This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing
v2: Fixed the build error from snd_soc_component_read32
v1: initial ASoC: codec: tlv3204: Codec workaround series
Michael Sit Wei Hong (3):
ASoC: codec: tlv3204: Enable 24 bit audio support
ASoC: codec: tlv3204: Increased maximum supported channels
ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset
sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++---------
1 file changed, 44 insertions(+), 16 deletions(-)
--
2.17.1
This series performs some minor cleanup on the driver for the analog
codec in the Allwinner A64, and hooks up the existing mute switches to
DAPM widgets, in order to provide improved power management.
Changes since v1:
- Collected Acked-by/Reviewed-by tags
- Used SOC_MIXER_NAMED_CTL_ARRAY to avoid naming a widget "Earpiece"
Samuel Holland (8):
ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits
ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend
ASoC: sun50i-codec-analog: Group and sort mixer routes
ASoC: sun50i-codec-analog: Make headphone routes stereo
ASoC: sun50i-codec-analog: Enable DAPM for headphone switch
ASoC: sun50i-codec-analog: Make line out routes stereo
ASoC: sun50i-codec-analog: Enable DAPM for line out switch
ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch
sound/soc/sunxi/sun50i-codec-analog.c | 176 ++++++++++++++++----------
1 file changed, 111 insertions(+), 65 deletions(-)
--
2.26.2
This series fixes a couple of issues with the digital audio codec in the
Allwinner A64 SoC:
1) Left/right channels were swapped when playing/recording audio
2) DAPM topology was wrong, breaking some kcontrols
This is the minimum set of changes necessary to fix these issues in a
backward-compatible way. For that reason, some DAPM widgets still have
incorrect or confusing names; those and other issues will be fixed in
later patch sets.
Samuel Holland (7):
ASoC: dt-bindings: Add a new compatible for the A64 codec
ASoC: sun8i-codec: Fix DAPM to match the hardware topology
ASoC: sun8i-codec: Add missing mixer routes
ASoC: sun8i-codec: Add a quirk for LRCK inversion
ARM: dts: sun8i: a33: Update codec widget names
arm64: dts: allwinner: a64: Update codec widget names
arm64: dts: allwinner: a64: Update the audio codec compatible
.../sound/allwinner,sun8i-a33-codec.yaml | 6 +-
arch/arm/boot/dts/sun8i-a33-olinuxino.dts | 4 +-
arch/arm/boot/dts/sun8i-a33.dtsi | 4 +-
.../dts/allwinner/sun50i-a64-bananapi-m64.dts | 8 +-
.../dts/allwinner/sun50i-a64-orangepi-win.dts | 8 +-
.../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +-
.../dts/allwinner/sun50i-a64-pinebook.dts | 8 +-
.../dts/allwinner/sun50i-a64-pinephone.dtsi | 8 +-
.../boot/dts/allwinner/sun50i-a64-pinetab.dts | 8 +-
.../allwinner/sun50i-a64-sopine-baseboard.dts | 8 +-
.../boot/dts/allwinner/sun50i-a64-teres-i.dts | 8 +-
arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 11 +-
sound/soc/sunxi/sun8i-codec.c | 137 ++++++++++++++----
13 files changed, 155 insertions(+), 71 deletions(-)
--
2.26.2
This patch series drops a printk message down to dev_dbg() because it
was noisy and then migrates this driver to use clk_hw based APIs instead
of clk based APIs because this device is a clk provider, not a clk
consumer. I've only lightly tested the last two patches but I don't have
all combinations of clks for this device.
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Stephen Boyd (3):
ASoC: rt5682: Use dev_dbg() in rt5682_clk_check()
ASoC: rt5682: Drop usage of __clk_get_name()
ASoC: rt5682: Use clk_hw based APIs for registration
sound/soc/codecs/rt5682.c | 73 ++++++++++++---------------------------
sound/soc/codecs/rt5682.h | 2 --
2 files changed, 23 insertions(+), 52 deletions(-)
Based on the last patch to this driver in linux-next.
base-commit: 6301adf942
--
Sent by a computer, using git, on the internet
simple-card.c and meson-card-utils.c use pretty much the same
helper function to parse auxiliary devices from the device tree.
Make it easier for other drivers to parse these from the device tree
as well by adding a shared helper function to soc-core.c.
snd_soc_of_parse_aux_devs() is pretty much a copy of
meson_card_add_aux_devices() from meson-card-utils.c
with two minor changes:
- Make property name configurable as parameter
- Change dev_err() message slightly for consistency with other
error messages in soc-core.c
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200801100257.22658-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
"AVDD" is for analog power supply, "DVDD" is for digital power
supply, they can improve the power management.
As the regulator is enabled in pm runtime resume, which is
behind the component driver probe, so accessing registers in
component driver probe will fail. Fix this issue by enabling
regcache_cache_only after pm_runtime_enable.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1597397561-2426-2-git-send-email-shengjiu.wang@nxp.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>
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>
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>
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>
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>
The audio codecs in the A33 and A64 are both integrated variants of the
X-Powers AC100 codec. However, there are some differences between them
that merit having a separate compatible:
- The A64 has a second DRC block, not present in the AC100 or A33.
- The A33 has some extra muxing options for AIF1/2/3 in the
AIF3_SGP_CTRL register, which are not present in the AC100 or A64.
- The A33 is missing registers providing jack detection functionality.
- The A33 is claimed to invert LRCK, but this is not seen on A64.
Since the driver will continue to work on the A64 using the A33
compatible, albeit without jack detection functionality and with
possibly inverted channels, as it does now, allow the A33 compatible
to be used as a fallback.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200726012557.38282-2-samuel@sholland.org
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>
