Fix cppcheck warnings:
sound/soc/intel/atom/sst/sst.c:427:13: style: Variable 'ret' is
assigned a value that is never used. [unreadVariable]
int i, ret = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warnings:
sound/soc/intel/atom/sst/sst.c:373:2: warning: Assignment of function
parameter has no effect outside the function. Did you forget
dereferencing it? [uselessAssignmentPtrArg]
ctx = NULL;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/atom/sst-mfld-platform-compress.c:46:14: style:
Variable 'ret_val' is assigned a value that is never
used. [unreadVariable]
int ret_val = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck complains of a possible NULL pointer dereference but setting
a pointer before using list_for_each_entry() is not useful.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some devices have broken extension unit where getting current value
doesn't work. Attempt that once when creating mixer control for it. If
it fails, just ignore it, so that it won't cripple the device entirely
(and/or make the error floods).
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Link: https://lore.kernel.org/r/5f3abc52.1c69fb81.9cf2.fe91@mx.google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously the driver would use devm_* related functions at
the codec level probe() to allocate clock resources for MCLK
and the DAI clocks exposed by the device. This caused issues
when registering clocks on a re-probe (no device level
remove/prove involved) as the devm_* resources were never
freed up so the clocks were still registered from the previous
codec level probe().
This commit updates the clock handling for MCLK usage and DAI
clock provision to fix this discrepancy and allow the codec level
probe/remove functionality to operate as intended.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/b92c461baeed27a6cd92e59e36a55c2547218683.1597164865.git.Adam.Thomson.Opensource@diasemi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As part of the reorganisation of the device level and codec
level probe functionlity, the soft reset handling should really
reside at the codec level and after the instantiation of supplies.
This commit makes the relevant changes to support this change of
scope including the remove of devm_* functions being called for
regulator instantiation at the codec level.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/f7603a4855647429b754ce76f887ec441622015c.1597164865.git.Adam.Thomson.Opensource@diasemi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During gapless playback, its possible for previous track to
end at unaligned boundary, starting next track on the same
boundary can lead to unaligned address exception in dsp.
So implement copy callback for finer control on the buffer offsets.
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/20200727093806.17089-11-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to gapless playback by implementing metadata,
next_track, drain and partial drain support.
Gapless on Q6ASM is implemented by opening 2 streams in a single
q6asm stream and toggling them on next track.
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/20200727093806.17089-10-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
rearrange code so that it will be easy to change the codec
profile at runtime. This means moving exiting set_params
to an internal wrapper which can be called when codec
profile changes.
This is also preparing the code for easy to use in gapless cases.
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/20200727093806.17089-9-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to metadata required to do a gapless playback.
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/20200727093806.17089-8-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to gapless flag to q6asm_open_write().
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/20200727093806.17089-7-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to ASM_DATA_CMD_REMOVE_INITIAL_SILENCE
and ASM_DATA_CMD_REMOVE_TRAILING_SILENCE q6asm command to support
compressed metadata for gapless playback.
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/20200727093806.17089-6-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add length to write command packet token so that we can track exactly
how many bytes are consumed by DSP in the command reply.
This is useful in some use-cases where the end of the file/stream
is not aligned with period size.
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/20200727093806.17089-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
use flags set by q6asm-dais directly!
This will be useful gapless case where write needs a special flag to indicate
that last buffer.
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/20200727093806.17089-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
As the recent fix addressed the channel swap problem more properly,
update the comment as well.
Fixes: 1b7ecc241a ("ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109")
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200816084431.102151-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same
interface. This was not supported by the composite quirk back in the day
when initial support for this device was added, thus only playback was
enabled until now.
Fixes: 11e424e88b ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB")
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable.vger.kernel.org>
Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All device-specific small fixes and quirks mostly for usual
suspects, USB-audio and HD-audio.
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Merge tag 'sound-fix-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All device-specific small fixes and quirks mostly for usual suspects,
USB-audio and HD-audio"
* tag 'sound-fix-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: echoaudio: Fix potential Oops in snd_echo_resume()
ALSA: hda/hdmi: Use force connectivity quirk on another HP desktop
ALSA: hda/realtek - Fix unused variable warning
ALSA: hda - reverse the setting value in the micmute_led_set
ALSA: echoaduio: Drop superfluous volatile modifier
ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control
ALSA: usb-audio: add quirk for Pioneer DDJ-RB
ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109
ALSA: hda - fix the micmute led status for Lenovo ThinkCentre AIO
ALSA: usb-audio: fix overeager device match for MacroSilicon MS2109
ALSA: hda/realtek: Fix pin default on Intel NUC 8 Rugged
ALSA: usb-audio: Creative USB X-Fi Pro SB1095 volume knob support
ALSA: usb-audio: fix spelling mistake "buss" -> "bus"
Freeing chip on error may lead to an Oops at the next time
the system goes to resume. Fix this by removing all
snd_echo_free() calls on error.
Fixes: 47b5d028fd ("ALSA: Echoaudio - Add suspend support #2")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20200813074632.17022-1-dinghao.liu@zju.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's another HP desktop has buggy BIOS which flags the Port
Connectivity bit as no connection.
Apply force connectivity quirk to enable DP/HDMI audio.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200811095336.32396-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous fix forgot to remove the unused variable that triggers a
compile warning now:
sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led':
sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable]
Fix it.
Fixes: 404690649e ("ALSA: hda - reverse the setting value in the micmute_led_set")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Link: https://lore.kernel.org/r/20200812070256.32145-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Before the micmute_led_set() is introduced, the function of
alc_gpio_micmute_update() will set the gpio value with the
!micmute_led.led_value, and the machines have the correct micmute led
status. After the micmute_led_set() is introduced, it sets the gpio
value with !!micmute_led.led_value, so the led status is not correct
anymore, we need to set micmute_led_polarity = 1 to workaround it.
Now we fix the micmute_led_set() and remove micmute_led_polarity = 1.
Fixes: 87dc36482c ("ALSA: hda/realtek - Add LED class support for micmute LED")
Reported-and-suggested-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200811122430.6546-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dsp_registers field of struct echoaduio has the volatile modifier,
but it's basically superfluous; the field is accessed only for the
base pointer of readl() and writel(), hence marking with __iomem alone
should suffice. OTOH, having the volatile prefix causes a compile
warning like:
sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers]
So it's better to drop this superfluous modifier.
Link: https://lore.kernel.org/r/20200803143958.24324-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel
line-in claim to support volume control, but it doens't seem to have an
AMP, so when line-in volume lowers below 80, nothing gets recorded
anymore.
Disable the volume control to workaround the issue.
Fixes: f8c11eb7da ("ALSA: usb-audio: Add support for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200810133108.31580-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After installing the Ubuntu Linux, the micmute led status is not
correct. Users expect that the led is on if the capture is disabled,
but with the current kernel, the led is off with the capture disabled.
We tried the old linux kernel like linux-4.15, there is no this issue.
It looks like we introduced this issue when switching to the led_cdev.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200810021659.7429-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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'
...
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a spelling mistake in a usb_audio_dbg debug message. Also
replace "param" with "parameter". Fix these.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tegra HDA has audio data buffer for upto tens of frames, this buffer
can help to avoid underflow. HW will keep issuing new data fetch
request when buffers are not full and current BDL is not done. When SW
disable DMA RUN bit for a stream, HW can't cancel the already issued data
fetch request and hence it can't stop DMA. HW has to wait for all issued
data fetch request get data returned before it stops DMA.
This HW behavior is not in sync with HDA spec which says DMA RUN bit
should be cleared within 1 audio frame. For Tegra, DMA RUN bit was
active for more than one audio frame, due to this the timeout in
snd_hdac_stream_sync function is not helping. When Stream reset set
and clear happens during DMA RUN bit active state it results in Memory
Decode error.
Unfortunately, there is no way to detect when these data accesses have
completed, but testing has shown that a 100us delay between Stream reset
set and clear operation for Tegra avoids the memory decode error.
Therefore, adding a 100us dma stop delay.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-4-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A variable dma_stop_delay is added as a new member in hdac_bus
structure to avoid memory decode error incase DMA RUN bit is not
disabled in the given timeout from snd_hdac_stream_sync function and
followed by stream reset which results in memory decode error between
reset set and clear operation.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-3-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set chip->align_buffer_size to 1 for Tegra platforms to make the buffer
alignment to be multiple of 128 bytes. This fix is applied as gstreamer
alsasink gets stuck with the default buffer-time and latency-time
parameters with 4 byte buffer alignment.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ioctls via OSS sequencer API may race and lead to UAF when the
port create and delete are performed concurrently, as spotted by a
couple of syzkaller cases. This patch is an attempt to address it by
serializing the ioctls with the existing register_mutex.
Basically OSS sequencer API is an obsoleted interface and was designed
without much consideration of the concurrency. There are very few
applications with it, and the concurrent performance isn't asked,
hence this "big hammer" approach should be good enough.
Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com
Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com
Suggested-by: Hillf Danton <hdanton@sina.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Prepare for tasklet API modernization (Romain Perier, Allen Pais, Kees Cook)
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Merge tag 'tasklets-v5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/kees/linux
Pull tasklets API update from Kees Cook:
"These are the infrastructure updates needed to support converting the
tasklet API to something more modern (and hopefully for removal
further down the road).
There is a 300-patch series waiting in the wings to get set out to
subsystem maintainers, but these changes need to be present in the
kernel first. Since this has some treewide changes, I carried this
series for -next instead of paining Thomas with it in -tip, but it's
got his Ack.
This is similar to the timer_struct modernization from a while back,
but not nearly as messy (I hope). :)
- Prepare for tasklet API modernization (Romain Perier, Allen Pais,
Kees Cook)"
* tag 'tasklets-v5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/kees/linux:
tasklet: Introduce new initialization API
treewide: Replace DECLARE_TASKLET() with DECLARE_TASKLET_OLD()
usb: gadget: udc: Avoid tasklet passing a global
HDMI on some platforms doesn't enable audio support because its Port
Connectivity [31:30] is set to AC_JACK_PORT_NONE:
Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0b000094: OUT Detect HBR HDMI DP
Pin Default 0x58560010: [N/A] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D3 EPSS
Power: setting=D0, actual=D0
Devices: 0
Connection: 3
0x02 0x03* 0x04
For now, use a quirk to force connectivity based on SSID. If there are
more platforms affected by the same issue, we can eye for a more generic
solution.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name
and shows up currently as "USB Sound Device" in ALSA.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20200804010616.3399256-1-cujomalainey@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with
two USB audio cards.
USB device (17aa:104d) provides functionality for Internal Speaker and
Front Headset. It's UAC v2, so it supports insertion control (jack
detection). However, when trying to get the connector status of the
speaker, an error occurs:
[ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0
Since the insertion control works perfectly for the headset, the error
for speaker is probably casued by connecting internally. So let's relax
the error for a bit if it's a speaker, and always reports it's connected.
USB device (17aa:1046) is for rear Line-in, Line-out and Microphone.
The insertion control works for all three jacks. However, there's an
Function Unit that doesn't work:
[ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4
[ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19)
So turn off the FU to avoid the error.
Also, add specific card name for both devices, so userspace can easily
indentify both cards.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200803142612.17156-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
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Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.9
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
This reverts commit 9a6418487b ("ALSA: hda: call runtime_allow()
for all hda controllers").
The reverted patch already introduced some regressions on some
machines:
- on gemini-lake machines, the error of "azx_get_response timeout"
happens in the hda driver.
- on the machines with alc662 codec, the audio jack detection doesn't
work anymore.
Fixes: 9a6418487b ("ALSA: hda: call runtime_allow() for all hda controllers")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200803064638.6139-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ca0113 command had the wrong group_id, 0x48 when it should've been
0x30. The front microphone selection should now work.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-1-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1596360400-32425-1-git-send-email-chenhc@lemote.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
A few wrap-up small fixes for the usual HD-audio and USB-audio stuff:
- A regression fix for S3 suspend on old Intel platforms
- A fix for possible Oops in ASoC HD-audio binding
- Trivial quirks for various devices
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Merge tag 'sound-5.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few wrap-up small fixes for the usual HD-audio and USB-audio stuff:
- A regression fix for S3 suspend on old Intel platforms
- A fix for possible Oops in ASoC HD-audio binding
- Trivial quirks for various devices"
* tag 'sound-5.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fixed HP right speaker no sound
ALSA: hda: fix NULL pointer dereference during suspend
ALSA: hda/hdmi: Fix keep_power assignment for non-component devices
ALSA: hda: Workaround for spurious wakeups on some Intel platforms
ALSA: hda/realtek: Fix add a "ultra_low_power" function for intel reference board (alc256)
ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289
ALSA: usb-audio: Add implicit feedback quirk for SSL2
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>
This converts all the existing DECLARE_TASKLET() (and ...DISABLED)
macros with DECLARE_TASKLET_OLD() in preparation for refactoring the
tasklet callback type. All existing DECLARE_TASKLET() users had a "0"
data argument, it has been removed here as well.
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Kees Cook <keescook@chromium.org>
HP NB right speaker had no sound output.
This platform was connected to I2S Amp for speaker out.(None Realtek I2S Amp IC)
EC need to check codec GPIO1 pin to initial I2S Amp.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/01285f623ac7447187482fb4a8ecaa7c@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
When the ASoC card registration fails and the codec component driver
never probes, the codec device is not initialized and therefore
memory for codec->wcaps is not allocated. This results in a NULL pointer
dereference when the codec driver suspend callback is invoked during
system suspend. Fix this by returning without performing any actions
during codec suspend/resume if the card was not registered successfully.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200728231011.1454066-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
It's been reported that, when neither nouveau nor Nvidia graphics
driver is used, the screen starts flickering. And, after comparing
between the working case (stable 4.4.x) and the broken case, it turned
out that the problem comes from the audio component binding. The
Nvidia and AMD audio binding code clears the bus->keep_power flag
whenever snd_hdac_acomp_init() succeeds. But this doesn't mean that
the component is actually bound, but it merely indicates that it's
ready for binding. So, when both nouveau and Nvidia are blacklisted
or not ready, the driver keeps running without the audio component but
also with bus->keep_power = false. This made the driver runtime PM
kicked in and powering down when unused, which results in flickering
in the graphics side, as it seems.
For fixing the bug, this patch moves the bus->keep_power flag change
into generic_acomp_notifier_set() that is the function called from the
master_bind callback of component ops; i.e. it's guaranteed that the
binding succeeded.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208609
Fixes: 5a858e79c9 ("ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200728082033.23933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've received a regression report on Intel HD-audio controller that
wakes up immediately after S3 suspend. The bisection leads to the
commit c4c8dd6ef8 ("ALSA: hda: Skip controller resume if not
needed"). This commit replaces the system-suspend to use
pm_runtime_force_suspend() instead of the direct call of
__azx_runtime_suspend(). However, by some really mysterious reason,
pm_runtime_force_suspend() causes a spurious wakeup (although it calls
the same __azx_runtime_suspend() internally).
As an ugly workaround for now, revert the behavior to call
__azx_runtime_suspend() and __azx_runtime_resume() for those old Intel
platforms that may exhibit such a problem, while keeping the new
standard pm_runtime_force_suspend() and pm_runtime_force_resume()
pair for the remaining chips.
Fixes: c4c8dd6ef8 ("ALSA: hda: Skip controller resume if not needed")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200727164443.4233-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401.
Fixes: ff53664daf ("ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289")
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140837.302763-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for headset mic to the ASUS ROG Zephyrus
G15(GA502) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140616.298892-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
