When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex
already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By
moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is
prevented. This problem was introduced in Linux 3.5
Signed-off-by: Andreas Irestål <Andreas.Irestal@axis.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Convert MicBias widgets to supply widget.
On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage. So, when power on mic bias, we need
reclaim it to voltage value.
Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg" platform data.
Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.
Since micbias is converted to supply widget, updated machine drivers as
well.
This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Playing a mono stream on the TLV320AIC3x results in too fast playback rate.
Remove mono support so that mono streams can be played correctly on this codec.
Tested with imx-ssi (i.MX25) and TLV320AIC3104.
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Device tree support for tlv320aic3x CODEC driver.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_i2c_driver makes the code simpler by eliminating module_init
and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
This patch adds input selection of main codec clock - from what pin.
Both registers set same value since codec uses clock divider or pll at one time.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds AGC target level and times settings for TLV320AIC3x.
Enums uses small arrays of two channels left and right since it uses different registers.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds missing deemphasis switch.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds missing register default values to cache.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In sound/soc/codecs/tlv320aic3x.c
data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
data | (pll_p << PLLP_SHIFT));
In the above code, pll-p value is OR'ed with previous value without
clearing it. Bug is not seen if pll-p value doesn't change across
Sampling frequency.
However on some platforms (like AM335x EVM-SK), pll-p may have different
values across different sampling frequencies. In such case, above code
configures the pll with a wrong value.
Because of this bug, when a audio stream is played with pll value
different from previous stream, audio is heard as differently(like its
stretched).
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output
gain control as a playback control. Fix this by changing the name as
"Class-D Playback Volume".
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
aic3x_set_headset_detection() isn't made available outside the driver or
referenced within the driver which sparse notices and complains about.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Now that we're not implementing GPIO or microphone detection support
there's no users for the read() function any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Looks like nobody is or will be using this code.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.
This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-
snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()
This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().
It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().
Finally, it updates the existing machine drivers that register controls to either :-
1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.
In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC
drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver
for devices can unconditionally support idle_bias_off.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
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>
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.
The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver only supports I2C so doesn't need to do things conditionally.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
According to the datasheet:
Page0 / Register8: Audio Serial Data interface Control Register A
BIT 7: Bit Clock Directional Control
0: Bit clock is an input (slave mode)
1: Bit clock is an output (master mode)
BIT 6: Word Clock Directional Control
0: Word clock is an input (slave mode)
1: Word clock is an output (master mode)
Current code sets BIT_CLK_MASTER and WORD_CLK_MASTER bits for master mode,
but does not clear these bits for slave mode.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write
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>
codec->hw_read is broken now, let's covert to snd_soc_read.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We really should be doing this in the core, not in a driver...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Save model information in driver_data so we can simplify the implementation.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit af46800 ("ASoC: Implement mux control sharing") revealed that
"Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing
to the same kcontrols and codec registers and thus soc-core falsely detected
them as shared controls. This is actually wrong since there are separate
registers in hardware that configure Line1L to RADC and Line1R to LADC cross
connects so these muxes should not be shared.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
TLV320AIC33, TLV320AIC34 and I believe others too in this family have some
hw bugs that cause that analogue and digital VDD supplies remain leaking
up to a few mA of current after certain use cases even the hw blocks inside
codec are driven to off.
Highest leakages occur after using the bypass paths inside codec but it
is possible to get smaller leakages just by toggling mute switches in
unused audio paths (i.e. no DAPM changes) while codec is on due another
active audio path.
While some cases are able to workaroud by making sure that e.g. output mixer
switches are muted before powering down the output stage this doesn't help
all the cases.
Therefore use the software reset command to clear possible leakage currents
since that works in every cases and affects only this codec instance. Only
drawback is that now cache sync is required everytime when codec bias comes
out from bias off state, not only when supply regulators were off.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
There is no need to sync first two registers from cache to hw after a reset.
First one is used to select page for register access and this driver is
normally accessing page 0 only. Second one does a software reset which is
obviously unneeded after hardware or previous software reset command.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol")
changed the control private data type that is passed to snd_soc_cnew when
creating dapm mixer and mux controls. Commit did not update a few codec
drivers that are using their own put callbacks and thus are accessing a
wrong data type.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
If aic3x_read failed , val is used uninitialized.
Fix it by initializing val to 0.
This patch fixes below compile warning:
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_get_gpio':
sound/soc/codecs/tlv320aic3x.c:1183: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_headset_detected':
sound/soc/codecs/tlv320aic3x.c:1211: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_button_pressed':
sound/soc/codecs/tlv320aic3x.c:1219: warning: 'val' may be used uninitialized in this function
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to include soc-dapm.h since soc.h includes it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We allocated memory for aic3x in aic3x_i2c_probe,
and will free the memory in either aic3x_i2c_probe error path or
aic3x_i2c_remove.
Thus we should not call kfree(aic3x) in aic3x_probe, otherwise
we have double free of aic3x.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is aimed to configurations where multiple aic3x codecs share the same
reset line and are powered from same supply voltages.
Currently aic3x_probe will fail if trying to request already requested
gpio_reset and passing -1 to another aic3x instances cause that those
instances cannot release reset in aic3x_set_power. That is, another
instances can work only if primary aic3x instance is powered and reset is
released.
Solve this by implementing a list of probed instances that is used for
checking if other instance shares the same gpio_reset number. If a shared
reset line exists, then only first instance tries to request and configure
it and the last instance releases it.
Runtime modifications are not needed since aic3x_regulator_event with help
of regulator framework takes already care that reset is pulled down only
when some or all supplies are disabled meaning that all instances using them
are idle.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
I promised to convert this at some point.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>