
voltage is available to feed one or more electret microphones.
Additional audio features include programmable high-pass filter in the
ADC input path.
This driver was originally written by Liam Girdwood with further updates
from me.
Signed-off-by: Mark 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: Stanley.Miao <stanley.miao@windriver.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver now registers the codec and DAI when probed as an I2C device.
Also convert the driver to use a single dynamic allocation to simplify
error handling.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Redo the instantiation of the WM8900 to do most of the initialisation
work when the I2C driver probes rather than when the ASoC device is
instantiated, registering the codec with the ASoC core when done.
Also move all dynamic allocations into a single kmalloc() to simplify
error handling and rename the I2C driver to make output more sensible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To avoid confusion the names for the DACs changed:
DACL1 -> DAC Left1
...
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mux switch related texts fits to on line, no need to wrap
them.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BUG() should be marked as not returning but for at least some
configurations (including some widely deployed compilers) that's either
not happening or being forgotten by the compiler. Add some extra return
statements to the affected paths.
Signed-off-by: Mark 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: Takashi Iwai <tiwai@suse.de>
This makes use of the support for delayed DAI registration to allow the
WM8900 I2C device to be registered by general platform/architecture code
rather than as part of the ASoC device probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Handsfree outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Carkit outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Headset outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the PreDrive outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Earpiece output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four APGA switch to DAPM routing and widgets.
Add user control for DA enable for all APGA as normal
control.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four DACs to dapm_widgets with power switch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Add aic3x_set_headset_detection() function to define the headset
detection mode for tlv32aic3x chips
- added aic3x_button_pressed()
- Read from the real-time registers in aic3x_headset_detected() to query
headset presence without an occured interrupt
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TWL4030 codec device has two ADCs. Both of them can have
several inputs routed to them, but TRM says that only one source
can be selected for every ADC, even though every source has a
dedicated bit in the registers.
This patch adds input source controls. It modifies default register
values to have no inputs selected and ADCs disabled. When some
input is selected, control handlers enable apropriate input
amplifier and ADC. If a microphone is selected, bias power is
automatically enabled. When some input is deselected, unused
chip parts are disabled.
Microphone and line input recording tested on OMAP3 pandora board.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All outputs have dedicated gain controls except the
HandsFree output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Playback volume controls for all four DACs.
All four paths has three levels of volume controls:
Digital Fine gain, Digital Coarse gain, Analog gain.
The controls are named to reflect their connection to the DACs.
Per DAC volume can be performed, if needed:
amixer sset 'DAC1 Analog' 5,10
DACL1 analog gain to 5
DACR1 analog gain to 10
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The digital Capture gain control has a range:
0 to 31 dB in 1 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown