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
Complete the separation of the twl6040 from the twl core since
it is a separate chip, not part of the twl6030 PMIC.
Make the needed Kconfig changes for the depending drivers at the
same time to avoid breaking the kernel build (vibra, ASoC components).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in
sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using
the (negative) return value as array index on the very next line of
code - that's bad.
Catch the negative return value and propagate it to the caller (which
checks for it) and things are a bit more sane :-)
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/codecs/wm9705.c: In function 'ac97_prepare':
sound/soc/codecs/wm9705.c:251: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/ac97.c: In function 'ac97_prepare':
sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function)
sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function)
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Later WM8994 class devices can bypass the FLL from BCLK. Do this
automatically when the FLL input and output frequencies match up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
release_firmware() does its own NULL ptr testing, it's redundant to
also test before calling it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than adding new arguments to regulator_register() every time we
want to add a new bit of dynamic information at runtime change the function
to take these via a struct. By doing this we avoid needing to do further
changes like the recent addition of device tree support which required each
regulator driver to be updated to take an additional parameter.
The regulator_desc which should (mostly) be static data is still passed
separately as most drivers are able to configure this statically at build
time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/codecs/lm49453.c: In function 'lm49453_set_dai_fmt':
sound/soc/codecs/lm49453.c:1189:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1193:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1197:4: warning: overflow in implicit
constant conversion [-Woverflow]
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute.
But current settings didn't care +1 step for mute.
This patch adds it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
In order to support CODEC<->CODEC links remove the assumption that there
is only a single CODEC on a DAI link by removing the use of the CODEC
pointer in the rtd from the CODEC drivers. They are already being passed
their DAI whenever they are passed an rtd and can get the CODEC from
there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following warning during kernel boot:
0-000a: 850 <--> 1600 mV at 1200 mV normal
0-000a: Voltage range but no REGULATOR_CHANGE_VOLTAGE
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle.
Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros.
Add single register macro : SOC_SINGLE_SX_TLV.
Use snd_soc_info_volsw for .info
Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double.
kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros.
The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As manual described, VAG is an internal voltage reference of DAC/ADC,
So enabled it before DAC/ADC up.
One more thing should care about is VAG fully ramped down requires 400ms,
wait it to avoid pop.
Signed-off-by: Zeng Zhaoming <zengzm.kernel@gmail.com>
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/codecs/max98095.c: In function 'max98095_jack_detect_enable':
sound/soc/codecs/max98095.c:2229:14: error: 'struct max98095_priv' has no member named 'jack_detect_delay'
sound/soc/codecs/max98095.c:2230:18: error: 'struct max98095_priv' has no member named 'jack_detect_delay'
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This increases the chances we'll manage to hit a partially configured
state on restart and the power savings are extremely small.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for spi regmap feature to existing da7210
driver.
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>
No current users and it's the last user of MICBIAS_E().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Zeng Zhaoming <zengzm.kernel@gmail.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Rather than trying to work around machine drivers which try to reprogram
the FLL while it is providing SYSCLK just return an error if they try.
This will avoid audio glitches during FLL reconfiguration, or at least
move the introduction of the glitches to the machine driver.
Since disabling the source for an active SYSCLK is not supported in the
first place systems shouldn't be doing this in the first place.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high
resistance to voltage noise. On chip regulator realizes power supply rejection
ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/
ML26124-02GD can deliver stable audio performance without being affected by noise
from the power supply circuit and peripheral components. The chip also includes
a composite video signal output, which can be applied to various portable device
requirements. The ML26124 is realized these functions into very small package
the size is only 2.56mm x 2.46mm therefore can be construct high quality sound
system easily.
ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package.
Signed-off-by: Tomoya MORINAGA <tomoya.rohm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's possible that the regulator enable will fail and if it does we may
as well just give up with trying to bring the rest of the device up and
report the original error.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
