| Commit message (Collapse) | Author | Age | Files | Lines |
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The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.
One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.
This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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With some buses the transfers may DMAed, especially for larger blocks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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I've removed several unreachable returns.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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idle_bias_off
Just found some cases that some codec drivers set the bias to _STANDBY and
set idle_bias_off to 1 during probing.
It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested,
there is no reason to start from _STANDBY bias with idle_bias_off == 1.
So here giving one warning when detected (dapm.idle_bias_off == 1) and
(dapm.bias_level != SND_SOC_BIAS_OFF) just after driver->probe().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
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Otherwise capture activity on a compressed DAI would mute any playback
on the same DAI.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
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The ASoC compressed API did not implement the copy callback in its
compressed ops which is required for DSPs that are not memory mapped.
This patch creates a local copy of the compress ops for each runtime and
modifies them with a copy callback as appropriate.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Delayed work was scheduled but not initialised, this patch adds the
actual work and initialises it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Use the pcm_mutex to serialise the compressed ops.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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disable clock and correct sequence when unload
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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correct the sequence when unload this module
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ssc private data has been set in ssc driver, this cause the error
private data set to ssc, remove it
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When build as module, it reports following error, using this patch
fix it
sound/soc/atmel/atmel-pcm-pdc.c:387: error: redefinition of 'atmel_pcm_pdc_platform_register'
sound/soc/atmel/atmel-pcm.h:95: note: previous definition of 'atmel_pcm_pdc_platform_register' was here
sound/soc/atmel/atmel-pcm-pdc.c:393: error: redefinition of 'atmel_pcm_pdc_platform_unregister'
sound/soc/atmel/atmel-pcm.h:99: note: previous definition of 'atmel_pcm_pdc_platform_unregister' was here
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Not very useful on non AT91/AVR32 platforms but it provides
more build coverage and prepares for ARM multiplatform.
Also fixes a couple of format type warnings.
Signed-off-by: Joachim Eastwood <manabian@gmail.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Place a cap on the number of channels clocks are generated for. This is
intended for use with systems which have the WM5102 master an I2S bus with
multiple data lines.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Some systems may wish to support switching between telephony and CD audio
clock rates but this is restricted by enforcement of constraints on the
current DAI clock. Support setting clocks to zero and don't enforce any
constraints in that case in order to facilitate this use case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Conflicts:
sound/soc/codecs/arizona.c
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We only log the result and since the interrupt triggers on loss of lock
during shutdown this may lead to spurious interrupts during shutdown
delaying the process.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In preparation for more advanced sample rate managment move the existing
code out of the main hw_params() function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-arizona
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Some systems use external analogue switches to connect more analogue
devices to the CODEC than are supported by the device. In some systems
this requires changing the switched output from single ended to
differential mode dynamically at runtime. Add a new function
arizona_set_output_mode() to support this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The references used for the noise gates and parameters for their triggering
are configurable, expose that to users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The references used for the noise gates and parameters for their triggering
are configurable, expose that to users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Though the controls are named as stereo controls in the part the main use
case for the analogue inputs to the WM5102 is mono. Reflect this in the
controls exposed to userspace, providing a series of mono controls rather
than stereo ones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Though the controls are named as stereo controls in the part the main use
case for the analogue inputs to the WM5102 is mono. Reflect this in the
controls exposed to userspace, providing a series of mono controls rather
than stereo ones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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CONFIG_HOTPLUG is always true now and the __dev* macros have been removed.
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Support for loading the ak4642 codec module via devicetree.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Allow the regmap API to use asynchronous I/O where supported to minimise
the delay between transfers, reducing firmware download times.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Conflicts:
sound/soc/codecs/wm_adsp.c
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Helpful for debugging.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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