Checking if IMX_SSI_DMA is set and then set it again is useless.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SSI counts in words, the DMA engine in bytes. (Wrong) factor got removed
in bf974a0 (ASoC i.MX: switch to new DMA api).
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
They got accidently removed by f0fba2a (ASoC: multi-component - ASoC
Multi-Component Support). Reintroduce them and get rid of the
superfluous defines because the fiq-driver has its own hardcoded values.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
DSP2 in the WM8958 can be used to support an upgraded EQ for use in
demanding applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
With appropriate firmware the WM8958 can support Virtual Surround Sound or
VSS, widening the stereo audio image for improved user experience. Enable
support for this mode of operation when the appropriate firmware can be
loaded at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In preparation for the addition of additional WM8958 algorithms
reorganise the code to make it easier to add such support later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow userspace to supply an update to the ROM firmware. The firmware
request is non-blocking so userspace can load the firmware at its
leisure without delaying startup, the driver will begin using the
firmware the next time MBC is started after it has been supplied.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The regulator is optional depending on board design.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The DSP2 startup requires that the clock be enable so if we've deferred
clock startup we need to defer DSP2 startup
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
DSP2 on the WM8958 has a default ROM which provides a multi-band
compressor for enhanced performance on mobile devices but can also
support runtime download of alternative firmware. In preparation for
more exploiting this functionality refactor the code to split the
handling of DSP2 into a separate file.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The first WM8958 revision requires similar treatment.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since not all registers need to be cached and the cache is entirely
optional anyway we shouldn't be checking that a register is in the
cached range. If the register is invalid then the actual I/O code
can determine that and report an error.
Similarly, the step size can and should be enforced by the lower level
code if it's important.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC audio for mini2440 platform in current kenrel doesn't work.
First problem is samsung_asoc_dma device is missing in initialization.
Next problem is with codec. Codec is initialized but never probed
because no platform_device exist for codec driver. It leads to errors
during codec binding to asoc dai. Next problem was platform data which
was passed from board to asoc main driver but not passed to codec when
called codec_soc_probe().
Following patch should fix issues. But not sure if in correct way.
Please review.
Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
MONO was renamed to MONO1.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This patch adds ASoC support for the MAX9850 codec with headphone
amplifier.
Supported features:
- Playback
- 16, 20 and 24 bit audio
- 8k - 48k sample rates
- DAPM
Signed-off-by: Christian Glindkamp <christian.glindkamp@taskit.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this fix the driver won't instantiate properly on relevant
devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Without this fix the driver won't instantiate properly on relevant
devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Enable 192kHz sample rate for EP93xx.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Improve EP93xx I2S clocks management.
Some freqs values are set not exact as they requested for MCLK and
original code was not able to find divisors for SCLK and LRCLK.
This code just picks up nearest value from 3 possible variants.
This patch makes 44100 and 192000 rates working and fixes
capture function (by selecting SCLK/LRCLK=64 where possible).
All other rates should work as before.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx
machine driver.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC.
This is done to make capture work properly.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're not only prefixing all controls, we're also prefixing the widget
names in the runtime data. This causes us to add the prefix twice - once
when using the widget name to generate the control name and once when
adding the control.
Really we shouldn't be prefixing the widget names at all, the matching
code should be handing this as we always know which DAPM context a
widget came from and always display the widget name in terms of a DAPM
context. However, we're quite close to the merge window and that's
relatively invasive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Now we've got multi-component we need to make sure that the DAPM context
(and hence register I/O context) we use to apply the pending updates at
the end of a DAPM sequence is the one we were processing rather than the
one that was used to initate the state change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now we have a register write minimisation code in DAPM we don't need to
worry about the ordering of the enable and disable of the PGA and the
output stage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
McBSP sidetone is needed in telephony applications. McBSP sidetone is a
configurable FIR filter that forms a loopback from McBSP input to output.
This patch enables the McBSP2 sidetone ALSA controls so that it can be used
on Nokia RX-51/N900.
Sidetone feature can be tested with following commands:
(set up codec input and output paths)
# Enable and configure sidetone
amixer -D hw:0 set 'McBSP2 Sidetone' on
amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767
echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps
# Do not loop audio via CPU
arecord -f dat >/dev/null |aplay /dev/zero
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The "ldo" variable was dereferenced after free on the error path.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Symmetric rate configuration can fail if the second stream starting tries
to apply the symmetric constraint before the first stream has got far
enough to pick a rate. Rather than try to enforce a nonsensical rate of
0Hz log a warning and allow the application to carry on. Things might go
wrong later on but the user will know about it and there's unlikely to be
lasting damage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Annoying as the __devinitdata is actually correct.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Remove warnings in ep93xx-i2s.c
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend range of supported sample rates for CS4271 CODEC.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a codec driver for the dfbmcs320 bluetooth module, which is used
on the neo1973 boards.
The patch also modifies the neo1937_wm8753 sound board driver to use the new
driver instead of registering the bluetooth DAI manually.
Previously there was a name mismatch between the bluetooth DAI and the bluetooth
DAI link and the sound card was not instantiated, with this patch the issue is
no longer present and sound support works again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The neo1973(GTA01) and neo1973_gta02(GTA02) have a very similar audio hardware
setup. They both use the same codec with the same routing to the gsm modem and
bluetooth chip. But they do use different AMPs though and there are some minor
differences in the speaker setup.
As a result most of the code of those two drivers is identical.
So from a maintenance point of view it makes sense to merge them into a single
driver. It also reduces the size of kernel images supporting both the GTA01 and
GTA02.
As a side-effect of this merge the GTA01 for example gains support for routing
audio to and from the bluetooth DAI.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using gpio_request_array instead of requesting and setting up each gpio by hand
makes the code more readable and more compact.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch drops the lm4853_{set,get}_state functions and the "Amp State Switch"
control.
Those were noops which existed to maintain alsa state file compatibility. Since
the control names have changed due to internal changes in the ASoC core and
state file compatibility was broken anyway it makes sense to drop them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
