Each ASM session can have multiple streams attached to it,
current design was to allow only one static stream id 1 per each session.
However for use-case like gapless, we would need 2 streams to open per session.
This patch converts all the q6asm apis to take stream id as argument
to allow multiple streams to open on a single session, This is useful
for gapless playback cases.
Now the dai driver can specify which stream id for each command.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Each q6asm session can have multiple streams, mixing usage of these
names in variable are bit misleading to reader, so rename them accordingly.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable I2S TDM audio capture for Intel Keem Bay platform.
The I2S TDM will support 4 channel and 8 channel audio capture only.
4 channel and 8 channel audio capture operates only in slave mode.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200811041836.999-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Moving GPIO reset to a later stage and before clock registration to
ensure that the host system and codec clocks are in sync. If the host
register clock values prior to gpio reset, the last configured codec clock
is registered to the host. The codec then gets gpio resetted setting the
codec clocks to their default value, causing a mismatch. Host system will
skip clock setting thinking the codec clocks are already at the requested
rate.
ADC reset is added to ensure the next audio capture does not have
undesired artifacts. It is probably related to the original code
where the probe function resets the ADC prior to 1st record.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200812094631.4698-4-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Increased maximum supported channel to 8 channels for audio capture
running in TDM mode.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200812094631.4698-3-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable 24 bit in 32 bit container audio support.
Using the params_physical_width to differentiate
24 bit in 32 bit container and 24 bit in 24 bit container modes.
Use the sample rate, bit depth and channel parameters to
calculate the bit clock needed.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200812094631.4698-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
By including the earpiece mute switch in the DAPM graph, both the
earpiece amplifier and the Mixer/DAC inputs can be powered off when
the earpiece is muted.
While the widget is really just a simple switch, it is represented
as a "mixer with named controls" to avoid including the widget name
in the kcontrol name. Otherwise, it is not possible to give the widget
an accurate, descriptive name without changing the kcontrol name
seen by userspace (which should be stable).
The mute switch is between the source selection and the amplifier,
as per the diagram in the SoC manual.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726025334.59931-9-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
By including the line out mute switch in the DAPM graph, the
Mixer/DAC inputs can be powered off when the line output is muted.
The line outputs have an unusual routing scheme. The left side mute
switch is between the source selection and the amplifier, as usual.
The right side source selection comes *after* its amplifier (and
after the left side amplifier), and its mute switch controls
whichever source is currently selected. This matches the diagram in
the SoC manual.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-8-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This matches the hardware more accurately, and is necessary for
including the (stereo) line out mute switch in the DAPM graph.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-7-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
By including the headphone mute switch to the DAPM graph, both the
headphone amplifier and the Mixer/DAC inputs can be powered off when
the headphones are muted.
The mute switch is between the source selection and the amplifier,
as per the diagram in the SoC manual.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-6-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This matches the hardware more accurately, and is necessary for
including the (stereo) headphone mute switch in the DAPM graph.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-5-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Sort the controls in the same order as the bits in the register. Then
group the routes by sink, and sort them in the same order as the
controls. This makes it much easier to verify that all mixer inputs are
accounted for.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The clock must be running for the zero-crossing mute functionality.
However, it must be gated for VDD-SYS to be turned off during system
suspend. Disable it in the suspend callback, after everything has
already been muted, to avoid pops when muting/unmuting outputs.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-3-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The same enable bits are currently used for both the "Left/Right ADC"
and the "Left/Right ADC Mixer" widgets. This happens to work in practice
because the widgets are always enabled/disabled at the same time, but
each register bit should only be associated with a single widget.
To keep symmetry with the DAC widgets, keep the bits on the ADC widgets,
and remove them from the ADC Mixer widgets.
Fixes: 42371f327d ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls")
Reported-by: Ondrej Jirman <megous@megous.com>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-2-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warnings:
sound/soc/intel/boards/bdw-rt5650.c:91:23: style: Local variable
'channels' shadows outer variable [shadowVariable]
sound/soc/intel/boards/bdw-rt5677.c:144:23: style: Local variable
'channels' shadows outer variable [shadowVariable]
sound/soc/intel/boards/broadwell.c:91:23: style: Local variable
'channels' shadows outer variable [shadowVariable]
This was fixed earlier in other machine drivers but keeps coming back
with copy/paste.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813175839.59422-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cppcheck reports the following warning:
sound/soc/sof/intel/hda-codec.c:191:1: style: Label 'error' is not
used. [unusedLabel]
This label is indeed only used conditionally, move it where it's
actually used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813175839.59422-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On the A64, as tested using the PinePhone, the current code causes the
left/right channels to be swapped during I2S playback from the CPU on
AIF1, and breaks DSP_A communication with the modem on AIF2. Both of
these are fixed when LRCK is no longer inverted.
Trusting that the comment in the code is correct, the existing behavior
is kept for the A33.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726012557.38282-5-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The sun8i-codec driver provides ALSA controls for enabling/disabling
each of the inputs to the AIF1 Slot 0 and DAC mixers. For two of these
inputs (ADC->DAC and AIF1 DA0->AIF1 AD0), the audio source is
implemented, so the mixer inputs can be used.
However, because the DAPM routes are missing, these mixer inputs only
work when both the source and the mixer happen to be part of other
active audio paths. Adding the appropriate routes makes these ALSA
controls function all of the time.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726012557.38282-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The A33/A64 digital codec has 4 physical inputs and 4 physical outputs:
3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by
a 4-channel mixer connected to each output.
The analog and digital sides of the ADC/DAC are in separate ASoC
components, so card-level DAPM routes (provided in the device tree) are
necessary to connect them together. Currently, these routes are wrong.
For AIF1 Playback, the correct topology is:
||<<============ sun8i-codec ===========>>||
|| ||
CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog)
|| ||
but the driver and device trees currently describe:
|| ||
CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog)
|| \--> DAC Mixer -> ??? [dead end] ||
For AIF1 Capture, there is an additional problem, because the Mixer
route is backward. The topology should be:
|| ||
ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI
|| ||
but the driver and device trees currently describe:
|| ||
ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI
|| \--> ADC Mixer -> ??? [dead end] ||
The ADC/DAC are only powered because AIF1 AD0 (capture) has supply
routes from the ADC, and AIF1 DA0 (playback) has supply routes from the
DAC. However, neither set of supply routes matches the hardware
topology. Audio can be routed among AIF1/2/3 without using the ADC or
DAC at all; and audio can be routed from the ADC to the DAC without
using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the
DAPM routes are wrong, both of these use cases are currently broken.
This commit adds the necessary widgets and routes to represent the real
hardware topology, with functionality equivalent to the current driver.
For the existing "allwinner,sun8i-a33-codec" compatible, widgets with
the old names are kept as wrappers around the new widgets, so existing
device trees will continue to work. For "allwinner,sun50i-a64-codec",
the old widgets can be omitted, because no device trees yet use that
compatible.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
As new function fsl_sai_dir_is_synced is included for checking if
stream is synced by the opposite stream, then replace the existing
synchronous checking with this new function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-4-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Tx synchronous with Rx: The RMR is the word mask register, it is used
to mask any word in the frame, it is not relating to clock generation,
So it is no need to be changed when Tx is going to be enabled.
Rx synchronous with Tx: The TMR is the word mask register, it is used
to mask any word in the frame, it is not relating to clock generation,
So it is no need to be changed when Rx is going to be enabled.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-3-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code enables TCSR.TE and RCSR.RE together, and disable
TCSR.TE and RCSR.RE together in trigger(), which only supports
one operation mode：
1. Rx synchronous with Tx: TE is last enabled and first disabled
Other operation mode need to be considered also:
2. Tx synchronous with Rx: RE is last enabled and first disabled.
3. Asynchronous mode: Tx and Rx are independent.
So the enable TCSR.TE and RCSR.RE sequence and the disable
sequence need to be refined accordingly for #2 and #3.
There is slightly against what RM recommennds with this change.
For example in Rx synchronous with Tx mode, case "aplay 1.wav;
arecord 2.wav" enable TE before RE. But it should be safe to
do so, judging by years of testing results.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-2-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
platform_get_irq_byname() is used when there is list
of interrupts in the device node. As lpass-platform
has only one interrupt entry, use platform_get_irq()
instead.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Link: https://lore.kernel.org/r/1597402388-14112-12-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
platform_get_resource_byname() is used when there
is list of reg entries. As lpass-cpu node has only
one reg entry, use platform_get_resource() instead.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-11-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
i2sctl register value is set to 0 during hw_free(). This
impacts any ongoing concurrent session on the same i2s
port. As trigger() stop already resets enable bit to 0,
there is no need of explicit hw_free. Removing it to
fix the issue.
Fixes: 80beab8e1d ("ASoC: qcom: Add LPASS CPU DAI driver")
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-7-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
I2SCTL and DMACTL registers has different bits alignment for newer
LPASS variants of SC7180 soc. Use REG_FIELD_ID() to define the
reg_fields in platform specific file and removed shifts and mask
macros for such registers from header file.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Link: https://lore.kernel.org/r/1597402388-14112-6-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
lpass_pcm_data is never freed. Free it in close
ops to avoid memory leak.
Fixes: 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data usage")
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-5-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We are allocating dma memory for component->dev but trying to mmap
such memory for substream->pcm->card->dev. Replace device argument
in mmap with component->dev to fix this.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-4-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Ahbix clock is optional clock and not needed for all platforms.
Move it to lpass-apq8016/ipq806x as it is not needed for sc7180.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-3-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS variants have their own soc specific clocks that needs to be
enabled for MI2S audio support. Added a common variable in drvdata to
initialize such clocks using bulk clk api. Such clock names is
defined in variants specific data and needs to fetched during init.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-2-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The (new?) style of clk registration uses clk_hw based APIs so that we
can more easily see the difference between clk providers and clk
consumers. Use the clk_hw based APIs to do this and migrate to devm for
the clkdev creation so that we can reduce the amount of code.
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200804000531.920688-4-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The __clk_get_name() API is deprecated. Use clk_hw_get_name() or
proper registration techniques to avoid it.
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200804000531.920688-3-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
I see a spew of "sysclk/dai not set correctly" whenever I cat
/sys/kernel/debug/clk/clk_summary on my device. This is because the
master pointer isn't set yet in this driver. A user isn't going to be
able to do much if this check is failing so this error message isn't
really an error, it's more of a kernel debug message. Lower the priority
to dev_dbg() so that it isn't so noisy.
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200804000531.920688-2-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When power_up_sst() fails, stream needs to be freed
just like when try_module_get() fails. However, current
code is returning directly and ends up leaking memory.
Fixes: 0121327c1a ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver supports WM1811, WM8994, WM8958 devices but according to
documentation and the regmap definitions the WM8958_DSP2_* registers
are only available on WM8958. In current code these registers are
being accessed as if they were available on all the three chips.
When starting playback on WM1811 CODEC multiple errors like:
"wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5"
can be seen, which is caused by attempts to read an unavailable
WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent
commit "e2329ee ASoC: soc-component: add soc_component_err()".
This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling
is only done for WM8958.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason interrupt set and clear register offsets are
not set correctly.
This patch corrects them!
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200811103452.20448-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC2 and DAC2 are not available on WM1811 device. This patch moves
the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing
them and avoid unreadable register access on WM1811.
This allows to get rid of warnings during boot like:
wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20200804141043.11425-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the DAPM widgets for DSP ASoC components reuse reg field
of the widgets for its internal calculations, however these are not
real registers. So read/writes to these numbers are not really
valid. However ASoC core will read these registers to get default
state during startup.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
To fix this add dummy read/write function to return default value.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like the q6afe-dai dapm widget registers are set as "0",
which is a not correct.
As this registers will be read by ASoC core during startup
which will throw up errors, Fix this by making the registers
as SND_SOC_NOPM as these should be never used.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Along with the recent unification of snd_soc_component_read*()
functions, the behavior of snd_soc_component_read() was changed
slightly; namely it returns the register read value directly, and even
if an error happens, it returns zero (but it prints an error
message). That said, the caller side can't know whether it's an error
or not any longer.
Ideally this shouldn't matter much, but in practice this seems causing
a regression, as John reported. And, grepping the tree revealed that
there are still plenty of callers that do check the error code, so
we'll need to deal with them in anyway.
As a quick band-aid over the regression, this patch changes the return
value of snd_soc_component_read() again to the negative error code.
It can't work, obviously, for 32bit register values, but it should be
enough for the known regressions, so far.
Fixes: cf6e26c71b ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200810134631.19742-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Replacing string compare with "codec_dai->name" instead of comparing with
"codec_dai->component->name" in hw_params because,
Here the component name for codec RT1015 is "i2c-10EC5682:00"
and will never be "rt1015-aif1" as it is codec-dai->name.
So, strcmp() always compares and fails to set the
sysclk,pll,bratio for expected codec-dai="rt1015-aif1".
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200807161046.17932-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
* API cleanups and conversions to the unified mute_stream() call
* Simplify I/O helper functions
* Use helper macros to retrieve RTD from substreams
ASoC drivers:
* Lots of fixes and cleanups in Intel ASoC drivers
* Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
* Minor code refacotring for SG-buffer handling
HD-audio:
* Generalization of mute-LED handling with LED classdev
* Intel silent stream support for HDMI
* Device-specific fixes: CA0132, Loongson-3
Others:
* Usual USB- and HD-audio quirks for various devices
* Fixes for echoaudio DMA position handling
* Various documents and trivial fixes for sparse warnings
* Conversion to adapt inclusive terms
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Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
- API cleanups and conversions to the unified mute_stream() call
- Simplify I/O helper functions
- Use helper macros to retrieve RTD from substreams
ASoC drivers:
- Lots of fixes and cleanups in Intel ASoC drivers
- Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
- Minor code refacotring for SG-buffer handling
HD-audio:
- Generalization of mute-LED handling with LED classdev
- Intel silent stream support for HDMI
- Device-specific fixes: CA0132, Loongson-3
Others:
- Usual USB- and HD-audio quirks for various devices
- Fixes for echoaudio DMA position handling
- Various documents and trivial fixes for sparse warnings
- Conversion to adopt inclusive terms"
* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
ALSA: pci: delete repeated words in comments
ALSA: isa: delete repeated words in comments
ALSA: hda/tegra: Add 100us dma stop delay
ALSA: hda: Add dma stop delay variable
ASoC: hda/tegra: Set buffer alignment to 128 bytes
ALSA: seq: oss: Serialize ioctls
ALSA: hda/hdmi: Add quirk to force connectivity
ALSA: usb-audio: add startech usb audio dock name
ALSA: usb-audio: Add support for Lenovo ThinkStation P620
Revert "ALSA: hda: call runtime_allow() for all hda controllers"
ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
ALSA: docs: fix typo
ALSA: doc: use correct config variable name
ASoC: core: Two step component registration
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
...
Pull crypto updates from Herbert Xu:
"API:
- Add support for allocating transforms on a specific NUMA Node
- Introduce the flag CRYPTO_ALG_ALLOCATES_MEMORY for storage users
Algorithms:
- Drop PMULL based ghash on arm64
- Fixes for building with clang on x86
- Add sha256 helper that does the digest in one go
- Add SP800-56A rev 3 validation checks to dh
Drivers:
- Permit users to specify NUMA node in hisilicon/zip
- Add support for i.MX6 in imx-rngc
- Add sa2ul crypto driver
- Add BA431 hwrng driver
- Add Ingenic JZ4780 and X1000 hwrng driver
- Spread IRQ affinity in inside-secure and marvell/cesa"
* 'linus' of git://git.kernel.org/pub/scm/linux/kernel/git/herbert/crypto-2.6: (157 commits)
crypto: sa2ul - Fix inconsistent IS_ERR and PTR_ERR
hwrng: core - remove redundant initialization of variable ret
crypto: x86/curve25519 - Remove unused carry variables
crypto: ingenic - Add hardware RNG for Ingenic JZ4780 and X1000
dt-bindings: RNG: Add Ingenic RNG bindings.
crypto: caam/qi2 - add module alias
crypto: caam - add more RNG hw error codes
crypto: caam/jr - remove incorrect reference to caam_jr_register()
crypto: caam - silence .setkey in case of bad key length
crypto: caam/qi2 - create ahash shared descriptors only once
crypto: caam/qi2 - fix error reporting for caam_hash_alloc
crypto: caam - remove deadcode on 32-bit platforms
crypto: ccp - use generic power management
crypto: xts - Replace memcpy() invocation with simple assignment
crypto: marvell/cesa - irq balance
crypto: inside-secure - irq balance
crypto: ecc - SP800-56A rev 3 local public key validation
crypto: dh - SP800-56A rev 3 local public key validation
crypto: dh - check validity of Z before export
lib/mpi: Add mpi_sub_ui()
...
The variable rtd was left unused in psc_dma_free(), even unnoticed
during conversion to a new style:
sound/soc/fsl/mpc5200_dma.c:342:30: warning: unused variable 'rtd' [-Wunused-variable]
Drop the superfluous one.
Fixes: 6d1048bc11 ("ASoC: fsl: mpc5200_dma: remove snd_pcm_ops")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803144630.9615-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_i2s.c:167:12: warning: 'tegra210_i2s_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_i2s.c:179:12: warning: 'tegra210_i2s_runtime_resume' defined but not used [-Wunused-function]
Fixes: c0bfa98349 ("ASoC: tegra: Add Tegra210 based I2S driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_dmic.c:43:12: warning: 'tegra210_dmic_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_dmic.c:55:12: warning: 'tegra210_dmic_runtime_resume' defined but not used [-Wunused-function]
Fixes: 8c8ff982e9 ("ASoC: tegra: Add Tegra210 based DMIC driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:579:12: warning: 'tegra_ahub_runtime_resume' defined but not used [-Wunused-function]
Fixes: 16e1bcc2ca ("ASoC: tegra: Add Tegra210 based AHUB driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_admaif.c:232:12: warning: 'tegra_admaif_runtime_resume' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
Fixes: f74028e159 ("ASoC: tegra: Add Tegra210 based ADMAIF driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra186_dspk.c:74:12: warning: 'tegra186_dspk_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra186_dspk.c:86:12: warning: 'tegra186_dspk_runtime_resume' defined but not used [-Wunused-function]
Fixes: 327ef64702 ("ASoC: tegra: Add Tegra186 based DSPK driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Recently we found an issue about the suspend and resume. If dmic is
recording the sound, and we run suspend and resume, after the resume,
the dmic can't work well anymore. we need to close the app and reopen
the app, then the dmic could record the sound again.
For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2
-r 48000 test.wav", then suspend and resume, after the system resume
back, we speak to the dmic. then stop the arecord, use aplay to play
the test.wav, we could hear the sound recorded after resume is weird,
it is not what we speak to the dmic.
I found two registers are set in the dai_hw_params(), if the two
registers are set during the resume, this issue could be fixed.
Move the code of the dai_hw_params() into the pdm_dai_trigger(), then
these two registers will be set during resume since pdm_dai_trigger()
will be called during resume. And delete the empty function
dai_hw_params().
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200730123138.5659-1-hui.wang@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause
audio distortion.
This change was sent as a comment below the --- line when submitting
commit 658bb297e3 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so
it was not supposed to get merged.
Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to
fix the regression.
Fixes: 658bb297e3 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE")
Reported-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20200803115233.19034-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With this case:
aplay -Dhw:x 16khz.wav 24khz.wav
There is sound distortion for 24khz.wav. The reason is that setting
PLL of WM8962 with set_bias_level function, the bias level is not
changed when 24khz.wav is played, then the PLL won't be reset, the
clock is not correct, so distortion happens.
The resolution of this issue is to remove fsl_asoc_card_set_bias_level.
Move PLL configuration to hw_params and hw_free.
After removing fsl_asoc_card_set_bias_level, also test WM8960 case,
it can work.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1596420811-16690-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Provide a mechanism for true two-step component registration. This
mimics device registration flow where initialization is the first step
while addition goes as second in line. Drivers may choose to modify
component's fields before registering component to ASoC subsystem via
snd_soc_add_component.
Patchset achieves status quo - behavior of snd_soc_register_component
remains unchanged.
Cezary Rojewski (3):
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Two step component registration
include/sound/soc-component.h | 3 --
include/sound/soc.h | 11 +++---
sound/soc/soc-component.c | 16 ---------
sound/soc/soc-core.c | 52 +++++++++++++++++----------
sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++---
sound/soc/stm/stm32_adfsdm.c | 9 +++--
6 files changed, 55 insertions(+), 50 deletions(-)
--
2.17.1
Modify snd_soc_add_component so it calls snd_soc_component_initialize
no longer and thus providing true two-step registration. Drivers may
choose to change component's fields before actually adding it to ASoC
subsystem.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move 'name' field initialization responsibility back to
snd_soc_component_initialize to prepare snd_soc_add_component function
for being called separatelly as a second registration step.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To allow for two-step component registration, expose
snd_soc_component_initialize function and move it back to soc-core.c.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enabling a whole subsystem from a single driver 'select' is frowned
upon and won't be accepted in new drivers, that need to use 'depends on'
instead. Existing selection of DMADEVICES will then cause circular
dependencies. Replace them with a dependency.
Signed-off-by: Laurent Pinchart <laurent.pinchart@ideasonboard.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20200731152433.1297-3-laurent.pinchart@ideasonboard.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The fifo_depth is 64 on i.MX8QM/i.MX8QXP, 128 on i.MX8MQ, 16 on
i.MX7ULP.
Original FSL_SAI_CR1_RFW_MASK value 0x1F is not suitable for
these platform, the FIFO watermark mask should be updated
according to the fifo_depth.
Fixes: a860fac420 ("ASoC: fsl_sai: Add support for imx7ulp/imx8mq")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1596176895-28724-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit b73287f0b0 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
changed the meaning of dpcm_playback/dpcm_capture and now requires the
CPU DAI BE to aligned with those flags.
This broke all Amlogic cards with uni-directional backends (All gx and
most axg cards).
While I'm still confused as to how this change is an improvement, those
cards can't remain broken forever. Hopefully, next time an API change is
done like that, all the users will be updated as part of the change, and
not left to fend for themselves.
Fixes: b73287f0b0 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200731120603.2243261-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Previous updates to set dailink capabilities and check dailink
capabilities were based on a flawed assumption that all dais support
the same capabilities as the dailink. This is true for TDM
configurations but existing configurations use an amplifier and a
capture device on the same dailink, and the tests would prevent the
card from probing.
This patch modifies the snd_soc_dai_link_set_capabilities()
helper so that the dpcm_playback (resp. dpcm_capture) dailink
capabilities are set if at least one dai supports playback (resp. capture).
Likewise the checks are modified so that an error is reported only
when dpcm_playback (resp. dpcm_capture) is set but none of the CPU
DAIs support playback (resp. capture).
Fixes: 25612477d2 ('ASoC: soc-dai: set dai_link dpcm_ flags with a helper')
Fixes: b73287f0b0 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Suggested-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200723180533.220312-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The various list iterators are able to handle an empty list.
The only effect of avoiding the loop is not initializing some
index variables.
Drop list_empty tests in cases where these variables are not
used.
The semantic patch that makes these changes is as follows:
(http://coccinelle.lip6.fr/)
<smpl>
@@
expression x,e;
iterator name list_for_each_entry;
statement S;
identifier i;
@@
-if (!(list_empty(x))) {
list_for_each_entry(i,x,...) S
- }
... when != i
? i = e
@@
expression x,e;
iterator name list_for_each_entry_safe;
statement S;
identifier i,j;
@@
-if (!(list_empty(x))) {
list_for_each_entry_safe(i,j,x,...) S
- }
... when != i
when != j
(
i = e;
|
? j = e;
)
@@
expression x,e;
iterator name list_for_each;
statement S;
identifier i;
@@
-if (!(list_empty(x))) {
list_for_each(i,x) S
- }
... when != i
? i = e
@@
expression x,e;
iterator name list_for_each_safe;
statement S;
identifier i,j;
@@
-if (!(list_empty(x))) {
list_for_each_safe(i,j,x) S
- }
... when != i
when != j
(
i = e;
|
? j = e;
)
// -------------------
@@
expression x,e;
statement S;
identifier i;
@@
-if (!(list_empty(x)))
list_for_each_entry(i,x,...) S
... when != i
? i = e
@@
expression x,e;
statement S;
identifier i,j;
@@
-if (!(list_empty(x)))
list_for_each_entry_safe(i,j,x,...) S
... when != i
when != j
(
i = e;
|
? j = e;
)
@@
expression x,e;
statement S;
identifier i;
@@
-if (!(list_empty(x)))
list_for_each(i,x) S
... when != i
? i = e
@@
expression x,e;
statement S;
identifier i,j;
@@
-if (!(list_empty(x)))
list_for_each_safe(i,j,x) S
... when != i
when != j
(
i = e;
|
? j = e;
)
</smpl>
---
drivers/media/pci/saa7134/saa7134-core.c | 14 ++---
drivers/media/usb/cx231xx/cx231xx-core.c | 16 ++----
drivers/media/usb/tm6000/tm6000-core.c | 24 +++-------
drivers/net/ethernet/mellanox/mlx5/core/steering/dr_matcher.c | 13 ++---
drivers/net/ethernet/mellanox/mlx5/core/steering/dr_rule.c | 5 --
drivers/net/ethernet/sfc/ptp.c | 20 +++-----
drivers/net/wireless/ath/dfs_pattern_detector.c | 15 ++----
sound/soc/intel/atom/sst/sst_loader.c | 10 +---
sound/soc/intel/skylake/skl-pcm.c | 8 +--
sound/soc/intel/skylake/skl-topology.c | 5 --
10 files changed, 53 insertions(+), 77 deletions(-)
PulseAudio (and perhaps other userspace utilities) can not detect any
jack for rk3399_gru_sound as the driver doesn't expose related Jack
kcontrols.
This patch adds two DAPM pins to the headset jack, where the
snd_soc_card_jack_new() call automatically creates "Headphones Jack" and
"Headset Mic Jack" kcontrols from them.
With an appropriate ALSA UCM config specifying JackControl fields for
the "Headphones" and "Headset" (mic) devices, PulseAudio can detect
plug/unplug events for both of them after this patch.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Link: https://lore.kernel.org/r/20200721182709.6895-1-alpernebiyasak@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the WM8962 datasheet, there is no register at address 0x200.
WM8962_GPIO_BASE is just a base address for the GPIO registers and not a
real register, so remove it from wm8962_readable_register().
Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip
its access.
This fixes the following errors:
wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16
wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200717135959.19212-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use resource_size rather than a verbose computation on
the end and start fields.
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
<smpl>
@@ struct resource ptr; @@
- (ptr.end - ptr.start + 1)
+ resource_size(&ptr)
</smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Link: https://lore.kernel.org/r/1595751933-4952-1-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
list_for_each_entry_safe is able to handle an empty list.
The only effect of avoiding the loop is not initializing the
index variable.
Drop list_empty tests in cases where these variables are not
used.
Note that list_for_each_entry_safe is defined in terms of
list_first_entry, which indicates that it should not be used on an
empty list. But in list_for_each_entry_safe, the element obtained by
list_first_entry is not really accessed, only the address of its
list_head field is compared to the address of the list head, so the
list_first_entry is safe.
The semantic patch that makes this change is as follows (with another
variant for the no brace case): (http://coccinelle.lip6.fr/)
<smpl>
@@
expression x,e;
iterator name list_for_each_entry_safe;
statement S;
identifier i,j;
@@
-if (!(list_empty(x))) {
list_for_each_entry_safe(i,j,x,...) S
- }
... when != i
when != j
(
i = e;
|
? j = e;
)
</smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Link: https://lore.kernel.org/r/1595761112-11003-2-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Passing specific snd_soc_card structure depending on the ACPI ID.
In future we can add other IDs in the ACPI table and pass the structure.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200728160255.31020-3-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As in future our machine driver supports multiple codecs
So changing naming convention of snd_soc_card struct and its fields.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200728160255.31020-2-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for voice and BT calls, along with standard
audio output via the speaker, earpiece, headphone jack, HDMI, and
any accessories compatible with Midas boards. This patch also supports
headphone/headset detection and headsets with inline buttons.
[m.szyprowski: adaptation to v5.1+ kernels (DAI links initialization)]
[s.nawrocki: removal of the clk API calls for CODEC MCLK, the jack data
structure moved to struct midas_priv, coding style and typo fixes,
conversion to new cpu/codec/dai-node binding]
Signed-off-by: Simon Shields <simon@lineageos.org>
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200728131111.14334-2-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reset the device before programming the registers or all programming
will be lost as the device resets registers to default settings.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200730142419.28205-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The header was updated to align with the data sheet to start the GPO_CFG
at GPO_CFG0. The code was not updated to the change and therefore the
GPO_CFG0 register was not written to.
Fixes: 6617cff6a0 ("ASoC: tlv320adcx140: Add GPO configuration and drive output config")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200730142419.28205-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All channels are enabled at boot up, this patch ensures that all
channels are disabled at boot and whenever the function is called.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200730055319.1522-3-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable 8kHz audio support for Intel Keem Bay platform.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200730055319.1522-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The allocation order of things in soc_new_pcm_runtime was changed to
move the device_register before the allocation of the rtd structure.
This was to allow the rtd allocation to be managed by devm. However
currently the sysfs entries are added by device_register and their
visibility depends on variables within the rtd structure, this causes
the pmdown_time and dapm_widgets sysfs entries to be missing for all
rtds.
Correct this issue by manually calling device_add_groups after the
appropriate information is available.
Fixes: d918a37610 ("ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200730120715.637-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Standard dai format property don't need the "amlogic," prefix.
There nothing amlogic specific about them. Just remove it.
Fixes: 435857e015 ("ASoC: meson: align axg card driver with DT bindings documentation")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
After carefully checking, it appears that both tdmout and tdmin require the
rising edge of the sclk they get to be synchronized with the frame sync
event (which should be a rising edge of lrclk).
TDMIN was improperly set before this patch. Remove the sclk_invert quirk
which is no longer needed and fix the sclk phase.
Fixes: 1a11d88f49 ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
After carefully checking the result provided by the TDMIN on the g12a and
sm1 SoC families, the TDMIN skew offset appears to be 3 instead of 2 on the
axg.
Fixes: f01bc67f58 ("ASoC: meson: axg-tdm-formatter: rework quirks settings")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The .set_fmt() callback of the axg tdm interface incorrectly
test the content of SND_SOC_DAIFMT_MASTER_MASK as if it was a
bitfield, which it is not.
Implement the test correctly.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add General Purpose Output (GPO) configuration and driver output
configuration. The GPOs can be configured as a GPO, IRQ, SDOUT or a
PDMCLK output. In addition the output drive can be configured with
various configurations.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200728160833.24130-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix white space issues and remove else case where it was not needed.
Convert "static const char *" to "static const char * const"
Fixes: 689c7655b5 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200728164339.16841-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This set of patches is required for facilitating system S0ix
entry when the DSP is in D0I3. This first patch adds the missing
CORB/RIRB DMA stop and restart to the suspend/resume sequence along
with powering up/down the links. The second patch ensures that the
FW traces are disabled when the system enters S0ix with the DSP in D0I3.
Marcin Rajwa (2):
ASoC: SOF: Intel: fix the suspend procedure to support s0ix entry
ASoC: SOF: Intel: disable traces when switching to S0Ix D0I3
sound/soc/sof/intel/hda-dsp.c | 48 ++++++++++++++++++++++++++++++++---
1 file changed, 44 insertions(+), 4 deletions(-)
--
2.25.1
Update the shutdown GPIO property to be shutdown from shut-down.
Fixes: c173dba44c ("ASoC: tas2562: Introduce the TAS2562 amplifier")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200723160838.9738-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We should always disable DMA trace on S0Ix. When staying at S0-D0I3,
we should enable DMA trace while both DMA Trace debug is enabled and
hda_enable_trace_D0I3_S0 is set. This commit corrects the existed
logic errors about that.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com>
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200727183613.1419005-3-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes the suspend & resume procedure to allow entry into the
low power states with some streams being active as a wake source - wake on
voice is a perfect example. The current implementation does not stop
the CORB/RIRB DMA and does not power down the HDA links. With firmware's
help, the platform has been able to still enter s0ix state on older
platforms, but the sequence is still incorrect, and the additional
driver actions are needed to ensure correct s0ix behaviour.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com>
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200727183613.1419005-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Right now the direction of a DAI has to be specified as a literal
number in the device tree, e.g.:
dai@0 {
reg = <0>;
direction = <2>;
};
but this does not make it immediately clear that this is a
playback/RX-only DAI.
Actually, q6asm-dai.c has useful defines for this. Move them to the
dt-bindings header to allow using them in the dts(i) files.
The example above then becomes:
dai@0 {
reg = <0>;
direction = <Q6ASM_DAI_RX>;
};
which is immediately recognizable as playback/RX-only DAI.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200727082502.2341-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
PME_EN state needs to restored to the value set by fmw.
For the devices which are not using I2S wake event which gets
enabled by PME_EN bit, keeping PME_EN enabled burns considerable amount
of power as it blocks low power state.
For the devices using I2S wake event, PME_EN gets enabled in fmw and the
state should be maintained after ACP Power On.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Link: https://lore.kernel.org/r/20200724195600.11798-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87tuxtydcz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87v9i9yddc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
runtime_usage of sound card has been observed to grow without bound.
For example:
$ cat /sys/devices/platform/sound/power/runtime_usage
46
$ sox -n -t s16 -r 48000 -c 2 - synth 1 sine 440 vol 0.1 | \
aplay -q -D hw:0,0 -f S16_LE -r 48000 -c 2
$ cat /sys/devices/platform/sound/power/runtime_usage
52
Commit 4e872a4682 ("ASoC: dapm: Don't force card bias level to be
updated") stops to force update bias_level on card. If card doesn't
provide set_bias_level callback, the snd_soc_dapm_set_bias_level()
is equivalent to NOP for card device.
As a result, dapm_pre_sequence_async() doesn't change the bias_level of
card device correctly. Thus, pm_runtime_get_sync() would be called in
dapm_pre_sequence_async() without symmetric pm_runtime_put() in
dapm_post_sequence_async().
Don't call pm_runtime_* on card device.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200724070731.451377-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment we have two separate functions to parse the sound card
properties from the device tree: qcom_snd_parse_of() for DPCM and
apq8016_sbc_parse_of() without DPCM. These functions are almost identical
except for a few minor differences.
This patch set extends qcom_snd_parse_of() to handle links without DPCM,
so that we can use one common function for all (qcom) machine drivers.
Stephan Gerhold (7):
ASoC: qcom: Use devm for resource management
ASoC: qcom: common: Use snd_soc_dai_link_set_capabilities()
ASoC: q6afe: Remove unused q6afe_is_rx_port() function
ASoC: qcom: common: Support parsing links without DPCM
ASoC: qcom: common: Parse properties with "qcom," prefix
ASoC: qcom: apq8016_sbc: Use qcom_snd_parse_of()
ASoC: qcom: common: Avoid printing errors for -EPROBE_DEFER
sound/soc/qcom/Kconfig | 1 +
sound/soc/qcom/apq8016_sbc.c | 120 ++++-------------------------------
sound/soc/qcom/apq8096.c | 28 +-------
sound/soc/qcom/common.c | 58 ++++++++++-------
sound/soc/qcom/qdsp6/q6afe.c | 8 ---
sound/soc/qcom/qdsp6/q6afe.h | 1 -
sound/soc/qcom/sdm845.c | 40 ++----------
7 files changed, 59 insertions(+), 197 deletions(-)
--
2.27.0
Modify dsm_init sequence and dsm param bin check condition.
- Move dsm_init() to after amp init setting to
make sure dsm init is last setting.
- dsm param bin check condition changed for extended register setting.
Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200724060149.19261-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With commit e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
every error different for ENOTSUPP or EPROBE_DEFER will log an error.
However, as explained in snd_soc_get_dai_name(), this callback may error
to indicate that the DAI is not matched by the component tested. If the
device provides other components, those may still match. Logging an error
in this case is misleading.
Don't use soc_component_ret() in snd_soc_component_of_xlate_dai_name()
to avoid spamming the log.
Fixes: e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200723142020.1338740-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
qcom_snd_parse_of() tends to produce lots of error messages during bootup:
MultiMedia1: error getting cpu dai name
This happens because the DAIs are not probed until the ADSP remoteproc
has booted, which takes a while. Until it is ready, snd_soc_of_get_dai_name()
returns -EDEFER_PROBE to retry probing later. This is perfectly normal,
so cleanup the kernel log a bit by not printing in case of -EPROBE_DEFER.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-8-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we have updated qcom_snd_parse_of() to handle the device
tree bindings used for apq8016_sbc, update the apq8016_sbc driver
to use the common function and remove the duplicated code.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-7-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The apq8016_sbc device tree binding uses a "qcom," vendor prefix
for all device tree properties, while qcom_snd_parse_of() uses the
same properties without a prefix.
In the future it would be nice to make this consistent, however,
for backwards compatibility we need to parse both names to allow
apq8016_sbc to use the common qcom_snd_parse_of() function.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-6-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
So far qcom_snd_parse_of() was only used to parse the device tree
for boards using the QDSP6 driver together with DPCM. apq8016_sbc
uses an almost identical version (apq8016_sbc_parse_of()) which
parses links without DPCM.
Given the similarity of the two functions it is useful to combine
these two. To allow using qcom_snd_parse_of() in apq8016_sbc we
need to support parsing links without DPCM as well.
This is pretty simple: A DPCM link in the device tree is defined using:
- DPCM frontend: "cpu"
- DPCM backend: "cpu", "platform" and "codec"
... while a link without DPCM has "cpu" and "codec" (but no "platform").
Add a few more if conditions to handle links without DPCM correctly.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-5-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 4a95737440 ("ASoc: q6afe: add support to get
port direction"), since the function is not needed anymore.
q6afe-dai already exposes the possible directions for a DAI through
the DAI capabilities (playback/capture-only DAI). Now we use
snd_soc_dai_link_set_capabilities() to infer the information
directly from the DAI capabilities.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-4-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a212008925 ("ASoC: qcom: common: set correct directions for dailinks")
introduced a call to q6afe_is_rx_port() to set the dpcm_playback/capture
parameters correctly. This is necessary because those parameters are now
validated to match the capabilities of the DAIs. [1]
The disadvantage of introducing the call to q6afe_is_rx_port() is that
it makes the qcom_snd_parse_of() helper dependent on the QDSP6 driver.
When the ADSP is bypassed (e.g. in apq8016-sbc) QDSP6 is not used.
There is a generic solution for this now: The correct direction for the links
is already defined by the DAI capabilities (e.g. rx ports only support playback).
Commit 25612477d2 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper")
introduced the snd_soc_dai_link_set_capabilities() function that we can use
to set dpcm_playback/dpcm_capture according to the capabilities of the DAIs.
Use that for both FE/BE DAI links to avoid the dependency on the QDSP6 driver.
[1]: https://lore.kernel.org/alsa-devel/20200616085409.GA110999@gerhold.net/
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify the machine drivers for newer SoCs a bit by using the
devm_* function calls that automatically release the resources
when the driver is removed or when probing fails.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Global EN register guide to off before AMP_EN register
when amp disable sequence.
- remove AMP_EN control before max98390_dac_event call
Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200724060058.19201-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Support same propeties as simple card for configuring fmt
from DT.
In order to make this change compatible with old DT, these
properties are optional.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1595302910-19688-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ESAI interfaces may share same interrupt line with EDMA on
some platforms (e.g. i.MX8QXP, i.MX8QM).
Add IRQF_SHARED flag to allow sharing the irq among several
devices
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1595476808-28927-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Build errors are seen on 32-bit platforms because of a plain 64-by-32
division. For example, following build erros were reported.
"ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko]
undefined!"
"ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko]
undefined!"
This can be fixed by using div_u64() helper from 'math64.h' header.
Fixes: 8c8ff982e9 ("ASoC: tegra: Add Tegra210 based DMIC driver")
Reported-by: Geert Uytterhoeven <geert@linux-m68k.org>
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595492011-2411-1-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_J721E_EVM should not select SND_SOC_PCM3168A_I2C when I2C
is not enabled. That causes build errors, so make this driver's
symbol depend on I2C.
WARNING: unmet direct dependencies detected for SND_SOC_PCM3168A_I2C
Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n]
Selected by [m]:
- SND_SOC_J721E_EVM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (DMA_OMAP [=y] || TI_EDMA [=m] || TI_K3_UDMA [=n] || COMPILE_TEST [=y]) && (ARCH_K3_J721E_SOC [=n] || COMPILE_TEST [=y])
../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: data definition has no type or storage class
module_i2c_driver(pcm3168a_i2c_driver);
^~~~~~~~~~~~~~~~~
../sound/soc/codecs/pcm3168a-i2c.c:59:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int]
../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: parameter names (without types) in function declaration
../sound/soc/codecs/pcm3168a-i2c.c:49:26: warning: ‘pcm3168a_i2c_driver’ defined but not used [-Wunused-variable]
static struct i2c_driver pcm3168a_i2c_driver = {
^~~~~~~~~~~~~~~~~~~
cc1: some warnings being treated as errors
Fixes: 6748d05590 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)")
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/e74c690c-c7f8-fd42-e461-4f33571df4ef@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718112403.13709-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Coefficient files now support additional metadata blocks, these
contain machine parsable text strings describing the parameters
contained in the coefficient file.
Signed-off-by: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200723110321.16382-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718111209.11760-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718110857.11520-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87o8ob0yun.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc-xxx are getting rtd from substream by
rtd = substream->private_data;
But, getting data from "private_data" is very unclear.
This patch adds asoc_substream_to_rtd() macro which is
easy to understand that rtd from substream.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wo2z0yve.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.
The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.
This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.
Tested for all use cases of the driver.
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Tested-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1595432147-11166-1-git-send-email-harshapriya.n@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The series re-uses mt8183-mt6358-ts3a227-max98357.c to support machine driver
with max98357b.
The 1st patch enables left justified format from mt8183 audio platform.
The 2nd patch adds document for the new proposed compatible string for
max98357b.
The 3rd patch supports machine driver with max98357b and uses left justified
format for it.
Tzung-Bi Shih (3):
ASoC: mediatek: mt8183: support left justified format for I2S
ASoC: dt-bindings: mt8183: add compatible string for using max98357b
ASoC: mediatek: mt8183: support machine driver with max98357b
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 +
sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 59 ++++++++++++++++---
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 22 ++++++-
3 files changed, 73 insertions(+), 9 deletions(-)
--
2.28.0.rc0.105.gf9edc3c819-goog
Daniel Baluta <daniel.baluta@nxp.com>:
From: Daniel Baluta <daniel.baluta@nxp.com>
This patchseries contains a couple of SOF IMX fixes
found during our first IMX SOF release.
Daniel Baluta (7):
ASoC: SOF: define INFO_ flags in dsp_ops for imx8
ASoC: SOF: imx: Use ARRAY_SIZE instead of hardcoded value
ASoC: SOF: imx8: Fix ESAI DAI driver name for i.MX8/iMX8X
ASoC: SOF: imx8m: Fix SAI DAI driver for i.MX8M
ASoC: SOF: imx8: Add SAI dai driver for i.MX/i.MX8X
ASoC: SOF: topology: Update SAI config bclk/fsync rate
ASoC: SOF: pcm: Update rate/channels for SAI/ESAI DAIs
sound/soc/sof/imx/imx8.c | 24 +++++++++++++++++++++---
sound/soc/sof/imx/imx8m.c | 4 ++--
sound/soc/sof/pcm.c | 8 ++++++++
sound/soc/sof/topology.c | 2 ++
4 files changed, 33 insertions(+), 5 deletions(-)
--
2.17.1
Commit 5bd70440cb ("ASoC: soc-dai: revert all changes to DAI
startup/shutdown sequence"), introduced a slight change of semantics
to DAI startup/shutdown. If startup() returns an error, shutdown()
is now called for the DAI.
This causes a deadlock in hdac_hda which issues a call to
snd_hda_codec_pcm_put() in case open fails. Upon error, soc_pcm_open()
will call shutdown(), and pcm_put() ends up getting called twice. Result
is a deadlock on pcm->open_mutex, as snd_device_free() gets called from
within snd_pcm_open(). Typical task backtrace looks like this:
[ 334.244627] snd_pcm_dev_disconnect+0x49/0x340 [snd_pcm]
[ 334.244634] __snd_device_disconnect.part.0+0x2c/0x50 [snd]
[ 334.244640] __snd_device_free+0x7f/0xc0 [snd]
[ 334.244650] snd_hda_codec_pcm_put+0x87/0x120 [snd_hda_codec]
[ 334.244660] soc_pcm_open+0x6a0/0xbe0 [snd_soc_core]
[ 334.244676] ? dpcm_add_paths.isra.0+0x491/0x590 [snd_soc_core]
[ 334.244679] ? kfree+0x9a/0x230
[ 334.244686] dpcm_be_dai_startup+0x255/0x300 [snd_soc_core]
[ 334.244695] dpcm_fe_dai_open+0x20e/0xf30 [snd_soc_core]
[ 334.244701] ? snd_pcm_hw_rule_muldivk+0x110/0x110 [snd_pcm]
[ 334.244709] ? dpcm_be_dai_startup+0x300/0x300 [snd_soc_core]
[ 334.244714] ? snd_pcm_attach_substream+0x3c4/0x540 [snd_pcm]
[ 334.244719] snd_pcm_open_substream+0x69a/0xb60 [snd_pcm]
[ 334.244729] ? snd_pcm_release_substream+0x30/0x30 [snd_pcm]
[ 334.244732] ? __mutex_lock_slowpath+0x10/0x10
[ 334.244736] snd_pcm_open+0x1b3/0x3c0 [snd_pcm]
Fixes: 5bd70440cb ("ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2159
Link: https://lore.kernel.org/r/20200717101950.3885187-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdac_hda remove implementation fails to free the hda codec
resources, leading to memleaks at module unload. This gap has been there
from the start, commit 6bae5ea949 ("ASoC: hdac_hda: add asoc
extension for legacy HDA codec drivers").
Instead of duplicating the cleanup logic, use the common
snd_hda_codec_cleanup_for_unbind() to free the resources. Remove
existing code in hdac_hda to cleanup "codec.jackpoll_work" and call to
snd_hdac_regmap_exit(), as these are already done in
snd_hda_codec_cleanup_for_unbind().
The cleanup is done in ASoC component remove() callback and not in the
HDAC bus hdev_detach(). This is done to ensure the codec specific
cleanup routines are run before the parent card is freed.
Fixes: 6bae5ea949 ("ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2195
Link: https://lore.kernel.org/r/20200717101950.3885187-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error handling for patch_ops in hdac_hda_codec_probe().
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200717101950.3885187-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200719153822.59788-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Supports machine driver with max98357b
("mt8183-mt6358-ts3a227-max98357b").
The key difference from max98357a: max98357b needs to use left
justified format.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200720012559.906088-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
MT8183 audio platform supports EIAJ and I2S formats. The code fixed to
use I2S format in the past.
Supports EIAJ mode via set_fmt ops and preserves to use I2S format as
the default format intentionally.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200720012559.906088-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Starting in commit cbc7a6b5a8 ("ASoC: soc-card: add
snd_soc_card_add_dai_link()"), error value from ASoc add_dai_link() is
no longer ignored.
The generic HDA machine driver relied on the old semantics to disable
i915 HDMI/DP audio codec at runtime. If no display codec was present,
add_dai_link() returned an error, but this was ignored and rest of the
card was successfully probed.
Fix the problem by changing the machine driver add_dai_link() to not
return an error in this case.
Fixes: cbc7a6b5a8 ("ASoC: soc-card: add snd_soc_card_add_dai_link()")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2261
Link: https://lore.kernel.org/r/20200714132804.3638221-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixup BE DAI links rate/channels parameters to match any values
from topology.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-8-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These parameters are read from topology file and sent to DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-7-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With SOF we support 1 ESAI interface and 1 SAI interface.
This patch adds SAI1 interface support existing on i.MX8/i.MX8X
boards.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-6-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This must match DAI name from topology. Also, sai-port
is too generic. Physical DAI port on i.MX8MP is labeled SAI3.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-5-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This must match DAI name from topology. Also, esai-port is too generic
as they are 2 ESAIs on i.MX8/i.MX8X boards.
SOF integration only uses ESAI0 for now.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-4-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With this change we no longer need to update num_drv when adding
new DAI driver.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-3-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the past, the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP were
defined in the SOF PCM core, but that was changed since
commit 27e322fabd ("ASoC: SOF: define INFO_ flags in dsp_ops")
Now these flags must be set in DSP ops.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
ADMAIF is the interface between ADMA and AHUB. Each ADMA channel that
sends/receives data to/from AHUB must intreface through an ADMAIF channel.
ADMA channel sending data to AHUB pairs with an ADMAIF Tx channel and
similarly ADMA channel receiving data from AHUB pairs with an ADMAIF Rx
channel. Buffer size is configurable for each ADMAIF channel, but currently
SW uses default values.
This patch registers ADMAIF driver with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes ADMAIF interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The ADMAIF device can be enabled in the DT via
"nvidia,tegra210-admaif" compatible binding.
Tegra PCM driver is updated to expose required PCM interfaces and
snd_pcm_ops callbacks.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-8-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the reset property name when allocating the GPIO descriptor.
The gpiod_get_optional appends either the -gpio or -gpios suffix to the
name.
Fixes: 1a476abc72 ("tas2770: add tas2770 smart PA kernel driver")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200720181202.31000-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Partially reverts commit 128f825aea ("ASoC: max98357a: move control
of SD_MODE to DAPM").
In order to have mute control of max98357 from machine drivers, commit
128f825aea ("ASoC: max98357a: move control of SD_MODE to DAPM")
moves the control of SD_MODE from DAI ops to DAPM events. However, pop
noise has been observed on rk3399-gru-kevin boards due to this commit.
The commit 128f825aea caused sequence of DAI clocks and SD_MODE
changed on rk3399-gru-kevin boards.
With the commit 128f825aea:
- SD_MODE will be set to 1 before DAI clocks start.
- SD_MODE will be set to 0 after DAI clocks stop.
As a result, pop noise.
Moves the control of SD_MODE back to DAI ops. In the meantime, uses an
additional flag in DAPM event to provide chance of mute control for
machine drivers.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Tested-By: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Link: https://lore.kernel.org/r/20200721114232.2812254-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This PR became fairly large, containing mostly the collection of
ASoC fixes that slipped from the previous request, so I sent now
a bit earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests
and fuzzing. The rest are other ASoC device-specific fixes (imx,
qcom, wm8974, amd, rockchip) as well as a trivial fix for a kernel
WARNING hit by syzkaller.
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Merge tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"This became fairly large, containing mostly the collection of ASoC
fixes that slipped from the previous request, so I sent now a bit
earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests and
fuzzing. The rest are other ASoC device-specific fixes (imx, qcom,
wm8974, amd, rockchip) as well as a trivial fix for a kernel WARNING
hit by syzkaller"
* tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits)
ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
ALSA: info: Drop WARN_ON() from buffer NULL sanity check
ASoC: rt5682: Report the button event in the headset type only
ASoC: Intel: bytcht_es8316: Add missed put_device()
ASoC: rt5682: Enable Vref2 under using PLL2
ASoC: rt286: fix unexpected interrupt happens
ASoC: wm8974: remove unsupported clock mode
ASoC: wm8974: fix Boost Mixer Aux Switch
ASoC: SOF: core: fix null-ptr-deref bug during device removal
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: codecs: max98373: Removed superfluous volume control from chip default
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: fix kernel oops on route addition error
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
ASoC: Intel: bdw-rt5677: fix non BE conversion
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
ASoC: core: Remove only the registered component in devm functions
MAINTAINERS: Change Maintainer for some at91 drivers
ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
...
In commit d696a61413 ("ASoC: rt1015: Add condition to prevent SoC
providing bclk in ratio of 50 times of sample rate."), PLL input at 50fs
is no longer supported, the new recommended settings at 48Khz rate are:
PLL input SSP bclk
------------------------
64fs 3.073Mhz
100fs 4.8Mhz
(bclk update is reflected in topoplogy.)
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The mc_private->hdmi_pcm_list is populated by elements loaded during
DSP topology load. Valid topologies for this machine driver will always
have PCM nodes for HDMI, but driver should fail gracefully even in the case
this is not true. Add a sanity check to sof_sdw_hdmi_card_late_probe()
for this case. Without the fix, a null pcm handle gets dereferenced.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Extend the generic SOF Soundwire machine driver to support systems where
iDisp HDMI/DP audio codec is disabled for some reason (i915 driver
disabled, HDMI/DP implemented with a discrete GPU, etc). Switch codecs
to SoC dummy in the affected DAI links. This allows to reuse existing
topologies for this case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt711 jack detection properties are set from the machine drivers
during the card probe, as done in other ASoC examples.
KASAN reports a use-after-free error when unbinding drivers due to a
confusing sequence between the ACPI core, the device core and the
SoundWire device cleanups.
Rather than fixing this sequence, follow the recommendation to have
the same caller add and remove properties, add an explicit
device_remove_properties() in the card .remove() callback.
In future patches the use of device_add/remove_properties will be
replaced by a direct handling of a swnode, but the sequence will
remain the same.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We can get codec name from dai link.
Suggested-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
The Digital Speaker Controller (DSPK) converts the multi-bit Pulse Code
Modulation (PCM) audio input to oversampled 1-bit Pulse Density Modulation
(PDM) output. From the signal flow perpsective, the DSPK can be viewed as
a PDM transmitter that up-samples the input to the desired sampling rate
by interpolation then converts the oversampled PCM input to the desired
1-bit output via Delta Sigma Modulation (DSM).
This patch registers DSPK component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DSPK interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DSPK devices can be enabled in the DT via
"nvidia,tegra186-dspk" compatible binding. This driver can be used
on Tegra194 chip as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-7-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Audio Hub (AHUB) comprises a collection of hardware accelerators for
audio pre/post-processing and a programmable full crossbar (XBAR) for
routing audio data across these accelerators in time and in parallel.
AHUB supports multiple interfaces to I2S, DSPK, DMIC etc., XBAR is a
switch used to configure or modify audio routing between HW accelerators
present inside AHUB.
This patch registers AHUB component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes AHUB interfaces, which can be used to connect different
components in the ASoC layer. Currently the driver takes care of XBAR
programming to allow audio data flow through various clients of the AHUB.
Makefile and Kconfig support is added to allow to build the driver. The
AHUB component can be enabled in the DT via below compatible bindings.
- "nvidia,tegra210-ahub" for Tegra210
- "nvidia,tegra186-ahub" for Tegra186 and Tegra194
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-6-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Inter-IC Sound (I2S) controller implements full-duplex, bi-directional
and single direction point to point serial interface. It can interface
with I2S compatible devices. Tegra I2S controller can operate as both
master and slave.
This patch registers I2S controller with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes I2S interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The I2S devices can be enabled in the DT via
"nvidia,tegra210-i2s" compatible binding.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-5-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Digital MIC (DMIC) Controller is used to interface with Pulse Density
Modulation (PDM) input devices. The DMIC controller implements a converter
to convert PDM signals to Pulse Code Modulation (PCM) signals. From signal
flow perspective, the DMIC can be viewed as a PDM receiver.
This patch registers DMIC component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DMIC interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DMIC devices can be enabled in the DT via
"nvidia,tegra210-dmic" compatible string. This driver can be used for
Tegra186 and Tegra194 chips as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio Client Interface (CIF) is a proprietary interface employed to route
audio samples through Audio Hub (AHUB) components by inter connecting the
various modules.
This patch exports an inline function tegra_set_cif() which can be used,
for now, to program CIF on Tegra210 and later Tegra generations. Later it
can be extended to include helpers for legacy chips as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Reviewed-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/1595134890-16470-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This configuration is for EHL with the RT5660 codec. RT5660
should use "10EC5660" ID instead of "INTC1027".
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All drivers are now using .mute_stream.
Let's remove .digital_mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87h7u72dqz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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Merge tag 'asoc-fix-v5.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
Some settings should set to default value after the calibration.
This patch also disables the 25MHz and 1MHz clock power when the jack unplugged.
The JD is triggered by JDH, therefore this patch removes JDL setting.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070228.28660-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the function q6adm_open(), q6adm_alloc_copp() doesn't return
NULL. Thus use IS_ERR() to validate the returned value instead
of IS_ERR_OR_NULL(). And delete the extra line.
Signed-off-by: Zhang Shengju <zhangshengju@cmss.chinamobile.com>
Signed-off-by: Tang Bin <tangbin@cmss.chinamobile.com>
Link: https://lore.kernel.org/r/20200714112744.20560-1-tangbin@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pin status of the widget was connected after the sound card registered.
The rt5682_headset_detect function will use the pin status of these two widgets
to decide the certain register setting on/off.
Therefore this patch disables the pin of these two widgets in the codec probe.
This patch could avoid the misjudgment.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070256.28712-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is used for both CPU and Codec.
For example, soc_pcm_prepare() / soc_pcm_hw_free() are caring
both CPU and Codec.
But soc_resume_deferred() / snd_soc_suspend() are not.
This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87ft9r2dqr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
-
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/87eepb2dnq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
axg_card_add_tdm_loopback() misses to call kfree() in an error path. We
can use devm_kasprintf() to fix the issue, also improve maintainability.
So use it instead.
Fixes: c84836d7f6 ("ASoC: meson: axg-card: use modern dai_link style")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Support hp and mic detection.
Add a parameter for asoc_simple_init_jack.
Shengjiu Wang (3):
ASoC: simple-card-utils: Support configure pin_name for
asoc_simple_init_jack
ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio
ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection
changes in v2:
- Add more comments in third commit
- Add Acked-by Nicolin.
.../bindings/sound/fsl-asoc-card.txt | 3 +
include/sound/simple_card_utils.h | 6 +-
sound/soc/fsl/Kconfig | 1 +
sound/soc/fsl/fsl-asoc-card.c | 77 ++++++++++++++++++-
sound/soc/generic/simple-card-utils.c | 7 +-
5 files changed, 86 insertions(+), 8 deletions(-)
--
2.27.0
Add missed return for calling soc_component_ret, otherwise the return
value is wrong.
Fixes: e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1594876028-1845-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use asoc_simple_init_jack function from simple card to implement
the Headphone and Microphone detection.
Register notifier to disable Speaker when Headphone is plugged in
and enable Speaker when Headphone is unplugged.
Register notifier to disable Digital Microphone when Analog Microphone
is plugged in and enable DMIC when Analog Microphone is unplugged.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the pin_name is fixed in asoc_simple_init_jack, but some driver
may use a different pin_name. So add a new parameter in
asoc_simple_init_jack for configuring pin_name.
If this parameter is NULL, then the default pin_name is used.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87pn95wiwa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87r1tlwiwe.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/87sge1wiwi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87tuyhwiwm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87v9ixwiwr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87wo3dwiwv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87y2ntwix0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87zh89wix5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/871rllxxhp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/873661xxhu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/874kqhxxhz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/875zaxxxi4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/878sftxxie.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87a709xxij.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87blkpxxip.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
For hdmi-codec, we need to update struct hdmi_codec_ops,
and all its users in the same time.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87d055xxj2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling "direction".
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
To prepare merging mute_stream()/digital_mute(),
this patch adds .no_capture_mute support to emulate .digital_mute().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87eeplxxj7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() will return -ENOTSUPP if driver doesn't
support mute.
In hdmi-codec case, hdmi_codec_digital_mute() will be used for it,
and each driver has .digital_mute() callback.
hdmi_codec_digital_mute() want to return -ENOTSUPP to follow it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fta1xxjc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq work will be manipulated by resume function, and it will report
the wrong jack type while the jack type is headphone in the button event.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that there's a function that calculates the SHA-256 digest of a
buffer in one step, use it instead of sha256_init() + sha256_update() +
sha256_final().
Also simplify the code by inlining calculate_sha256() into its caller
and switching a debug log statement to use %*phN instead of bin2hex().
Acked-by: Tzung-Bi Shih <tzungbi@google.com>
Reviewed-by: Ard Biesheuvel <ardb@kernel.org>
Cc: alsa-devel@alsa-project.org
Cc: Ard Biesheuvel <ardb@kernel.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Cc: Guenter Roeck <groeck@chromium.org>
Cc: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Eric Biggers <ebiggers@google.com>
Signed-off-by: Herbert Xu <herbert@gondor.apana.org.au>
siu is using discriminatory terms for function parameter.
This patch changes it to "secondary"
One note here is that it do nothing to DMA related naming
for now, because it needs framework level modification.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d04z3qqg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd is using discriminatory terms for function names.
This patch changes it to "secondary"
One note here is that it do nothing to DMA related naming
for now, because it needs framework level modification.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h7ub3qra.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series is a follow up for a long time ago series
(https://patchwork.kernel.org/cover/11204303/).
The old series bound too much on the patches of DRM bridge and ASoC
machine driver. And unluckily, the dependencies
(https://lore.kernel.org/patchwork/patch/1126819/) have not applied.
Revewing the ASoC patches in the old series, I found that they could be
decoupled from the DRM bridge patches. And they are harmless as it is
an optional attribute ("hdmi-codec") in DTS.
This series arranges and rebases the harmless ASoC patches for
mt8183-mt6358-ts3a227-max98357 and mt8183-da7219-max98357.
The 1st and 4th patch add an optional DT property. The 1st patch was
acked long time ago (https://patchwork.kernel.org/patch/11204321/).
The 2nd and 5th patch add DAI link for using hdmi-codec.
The 3rd and 6th patch support the HDMI jack reporting.
Tzung-Bi Shih (6):
ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec"
ASoC: mediatek: mt8183: use hdmi-codec
ASoC: mediatek: mt8183: support HDMI jack reporting
ASoC: dt-bindings: mt8183-da7219: add a property "mediatek,hdmi-codec"
ASoC: mediatek: mt8183-da7219: use hdmi-codec
ASoC: mediatek: mt8183-da7219: support HDMI jack reporting
.../bindings/sound/mt8183-da7219-max98357.txt | 4 +++
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 ++
sound/soc/mediatek/Kconfig | 2 ++
.../mediatek/mt8183/mt8183-da7219-max98357.c | 29 +++++++++++++++++--
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 +++++++++++++++++--
5 files changed, 60 insertions(+), 6 deletions(-)
--
2.27.0.383.g050319c2ae-goog
Both Lee Jones and I submitted separate series, this is the second
part of the merged result, for which no feedback was provided.
I picked Lee's patches for rt5659 and ak4458 and added the pxa and
ux500 that I didn't fix. The rest is largely identical between our
respective series, with the exception of the sunxi which I documented
and Lee removed. I don't have any specific preference and will go with
the flow on this.
Changes since v3:
Improved commit subjects from 'fix kernel-doc' as suggested by Lee
Jones. In a couple of cases I just reverted to Lee's patches when the
code was identical.
Added a couple of CC: tags from Lee's patches.
Added Arnaud Pouliquen's Acked-by tag in first patch.
Lee Jones (6):
ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in
kernel-doc
ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header
ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and
'reg_val_TSTDR'
ASoC: codecs: rt5659: Remove many unused const variables
ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc
header
ASoC: codecs: ak4458: Remove set but never checked variable 'ret'
Pierre-Louis Bossart (4):
ASoC: sti: uniperif: fix 'defined by not used' warning
ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile'
ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in
kernel-doc
ASoC: codecs: rt5631: fix spurious kernel-doc start and missing
arguments
sound/soc/codecs/ak4458.c | 6 +++---
sound/soc/codecs/rt5631.c | 8 +++++--
sound/soc/codecs/rt5659.c | 37 ---------------------------------
sound/soc/codecs/tlv320aic26.c | 2 +-
sound/soc/pxa/pxa-ssp.c | 2 +-
sound/soc/qcom/qdsp6/q6asm.c | 2 +-
sound/soc/sti/uniperif.h | 2 +-
sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++-
sound/soc/sunxi/sun4i-spdif.c | 2 +-
sound/soc/ux500/ux500_msp_i2s.c | 8 +++----
10 files changed, 27 insertions(+), 52 deletions(-)
base-commit: 6940701c71
--
2.25.1
Clear the validity bit for TX
Add kctl for configuring TX validity bit
Shengjiu Wang (2):
ASoC: fsl_spdif: Clear the validity bit for TX
ASoC: fsl_spdif: Add kctl for configuring TX validity bit
sound/soc/fsl/fsl_spdif.c | 51 ++++++++++++++++++++++++++++++++++++---
1 file changed, 47 insertions(+), 4 deletions(-)
--
2.21.0
As Pierre-Louis Bossart pointed out, saying that the default mode for the
SSP is TDM 4 slot is not entirely accurate.
There really are 2 default modes:
The default mode for the SSP configuration is TDM 4 slot for the
cpu-dai (hard-coded in DSP firmware),
The default mode for the SSP configuration is I2S for the codec-dai
(hard-coded in the 'SSP2-Codec" .dai_fmt masks, so far unused).
This commit updates the comment in cht_codec_fixup() to properly reflect
this.
Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200703103840.333732-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add one kctl for configuring TX validity bit from user
space.
The type of this kctl is boolean:
on - Outgoing validity always set
off - Outgoing validity always clear
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1594112066-31297-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In IEC958 spec, "The validity bit is logical "0" if the
information in the main data field is reliable, and it
is logical "1" if it is not".
The default value of "ValCtrl" is zero, which means
"Outgoing Validity always set", then all the data is not
reliable, then some spdif sink device will drop the data.
So set "ValCtrl" to 1, that is to clear "Outgoing Validity"
in default.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1594112066-31297-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks as though the result of snd_soc_update_bits() has never been checked.
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/ak4458.c: In function ‘ak4458_set_dai_mute’:
sound/soc/codecs/ak4458.c:408:16: warning: variable ‘ret’ set but not
used [-Wunused-but-set-variable]
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp>
Cc: Mihai Serban <mihai.serban@nxp.com>
Link: https://lore.kernel.org/r/20200709162328.259586-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the only use of kerneldoc in the sourcefile and no
descriptions are provided.
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/tlv320aic26.c:138: warning: Function parameter or
member 'dai' not described in 'aic26_mute'
sound/soc/codecs/tlv320aic26.c:138: warning: Function parameter or
member 'mute' not described in 'aic26_mute'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Grant Likely <grant.likely@secretlab.ca>
Link: https://lore.kernel.org/r/20200709162328.259586-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks as though they've never been used.
Fixes the following W=1 kernel build warning(s):
In file included from sound/soc/codecs/rt5659.c:25:
In file included from sound/soc/codecs/rt5659.c:25:
sound/soc/codecs/rt5659.c:1232:2: warning: ‘rt5659_ad_monor_asrc_enum’ defined but not used [-Wunused-const-variable=]
1232 | rt5659_ad_monor_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_R_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1231:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1231 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1228:2: warning: ‘rt5659_ad_monol_asrc_enum’ defined but not used [-Wunused-const-variable=]
1228 | rt5659_ad_monol_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_L_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1227:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1227 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1224:2: warning: ‘rt5659_ad_sto2_asrc_enum’ defined but not used [-Wunused-const-variable=]
1224 | rt5659_ad_sto2_asrc_enum, RT5659_ASRC_3, RT5659_AD_STO2_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1223:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1223 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1220:2: warning: ‘rt5659_ad_sto1_asrc_enum’ defined but not used [-Wunused-const-variable=]
1220 | rt5659_ad_sto1_asrc_enum, RT5659_ASRC_2, RT5659_AD_STO1_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1219:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1219 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1216:2: warning: ‘rt5659_da_monor_asrc_enum’ defined but not used [-Wunused-const-variable=]
1216 | rt5659_da_monor_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_R_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1215:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1215 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1212:2: warning: ‘rt5659_da_monol_asrc_enum’ defined but not used [-Wunused-const-variable=]
1212 | rt5659_da_monol_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_L_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1211:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1211 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1208:2: warning: ‘rt5659_da_sto_asrc_enum’ defined but not used [-Wunused-const-variable=]
1208 | rt5659_da_sto_asrc_enum, RT5659_ASRC_2, RT5659_DA_STO_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1207:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1207 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
Signed-off-by: Lee Jones <lee.jones@linaro.org>
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/20200709162328.259586-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'component' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'reg' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'value' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:82: warning: Function parameter or member
'component' not described in 'rt5631_read_index'
sound/soc/codecs/rt5631.c:82: warning: Function parameter or member
'reg' not described in 'rt5631_read_index'
sound/soc/codecs/rt5631.c:367: warning: Function parameter or member
'component' not described in 'onebit_depop_power_stage'
sound/soc/codecs/rt5631.c:405: warning: Function parameter or member
'component' not described in 'onebit_depop_mute_stage'
sound/soc/codecs/rt5631.c:443: warning: Function parameter or member
'component' not described in 'depop_seq_power_stage'
sound/soc/codecs/rt5631.c:515: warning: Function parameter or member
'component' not described in 'depop_seq_mute_stage'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709162328.259586-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like these have been unchecked since the driver's inception in 2012.
Fixes the following W=1 kernel build warning(s):
sound/soc/ux500/ux500_msp_i2s.c: In function ‘flush_fifo_rx’:
sound/soc/ux500/ux500_msp_i2s.c:398:6: warning: variable ‘reg_val_DR’
set but not used [-Wunused-but-set-variable]
sound/soc/ux500/ux500_msp_i2s.c: In function ‘flush_fifo_tx’:
sound/soc/ux500/ux500_msp_i2s.c:415:6: warning: variable
‘reg_val_TSTDR’ set but not used [-Wunused-but-set-variable]
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: zhong jiang <zhongjiang@huawei.com>
Cc: Ola Lilja <ola.o.lilja@stericsson.com>
Cc: Roger Nilsson <roger.xr.nilsson@stericsson.com>
Cc: Sandeep Kaushik <sandeep.kaushik@st.com>
Link: https://lore.kernel.org/r/20200709162328.259586-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the only use of kerneldoc in the sourcefile and full
descriptions are not provided.
Fixes the following W=1 kernel build warning(s):
sound/soc/pxa/pxa-ssp.c:186: warning: Function parameter or member
'ssp' not described in 'pxa_ssp_set_scr'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Daniel Mack <daniel@zonque.org>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Robert Jarzmik <robert.jarzmik@free.fr>
Link: https://lore.kernel.org/r/20200709162328.259586-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Property name descriptions need to match exactly.
Fixes the following W=1 kernel build warning(s):
sound/soc/sunxi/sun4i-spdif.c:178: warning: Function parameter or
member 'reg_dac_txdata' not described in 'sun4i_spdif_quirks'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Maxime Ripard <mripard@kernel.org>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: Philipp Zabel <p.zabel@pengutronix.de>
Cc: Andrea Venturi <be17068@iperbole.bo.it>
Cc: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20200709162328.259586-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings - missing fields in description
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or
member 'bclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'num_bclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'mclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'num_mclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_bclk_parent_rate' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_sr' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_wss' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'set_chan_cfg' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'set_fmt' not described in 'sun4i_i2s_quirks'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Maxime Ripard <mripard@kernel.org>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: Philipp Zabel <p.zabel@pengutronix.de>
Cc: Andrea Venturi <be17068@iperbole.bo.it>
Link: https://lore.kernel.org/r/20200709162328.259586-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
sound/soc/qcom/qdsp6/q6asm.c:924: warning: Function parameter or
member 'codec_profile' not described in 'q6asm_open_write'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200709162328.259586-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning. The table uni_tdm_hw is declared in a header included
by multiple C file. This isn't really a good practice but for now
using __maybe_unused makes the following warning go away.
sound/soc/sti/sti_uniperif.c:12:
sound/soc/sti/uniperif.h:1351:38: warning: ‘uni_tdm_hw’ defined but
not used [-Wunused-const-variable=]
1351 | static const struct snd_pcm_hardware uni_tdm_hw = {
| ^~~~~~~~~~
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Link: https://lore.kernel.org/r/20200709162328.259586-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series tries to reuse mt8183-da7219-max98357.c for supporting machine
driver with rt1015 speaker amplifier.
The first 3 patches refactor the code for easier to change for subsequent
patches.
The 4th patch adds document for the new proposed compatible string.
The 5th patch changes the machine driver to support either "MAX98357A" or
"RT1015" codecs.
Tzung-Bi Shih (5):
ASoC: mediatek: mt8183-da7219: sort header inclusions in alphabetical
ASoC: mediatek: mt8183-da7219: remove forward declaration of
headset_init
ASoC: mediatek: mt8183-da7219: extract codec and DAI names
ASoC: mediatek: mt8183-da7219: add compatible string for using rt1015
ASoC: mediatek: mt8183-da7219: support machine driver with rt1015
.../bindings/sound/mt8183-da7219-max98357.txt | 5 +-
sound/soc/mediatek/Kconfig | 5 +-
.../mediatek/mt8183/mt8183-da7219-max98357.c | 244 ++++++++++++++----
3 files changed, 197 insertions(+), 57 deletions(-)
--
2.27.0.383.g050319c2ae-goog
Add the TX offset slot programming. There is no RX offset slot
register.
Since there is no RX offset the check for slot symmetry can be removed.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200709185129.10505-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CPU and the codec both are represented now as components, so for
PDMIC we are registering two componenets with the same name. Since
there is no actual codec, we will merge the codec component into the
CPU one and use a dummy codec instead, for the DAI link.
As a bonus, debugfs will no longer report an error when will try to
create entries for both componenets with the same name.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200708163359.2698696-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CPU and the codec both are represented now as components, so for
CLASS-D we are registering two componenets with the same name. Since
there is no actual codec, we will merge the codec component into the
CPU one and use a dummy codec instead, for the DAI link.
As a bonus, debugfs will no longer report an error when will try to
create entries for both componenets with the same name.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200708101249.2626560-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning and removed unused table. In this case this a
duplicate of
static const struct of_device_id max98390_of_match[] = {
{ .compatible = "maxim,max98390", },
{}
};
MODULE_DEVICE_TABLE(of, max98390_of_match);
already used in the rest of the code.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200707190612.97799-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like 'w' has remained unchecked since the driver's inception.
Fixes the following W=1 kernel build warning(s):
sound/soc/ti/omap-mcbsp-st.c: In function ‘omap_mcbsp_st_chgain’:
sound/soc/ti/omap-mcbsp-st.c:145:6: warning: variable ‘w’ set but not used [-Wunused-but-set-variable]
Peter suggested that the whole read can be removed, so that's
been done too.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Samuel Ortiz <samuel.ortiz@nokia.com>
Cc: linux-omap@vger.kernel.org
Link: https://lore.kernel.org/r/20200707190612.97799-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
In file included from include/sound/tlv.h:10,
from sound/soc/codecs/jz4770.c:19:
sound/soc/codecs/jz4770.c:306:35: warning: ‘mic_boost_tlv’ defined but not used [-Wunused-const-variable=]
306 | static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0);
| ^~~~~~~~~~~~~
include/uapi/sound/tlv.h:64:15: note: in definition of macro ‘SNDRV_CTL_TLVD_DECLARE_DB_SCALE’
64 | unsigned int name[] = { | ^~~~
sound/soc/codecs/jz4770.c:306:14: note: in expansion of macro ‘DECLARE_TLV_DB_SCALE’
306 | static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0);
| ^~~~~~~~~~~~~~~~~~~~
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Paul Cercueil <paul@crapouillou.net>
Cc: ter Huurne <maarten@treewalker.org>
Link: https://lore.kernel.org/r/20200707190612.97799-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning, the kernel-doc syntax was probably from Doxygen?
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/20200707190612.97799-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings - missing fields in structure
Credits to Sylwester Nawrocki for the pclk and cclk descriptions.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20200707190612.97799-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning
Kernel-doc is not used in one file and missing argument in the second.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20200707190612.97799-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Disable Left and Right Spk pin after boot so that sof can get
suspended.
This follows the same logic added to another machine driver with
commit 94d2d08974 ("ASoC: Intel: Boards: tgl_max98373: add dai_trigger function")
Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reflect Kconfig changes and add both SoundWire and I2C modes
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add SoundWire specific parts and extend common ones already split from
I2C.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To prepare support for SoundWire, let's first split the I2C and common
parts. No new functionality, just indents and formatting to make
checkpatch happy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Supports machine driver with rt1015 ("mt8183-da7219-rt1015"). Embeds in
existing mt8183-da7219-max98357.c because they share most of the code.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200709122445.1584497-6-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HV/VREF should not turn off if the headphone jack plug-in.
This patch could solve the unexpected interrupt issue in some devices.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie.
inverted frame clock is not support in this mode).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees. In addition, a few usual HD-audio
and USB-audio quirks, Intel HDMI fixes, ASoC fsl and rt5682 fixes,
as well as the fix in compress-offload partial drain operation.
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Merge tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees.
In addition, a few usual HD-audio and USB-audio quirks, Intel HDMI
fixes, ASoC fsl and rt5682 fixes, as well as the fix in
compress-offload partial drain operation"
* tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: compress: fix partial_drain completion state
ALSA: usb-audio: Add implicit feedback quirk for RTX6001
ALSA: usb-audio: add quirk for MacroSilicon MS2109
ALSA: hda/realtek: Enable headset mic of Acer Veriton N4660G with ALC269VC
ALSA: hda/realtek: Enable headset mic of Acer C20-820 with ALC269VC
ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
ALSA: hda/hdmi: improve debug traces for stream lookups
ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
ALSA: opl3: fix infoleak in opl3
ALSA: usb-audio: Replace s/frame/packet/ where appropriate
ALSA: usb-audio: Fix packet size calculation
AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module
ALSA: hda - let hs_mic be picked ahead of hp_mic
ASoC: rt5682: fix the pop noise while OMTP type headset plugin
ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
ASoC: fsl_mqs: Don't check clock is NULL before calling clk API
This series tries to reuse mt8183-mt6358-ts3a227-max98357.c for supporting
machine driver with rt1015 speaker amplifier.
The 1st patch is straightforward: re-order the header inclusions.
The 2nd patch adds document for the new proposed compatible string.
The 3rd patch changes the machine driver to support either "MAX98357A" or
"RT1015" codecs.
Tzung-Bi Shih (3):
ASoC: mediatek: mt8183: sort header inclusions in alphabetical
dt-bindings: mt8183: add compatible string for using rt1015
ASoC: mediatek: mt8183: support machine driver with rt1015
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 5 +-
sound/soc/mediatek/Kconfig | 5 +-
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 171 +++++++++++++++---
3 files changed, 153 insertions(+), 28 deletions(-)
--
2.27.0.383.g050319c2ae-goog
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac579
--
2.25.1
This patchset adds gapless compressed audio support on q6asm.
Gapless on q6asm is implemented using 2 streams in a single asm session.
First few patches are enhacements done to q6asm interface to allow
stream id per each command, gapless flags and silence meta data.
Along with this there are few trivial changes which I thought are necessary!
Last patch implements copy callback to allow finer control over buffer offsets,
specially in partial drain cases.
This patchset is tested on RB3 aka DB845c platform.
Thanks,
srini
Srinivas Kandagatla (11):
ASoC: q6asm: add command opcode to timeout error report
ASoC: q6asm: rename misleading session id variable
ASoC: q6asm: make commands specific to streams
ASoC: q6asm: use flags directly from asm-dai
ASoC: q6asm: add length to write command token
ASoC: q6asm: add support to remove intial and trailing silence
ASoC: q6asm: add support to gapless flag in asm open
ASoC: q6asm-dai: add next track metadata support
ASoC: qdsp6: use dev_err instead of pr_err
ASoC: qdsp6-dai: add gapless support
ASoC: q6asm-dai: add support to copy callback
sound/soc/qcom/qdsp6/q6asm-dai.c | 397 +++++++++++++++++++++++--------
sound/soc/qcom/qdsp6/q6asm.c | 173 +++++++++-----
sound/soc/qcom/qdsp6/q6asm.h | 48 ++--
3 files changed, 458 insertions(+), 160 deletions(-)
--
2.21.0
Supports machine driver with rt1015 ("mt8183-mt6358-ts3a227-rt1015").
Embeds in existing mt8183-mt6358-ts3a227-max98357.c because they share
most of the code.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200708113233.3994206-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the error reporting more useful by adding opcode to it.
Without this its almost impossible to say which command actually
timed out.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200707163641.17113-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This is hopefully the last set of fixes to avoid probe errors due to
stricter checks of DAI capabilities introduced late in the 5.8 cycle.
Daniel Baluta (1):
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
Pierre-Louis Bossart (2):
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
ASoC: Intel: bdw-rt5677: fix non BE conversion
include/sound/soc-dai.h | 1 +
sound/soc/generic/audio-graph-card.c | 4 +--
sound/soc/generic/simple-card.c | 4 +--
sound/soc/intel/boards/bdw-rt5677.c | 1 +
sound/soc/soc-dai.c | 38 ++++++++++++++++++++++++++++
sound/soc/sof/imx/imx8.c | 8 ++++++
sound/soc/sof/imx/imx8m.c | 8 ++++++
7 files changed, 60 insertions(+), 4 deletions(-)
base-commit: a5911ac579
--
2.25.1
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac579
--
2.25.1
