Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.
The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.
The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the data sheet data is clocked out on the falling edge
and latched on the rising edge of the bit clock. While the left sample
is transmitted the word clock line is low.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4535_remove() from sound/soc/codecs/ak4535.c calls
i2c_unregister_device() with a possibly null pointer.
This bug was found by smatch (http://repo.or.cz/w/smatch.git/).
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds machine specific code for the audio part of the Stretch
s6105 IP camera reference design.
The device uses the tlv320aic31(01) codec to generate the clock for
both I2S ports of the soc. While the master clock is generated by a
configurable PLL chip, the code assumes the factory default settings.
An additional kcontrol has been added to handle the special routing of
the board, connecting both HPLCOM and HPROUT to the same pin of the audio
jack. One of these should always be switched off.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a driver for the I2S interface found on Stretch s6000
family processors.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds the needed code to be able to use 96KHz playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this the WM9705 driver fails badly when resuming.
Tested-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that any AC97 devices that bind to the CODEC are below the
ASoC device in the device tree so the suspend and resume code can
figure out what order to handle them in.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AC97 devices may have other drivers hanging off them directly so need to
have resumed when the resume function returns meaning that we can't defer
the resume - complete it immediately for them. Non-AC97 devices should
not have other drivers hanging directly off the ASoC devices.
We only really need the deferral for non-AC97 devices - it's there since
some I2C buses are very slow and non-AC97 codecs often have large numbers
of registers to restore and require delays to bring the codec up cleanly
leading to a substantial impact on overall resume time.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During
initial playback startup, this FIFO is keeping the DMA request active
until the FIFO is full.
So now if ALSA buffer size is smaller, DMA is looping around it while
filling up the HW FIFO, generating burst of interrupts as well and SW
doesn't have any change to fill enough data.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 supports 96KHz sample playback, but only playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Optimize the display of SSI statistics in the Freescale MPC8610 sound driver
to display the status count only of the interrupts that were actually enabled.
Previously, it would display the counts of all SISR status bits, even those
that were not enabled.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the delay from the trigger function in the Freescale MPC8610 sound
driver when capture is started. This delay was used to ensure that the DMA
controller was active when ALSA call the .pointer function to request a
DMA transfer status. A better approach is for the .pointer function to detect
that DMA has not started, and return zero instead. This change eliminates
the need for the delay.
Also add some related code to check for a DMA programming error, and report
XRUN if it occurs.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
HTC Magician has a Philips UDA1380 codec connected via
SSP1 (playback) and I2S (capture).
There is a flip-flop between the SSP frame clock output
and the codec's word select input pin. To make the codec
see proper I2S input, the SSP has to send two frames per
sample.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now magician and similar boards can use network mode with only one
active slot to explicitly set 16 bit frame width, even for S16_LE
stereo sound.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.
Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:
- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.
Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit 14fa43f53f ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister(). This patch adds same condition for
soc_ac97_dev_unregister(). Without this fix, kernel crashes when
unloading an asoc driver.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark 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: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Two issues are fixed here:
- I2S transmits the left frame with the clock low but I don't seem to
get LRCLK out without SFRMDLY being set so invert SFRMP and set a
delay.
- I2S has a clock cycle prior to the first data byte in each channel
so we need to delay the data by one cycle.
Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.
Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application. This driver supports the
primary audio CODEC features, including:
- 1W speaker driver
- Fully differential headphone output
- Up to 4 differential microphone inputs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Buildfix:
CC sound/soc/omap/osk5912.o
sound/soc/omap/osk5912.c: In function 'osk_soc_init':
sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
make[3]: *** [sound/soc/omap/osk5912.o] Error 1
There's no such (standard) clock interface.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
