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>
This will allow codec drivers to be refactored to allow them to be
registered out of line with the ASoC device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the lists of platforms, platform DAIs and cards to check to see that
everything has registered. Since relationships are still specified by
direct references to the structures in the drivers and the drivers all
register everything at modprobe there should be no practical effect yet.
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>
This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register all platform DAIs with the core. In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add API calls to register and unregister DAIs with the core. Currently
these APIs are ineffective. Since multiple DAIs for a given device are
a common case bulk variants are provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
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>
This patch adds basic support for OMAP3 Pandora.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
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>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The optimal change would be to move the AC97 register definitions into
the AC97 driver, unfortunately, the registers are shared between several
files. Move them into a dedicated regs-ac97.h first.
Signed-off-by: Eric Miao <eric.miao@marvell.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>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the old-style trigger callback in s3c2443-ac97.c:
sound/soc/s3c24xx/s3c2443-ac97.c:378: warning: initialization from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrong shutdown callback type. Also removed the unused variables
there:
sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
sound/soc/pxa/corgi.c: At top level:
sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9171e5e6a2.
I can't reproduce the compile warnings any more. The warnings
might be some weird cross-compiling set up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dependency on SND_SOC is already fulfilled in sound/soc/Kconfig,
thus no more need in Kconfig of each sub directory.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hide annoying uninitialized warnings:
sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function
sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables more routing functions for tlv320aic3x codecs.
It is now possible to
- control the volume of the PGA bypass path for the HPL, HPR, HPLCOM
and HPRCOM outputs individually
- route right line1 input to the left ADC channel
- route left line1 input to the right ADC channel
- route right mic3 input to left DAC channel
- route left mic3 input to right DAC channel
- route left line1 input to right line1 output
- route right line1 input to left line1 output
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no argument named @clk_id in snd_soc_dai_set_fmt,
remove its' comment.
Signed-off-by: Qinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add ASoC support for TI SDP3430. It's based on Gumstix
Overo SoC code by Steve Sakoman.
Signed-off-by: Misael Lopez Cruz <mesak82@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes Kconfig dependency of TWL4030 audio codec driver
with TWL4030 core driver on both overo and omap2evm
boards
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Acked-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch adds support for mono audio links so that McBSP DAI can operate with
real mono codecs. In I2S, the signalling remains the same but only first
frame (left channel) is transmitting audio data and second frame having null
data. In DSP_A, only first frame is transmitted.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Prepare for upcoming McBSP DAI update adding support for mono links by
restricting number of channels to 2 in N810. This is due tlv320aic3x which
claims channels_min = 1 and playing pure mono audio over I2S would cause
it to be played only from left channel if both cpu and codec DAI's claim to
support mono.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that the ASoC resume has been punted to a workqueue for a release
cycle without attracting bug reports it should be safe to make the
log messages associated with it debug level, reducing noise and kernel
size in production configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Special handling is required for suspend and resume of AC97 codecs
due to the control path going over the data bus.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the gain controls in TWL (mostly those which are associated with
the outputs) are implemented in an interesting way:
0x0 : Power down (mute)
0x1 : 6dB
0x2 : 0 dB
0x3 : -6 dB
Inverting not going to help with these.
Custom volsw and volsw_2r get/put functions to handle these gains.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add CGAIN (Coarse gain control) to TWL4030 codec.
The range of the CGAIN is:
0 dB to 12 dB in 6 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 FGAIN volume control has a range:
-62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Keep Soft-volume disabled for now, since if it is enabled
the FGAIN volume controls are not working in the current
configuration:
CODEC_MODE:OPT_MODE = 1
OPTION:ARXR2_EN = 1
OPTION:ARXL2_EN = 1
OPTION:ARXR1_EN = 0
OPTION:ARXL1_VRX_EN = 0
RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1)
RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1)
After the patch, FGAIN volume control works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.
The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up our record of the active streams in shutdown(), fixing
subsequent failures of snd_pcm_hw_constraints_complete after closure of
a stream.
NOTE:
- The ssm2602 allows pairs of non-matching PB/REC rates.
- This is a fix for less evil:
The logic is flawed (e.g. the slave might startup before the
master's rate and sample_bits are set).
Signed-off-by: Karl Beldan <karl.beldan@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds twl4030 audio support on omap2evm
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an
FPGA that generates the bit clock and the master clock
[Downgraded the rate debug print to pr_debug() in hw_params, converted
asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is a 16-bit stereo audio codec. It accepts
left-justified format for ADC, and right-justified format
for DAC. Independent power-down modes for ADC and DAC are
provided, as well as a digital de-emphasis filter (4 modes).
[Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A probe function should have a clean return 0 path.
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Michael Hennerich <michael.hennerich@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clean up redudent code and correct building problem in non-mmap mode
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch provides a option for users to enable multi-channel function support
in Blackfin ASoC driver. Because Blackfin is without MMU, it is easy for us and
the user to enable this function at compiling stage not dynamically on the fly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We added multi-channel function to this codec driver and Blackfin ASoC driver as well.
It was tested on Blackfin hardware.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
tweak SPORT range for non-BF54x so we get proper behavior for BF52x parts
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A small additional power saving can be achieved for the WM8990 by
maintaining VMID using a 2*250k divider when in standby mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable a hardware workaround which avoids problems with the clocking of
the ADCs in certain configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only fully documented registers are cached in the WM8990 but additional
registers exist.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FGAIN for playback is in range of 0-0x3f, while for capture GAIN it
is in the range of 0-0x1f.
The original value of 128 (0x7f) would modify the CGAIN also for
playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8728 is a high performance stereo DAC designed for applications
such as DVD, home theatre and digital TV.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 8dc840f88d. Christian
Pellegrin <chripell@gmail.com> reported that on some systems the patch
caused DMA to fail which is much more serious than the original skipped
audio issue. Further investigation by Dave shows that the behaviour
depends on the clock speed of the SoC - a better fix is neeeded.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this patch it is possible to select drivers which require
bestcomm support without bestcomm support being selected. This
patch reworks the bestcomm dependencies to ensure the correct
bestcomm tasks are always enabled.
Reported-by: Hans Lehmann <hans.lehmann@ritter-elektronik.de>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Originally it was put too tight limits to support only 44.1 kHz and 48 kHz
sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With
96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?).
Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas
Instruments Beagle with TWL4030 from rates 8 - 48 kHz.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec
mode register accordingly in twl4030_hw_params. Expose this info so that
ASoC can match other rates than 44.1 kHz or 48 kHz as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes swapping of channels at start of stereo playback.
Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
