Use the devm_ versions of the regmap and memory allocation functions,
saving some error handling code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Makes sparse happy and avoids polluting the global namespace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
We need to read the real register values
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're trying to remove all usage of the ASoc level cache and I/O code and
for a device like this with a pretty sparse register map the rbtree cache
is a better idea anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the machines uses the gain ramp possibility for HS/HF.
This code path is mostly unused and it does not reduces the pop
noise on the output (it alters it to sound a bit different).
The preferred method to reduce pop noise is to use ABE.
Remove the gain ramp, and related features form the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As good as nothing exciting here; just a few trivial fixes for
various ASoC stuff.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
This patch improves playback quality for few sample rates like 8000 and
11025 Hz.
This also fixes an issue observed during testing of pll slave mode. Due
to the issue, on some rare occasions there was no sound output for first
time playback after system boot, though all subsequent playbacks were
fine. It was mainly because of the sequence in which SRM bit was
enabled.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than invalidating the cached DCS value every time the headphone
gain changes store multiple values, indexed by gain. This allows the
optimisation we get from the cache to take effect more often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is now very standard behaviour for CODECs so shouldn't be device
specific and we shouldn't really be trying to peer into the register
cache from atomic context anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the conversion to module_init_i2c() the original open coded module
exit function was left. Remove it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value
* this patch solves the problem by only working on the 9 bits the
register contains.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Class W can be used for any path where only data from the DAC is routed
to the headphones. Currently we only enable it when the direct DAC to
headphone path is used but it can also be enabled for paths that go via
the output mixer providing the DAC is the only input to the output mixer.
Implement support for this, including updates to the class W status when
the output mixer configuration is changed. This also allows us to enable
the DC servo optimisations for DAC to headphone paths where the output
mixer is used.
In general the direct DAC path is still preferred as this will offer
better performance on most wm_hubs devices but these additional paths
can simplify use case management.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the analogue portions of the checks for class W are the same over
all the devices factor out these checks into wm_hubs and while we're at
it also use wm_hubs_dac_hp_direct() to enable class W optimisations on
more paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The optimisations which we can do with caching the headphone DCS result in
wm_hubs have only been enabled in cases where class W is enabled. However,
there are more use cases which can benefit from the cache, especially with
WM8994 series devices with their more advanced digital routing.
Rather than keying off the class W information from the CODECs have a
check in wm_hubs for a suitable path and use that to determine if we can
deploy our headphone optimisations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a bug discovered during testing of non pll slave mode.
Due to the bug chip was not getting correctly configured and as a result
there was no sound output while playback. After applying this patch,
both pll and non pll modes work fine.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A workaround for an ASUS laptop and a few ASoC changes;
most of the commits are tagged for stable, too.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A workaround for an ASUS laptop and a few ASoC changes; most of the
commits are tagged for stable, too."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Improve sequencing of AIF channel enables
ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
ASoC: fsi: update for dmaengine prep_slave_sg fallout.
ASoC: core: Fix card RTD count for deferred probe.
ASoC: cs42l73: don't use negative array index
ASoC: dapm: Ensure power gets managed for line widgets
If a driver using a custom mic detection callback has provided a table
of mic detection rates via platform data then use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a slightly larger debounce when identifying accessory type and a
slightly smaller one when detecting buttons in response to user feedback
from large scale testing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When we're not actively doing audio we don't need the microphone biases
to be regulated, noise is not important when we are not looking at the
audio signal. Save some power by putting the MICBIAS regulators into
bypass mode when not doing audio.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mostly a one to one converion. On one occasion the patch replaces a
snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps
to keep the conversion simple.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have never really updated that version number and probably never will, so
just remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all advertised rates are available for all sysclk frequencies. Add
additional sysclk based rate constraints.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If FLL bypass is left enabled when we disable the CODEC then the output
clock will be left running which consumes a small amount of additional
current. Only enable bypass when there is an output.
Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ensures a clean startup of the channels, without this change some
use cases could result in issues in a small proportion of cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.
This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,
(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled
This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.
Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
