skl_tplg_bind_sinks() takes only the first sink widget. This
breaks in case we have multiple sinks for a module.
So pass source widget to skl_tplg_bind_sinks() and bind for all
sinks by calling this recursively
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should not stop the sink pipe in it's pmd handler for a mixin
module as this module may still be connected to other pipes.
This will be stopped and freed by current implementation on last
connected pipe unbind.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For binding modules we should check if source or destination
module is in UNINT state. We canot bind even if one of them is
in this state.
So update the check from logical AND to logical OR and do not
bind modules for this case
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In unbind modules, the skl_get_queue_index() can return error
if the pin is dynamic and module is not bound yet. So instead
of returning error this check should return success as modules
is not yet bound. This will let the module be bound when connected
pipes are enabled and will bind this as well.
So change the return value to 0
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We check and allocate pipeline resources in one shot. That causes
leaks if module creation fails later as that is not freed.
So split the resource allocation into two, first check if
resources are available and then add the resources upon
successful creation. So two new functions are added for checking
and current functions are re-purposed to only add the resources
for memory and MCPS.
Signed-off-by: Dharageswari.R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While cleaning resources on module pmd event, we check for return
of skl_unbind_modules(). On failure this causes leak as all modules
attached do not have resources freed.
So ignore return value of module unbind and continue freeing
resources. This makes dapm state and resources correct.
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When delay reported by HW is equal to buffersize, it means the
value is wrapped so we should report as 0. So add the condition
to check this while reporting the delay from LPIB.
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Signed-off-by: Dharageswari.R <dharageswari.r@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
TLV buffer can be smaller than the module data, so update the
size of data to be copied before doing the copy.
Also TLV header consists of two unsigned ints, this is also taken
into account here and size modified to reflect this
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a106804 ("ASoC: compress: Fix compress device direction check")
added a dependency on the compress-cpu-dai channel_min field
which was removed earlier by commit 77095796
("ASoC: Intel: Atom: clean-up compressed DAI definition")
as part of the baytrail cleanups.
The net result was a regression at probe on all Atom platforms
with no sound card created.
Fix by adding explicit initialization for channel_min to 1
for the compress-cpu-dai.
Reported-by: Tobias Mädel <alsa-devel@tbspace.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If snd_soc_tplg_component_load() fails we just printed an error message
and returned the error code but we missed releasing the firmware.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a status bit on RT5677_PLL1_CTRL2 and RT5677_PLL2_CTRL2.
That's why those registers are set volatile. However, the status
bit is currently not used by codec driver. So, it should be no
problem if we set them non-volatile.
The purpose of setting them non-volatile is to restore the setting
after a syspend/resume cycle.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As we want gpio_chip .get() calls to be able to return negative
error codes and propagate to drivers, we need to go over all
drivers and make sure their return values are clamped to [0,1].
We do this by using the ret = !!(val) design pattern.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
As we want gpio_chip .get() calls to be able to return negative
error codes and propagate to drivers, we need to go over all
drivers and make sure their return values are clamped to [0,1].
We do this by using the ret = !!(val) design pattern.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The stream is created whilst the compressed stream is opened and a
buffer is created when the DSP powers up. It is necessary at a point
once both the DSP has powered up and the the stream has been opened to
connect a stream to a buffer on the DSP. This is done in the trigger
callback as this is after the DSP has been powered and obviously the
stream must be open. Note that whilst the connect is currently trivial
it is expected that this will get more complex when support for multiple
buffers/streams per DSP is added.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add code that locates and initialises the buffer of compressed data on
the DSP if the firmware supported compressed data capture. The buffer
struct (wm_adsp_compr_buf) is kept separate from the stream struct
(wm_adsp_compr) this will allow much easier support of multiple
streams of data from the one DSP in the future, although support for
this will not be added in this patch chain.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow user-space to open a compressed stream, although no data will be
passed yet, as part of this adding the ability to define supported
capabilities per firmware and check these match the stream being opened.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Register a platform driver for the CODEC and add DAIs that will be used
to connect a compressed record path for the voice control functionality.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PLL mode based on 32KHz master clock not supported in
AB silicon so remove support from the driver.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HW can provide 1.6V micbias level as well the existing levels
already provided in the driver. This patch adds support for 1.6V
to the DT binding.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In AB silicon, the internal LDO is not supported so remove
DT and driver references to this (digital voltage direct from
'VDD' supply)
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current AB silicon, BIAS_EN field is enabled by default in the
REFERENCES register, so the regmap default value should reflect
this.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If codec probe() function fails after supplies have been enabled
it should really tidy up and disable them again. This patch updates
the probe function to do just that.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously Sidetone would operate only when capture to DAI was in
progress, due to DAPM path configuration. There is no reason why
this should not operate without DAI capture, so this patch updates
the DAPM path accordingly.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture")
in AC'97 mode so in this case fsl-asoc-card route map should
also be using them.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support.
According referance menual, "Limited support for the case when
output sampling rates is between 8kHz and 30kHz. The limitation
is the supported ratio (Fsin/Fsout) range as between 1/24 to 8."
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sometimes the audio play can not be resumed after it is
suspended. Add snd_soc_pm_ops to execute power management
operations, then this issue is fixed.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
identifier fld;
@@
* x = devm_kzalloc(...);
... when != x == NULL
x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
