Reset the codec according to the Audio power-up delay errata for the 88PM8607.
Signed-off-by: Bas Vermeulen <bas.vermeulen@novero.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce a I2S CLK supply so playback and capture can operate independently.
Signed-off-by: Bas Vermeulen <bas.vermeulen@novero.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not used outside this driver so no need to make the symbol global.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver
field) broke generation of a driver name for all ASoC cards relying on the
automatic generation of one. Fix this by using the old default with spaces
replaced by underscores.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
The indentation is getting a little deep. Should be straight code motion,
no functional changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation. This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The driver assumes that control_data points to the drivers i2c_client struct,
but this is no longer the case since the ASoC core has switched to regmap.
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>
Call platform_device_put() instead of platform_device_unregister() if
platform_device_add() fails.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This functionality is now subsumed within the bias management, using the
standard cache management functionality, without assuming the cache type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For digital only paths we need to make sure the bandgap is enabled prior
to starting the FLL which isn't tied into DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
twl6040 supports 5 playback, and 2 capture channels
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only mono audio can be used for vibra (DL4 channel).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset the twl6040_reg array to hold the chip default values.
The only changed values were for the microphone input selection.
Select no input for the microphones in the twl6040_init_chip function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to write to the vio registers at probe time, since most
them either read only, or shared with MFD or not used.
On the other hand it is a good idea to updated the ASICREV register in
the cache at this time.
After power up we need to restore some registers. Clean up the list to
contain only the registers we are going to restore.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoid requesting the audpwron gpio in case of ES1.0
revision.
In the past we requested the gpio, but we did not
free it up, since we made the check for the revision
later. This results later checks for gpio validity to
fail, leaving the gpio reserved (even after the driver
has been removed).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For client driver to use, if they need chip resvision information.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to keep global pointer for the platform
device, since it is only used for dev_* prints, and the
device pointer available within the twl6040 structure.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.
The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.
2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8996 can measure the impedance of accessories connected to the
headphone output. Implement initial support for this, measuring the
left channel impedance when an accessory is detected and using this
to distinguish between a line load and a headphone load.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than managing the bandgap in the bias level control use a supply
widget as we only actually need to enable it for analogue paths, not
fully digital ones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The WM8776 supports a continuous range of sample rates rather than
discrete values and supports a wider range of sample rates on the
playback path than is currently supported. Update the constraints on
the DAIs to reflect this.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The i2c core will clear the clientdata pointer automatically,
we don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ambiguously named variable 'link' is used as a temporary throughout
davinci-pcm -- its presence makes grepping (and groking) the code
difficult.
Replace link with the value of link in almost all sites. The exception
is a couple places where the last-assigned link/chan needs to be
returned by a function -- in these cases, rename to last_link.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current method for machine driver to register with the ASoC core is to
use snd_soc_register_card() instead of creating a "soc-audio" platform device.
In addition we use platform_device_register_simple() to create a platform
device for the codec. This function will handle putting and deleting the
device automatically which simplifies the error handling in the machine
driver.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To get the PCM module loaded automatically by udev et al. we need to add a
proper MODULE_ALIAS.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the ASoC machine driver is now a platform driver we need to register a
matching platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the ASoC machine driver is now a platform driver we need to register a
matching platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the ASoC machine driver is now a platform driver we need to register
a matching platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_pcm_format_width() to determine the sample size, instead of
checking specify sample formats and assuming that those are the only
valid format.
This change adds support for big-endian architectures (which use the _BE
formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE).
[Fixed single letter variable name legibility problem -- broonie]
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to count the timeout down.
Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Fix below build warning:
sound/soc/blackfin/bf5xx-ad73311.c: warning: initialization from incompatible pointer type
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clk_get() returns a pointer to the struct clk or an ERR_PTR().
This patch also use PTR_ERR() for return value.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Freescale SSI audio controller supports "synchronous" and "asynchronous"
modes. In synchronous mode, playback and capture use the same input clock,
so sample rates must be the same during simultaneous playback and capture.
Unfortunately, the code which supports asynchronous mode is just broken in
various ways. In particular, it was constraining sample sizes as well as
the sample rate.
The fix also allows us to simplify the code by eliminating the 'asynchronous',
'playback', and 'capture' variables that were used to keep track of playback
and capture streams.
Unfortunately, it turns out that simulataneous playback and record does not
actually work on the only platform that supports asynchronous mode: the
Freescale P1022DS reference board. If a second stream is started, the SSI
grinds to halt for both streams. This is true even if the P1022 is configured
for synchronous mode, so it's likely a hardware problem that needs to be
worked around.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
