Recent addition of SoundWire stream state-machine checks in linux-next
have shown an existing issue with handling soundwire streams in codec drivers.
In general soundwire stream prepare/enable/disable can be called from either
codec/machine/controller driver. However calling it in codec driver means
that if multiple instances(Left/Right speakers) of the same codec is
connected to the same stream then it will endup calling stream
prepare/enable/disable more than once. This will mess up the stream
state-machine checks in the soundwire core.
Moving this stream handling to machine driver would fix this issue
and also allow board/platform specfic power sequencing.
Changes since v1:
- removed false error check while setting sruntime.
Srinivas Kandagatla (2):
ASoC: qcom: sdm845: handle soundwire stream
ASoC: codecs: wsa881x: remove soundwire stream handling
sound/soc/codecs/wsa881x.c | 44 +------------------------
sound/soc/qcom/Kconfig | 2 +-
sound/soc/qcom/sdm845.c | 67 ++++++++++++++++++++++++++++++++++++++
3 files changed, 69 insertions(+), 44 deletions(-)
--
2.21.0
There could be multiple instances of this codec on any platform,
so handling stream directly in this codec driver can lead to
multiple calls to prepare/enable/disable on the same SoundWire stream.
Move this stream handling to machine driver to fix this issue.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
snprintf() is a hard-to-use function, it's especially difficult to use
it for concatenating substrings in a buffer with a limited size.
Since snprintf() returns the would-be-output size, not the actual
size, the subsequent use of snprintf() may point to the incorrect
position.
Use scnprintf() instead for fixing such potential errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200313130334.9028-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Gcc reports the following warnings:
sound/soc/codecs/rt5682-sdw.c:286:12: warning: 'rt5682_dev_resume'
defined but not used [-Wunused-function]
static int rt5682_dev_resume(struct device *dev)
^~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5682-sdw.c:273:12: warning: 'rt5682_dev_suspend'
defined but not used [-Wunused-function]
static int rt5682_dev_suspend(struct device *dev)
^~~~~~~~~~~~~~~~~~
Fix by adding maybe_unused as done for other SoundWire codecs
Fixes: 03f6fc6de9 ('ASoC: rt5682: Add the soundwire support')
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200310163509.14466-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt5682 code can be used in I2C or SoundWire mode. When I2C is not
selected, we have the following issue:
WARNING: unmet direct dependencies detected for SND_SOC_RT5682
Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] &&
I2C [=n]
Selected by [m]:
- SND_SOC_RT5682_SDW [=m] && SOUND [=m] && !UML && SND [=m] &&
SND_SOC [=m] && SOUNDWIRE [=m]
Fix by adding SOUNDWIRE as a dependency.
Fixes: 03f6fc6de9 ('ASoC: rt5682: Add the soundwire support')
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200310163509.14466-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
More fixes that have arrived since the merge window, spread out all
over. There's a few things like the operation callback addition for
rt1015 and the meson reset addition which add small new bits of
functionality to fix non-working systems, they're all very small and for
parts of newly added functionality.
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Merge tag 'asoc-fix-v5.6-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.6
More fixes that have arrived since the merge window, spread out all
over. There's a few things like the operation callback addition for
rt1015 and the meson reset addition which add small new bits of
functionality to fix non-working systems, they're all very small and for
parts of newly added functionality.
slimbus addresses are 16 bit wide, masking page numbers
to wcd register at offset of 12 will limit the number for pages.
So it becomes impossible to write to page 0x10 registers.
Remove masking 0x800 (slimbus address range) from register address
and making use of window parameters in regmap config should fix it
and also will represent the registers exactly inline with Datasheet.
Remove this unnessary masking and make the registers be inline
with datasheet.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200306152633.25836-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like there are some unused headers, remove them.
Seems to be missed while moving to mfd.
Reported-by: Stephen Boyd <swboyd@chromium.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200306132806.19684-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the range verification check for the mic_bias and vref device tree
entries.
Fixes 37bde5acf040 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200304193427.16886-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_driver should set ops in rt1015_dai driver.
Also make the two variable static to fix sparse warnings.
Fixes: df31007400 ("ASoC: rt1015: add rt1015 amplifier driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200302070522.48104-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix sparse warning:
sound/soc/codecs/rt5682-sdw.c:163:5: warning:
symbol 'rt5682_clock_config' was not declared. Should it be static?
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200228075609.38236-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The TAS2563 is register compatible with the TAS2562. The main
difference is the TAS2563 has a programmable DSP to manage different
audio profiles.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200226130305.12043-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On i386 randconfig:
sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume':
wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset'
sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume':
wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset'
sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume':
wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset'
Fix this by adding the missing dependencies on SND_SOC_AC97_BUS.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200224112537.14483-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add decimation filter selection support.
Per Section 8.3.6.7 the Digital Decimation Filter is selectable between
a Linear Phase, Low Latency, and Ultra Low Latency filer.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200221181358.22526-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined
in Section 8.3.2 of the data sheets.
The DRE achieves a complete-channel dynamic range as high as 120 dB.
At a system level, the DRE scheme enables far-field, high-fidelity recording
of audio signals in very quiet environments and low-distortion recording in
loud environments.
There are 2 enables for DRE. The first is a global setting that enables
the DRE engine in the device and the other enable is per channel. If
the DRE is enabled globally then either DRE or AGC can be used per each
configured channel. If global DRE is disabled then even setting the DRE
enable bit in the channel config register will have no effect.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200221181358.22526-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for digital volume control. There is no dedicated register
for volume control but instead there are 4. The values of the registers
are determined with exponential floating point math.
So a table was created with register values for 2dB step increments
from -110dB to 0dB.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200221124151.8774-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the tlv320adcx140 codec driver family.
The TLV320ADCx140 is a Burr-Brown™ highperformance, audio analog-to-digital
converter (ADC) that supports simultaneous sampling of up to four analog
channels or eight digital channels for the pulse density modulation (PDM)
microphone input. The device supports line and microphone inputs, and
allows for both single-ended and differential input configurations.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200220210759.31466-3-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When we get a clock error during probe we have to call
regulator_bulk_disable before bailing out, otherwise we trigger
a warning in regulator_put.
Fix this by using "goto err" like in the error cases above.
Fixes: 5a3af12931 ("ASoC: pcm512x: Add PCM512x driver")
Signed-off-by: Matthias Reichl <hias@horus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200220202956.29233-1-hias@horus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The RK3328 reference design uses an external line driver IC as a buffer
on the analog codec output, enabled by the GPIO_MUTE pin, and such a
configuration is currently assumed in the codec driver's direct poking
of GRF_SOC_CON10 to control the GPIO_MUTE output value. However, some
boards wire up analog audio yet use that pin for some other purpose, so
that assumption doesn't always hold. Update this functionality to rely
on an explicit GPIO descriptor, such that it can be managed at the
board level.
Signed-off-by: Robin Murphy <robin.murphy@arm.com>
Link: https://lore.kernel.org/r/5bc383ed1832f0f5d1dcb3c97ad92fd68e5217e3.1581376744.git.robin.murphy@arm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add additional support for ISENSE and VSENSE feature for the TAS2562.
This feature monitors the output to the loud speaker attempts to
eliminate IR drop errors due to packaging.
This feature is defined in Section 8.4.5 IV Sense of the data sheet.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200219134622.22066-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAI has playback_active and capture_active to care usage count.
OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.
But because of this kind of implementation mismatch,
ALSA SoC has many verbose code.
To solve this issue, this patch merge playback_active/capture_active
into stream_active[2];
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/875zg5botu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a need to use RT5682 as DAI clock master for other codecs
within a platform, which means that the DAI clocks are required to
remain, regardless of whether the RT5682 is actually running
playback/capture.
The RT5682 CCF basic functions are implemented almost by the existing
internal functions and asoc apis. It needs a clk provider (rt5682 mclk)
to generate the bclk and wclk outputs.
The RT5682 CCF supports and restricts as below:
1. Fmt of DAI-AIF1 must be configured to master before using CCF.
2. Only accept a 48MHz clk as the clk provider.
3. Only provide a 48kHz wclk and a set of multiples of wclk as bclk.
There are some temporary limitations in this patch until a better
implementation.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1582033912-6841-1-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If the bitwidth passed in to the set_bitwidth function is not supported
then return an error.
Fixes: 29b74236bd57 ("ASoC: tas2562: Introduce the TAS2562 amplifier")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200218185252.26290-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few fixes sent in since the merge window, none of them with global
impact but all important for the users they affect.
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Merge tag 'asoc-fix-v5.6-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.6
A few fixes sent in since the merge window, none of them with global
impact but all important for the users they affect.
Reverts commit 62d5ae4caf ("ASoC: max98090: save and restore
SHDN when changing sensitive registers").
A critical side-effect was observed: when keep playing something,
the recorded sound has chance to break (clipping).
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200214105744.82258-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
