This series adds audio graph based sound card support for Tegra210
platforms like Jetson-TX1 an Jetson-Nano. The following preparatory
audio graph enhancement series is already merged.
* https://patchwork.kernel.org/project/alsa-devel/list/?series=375629&state=*
Following are the summary of changes:
* Add graph/audio-graph based schemas or schema updates for Tegra210
component and machine drivers.
* Add Tegra audio graph machine driver.
* Add required DT support for Jetson-TX1/Nano.
This work is based on earlier discussion of DPCM usage for Tegra
and simple card driver updates.
* https://lkml.org/lkml/2020/4/30/519
* https://lkml.org/lkml/2020/6/27/4
Original v6 series was sent about 6-7 weeks back. The dependency commit,
https://lore.kernel.org/alsa-devel/1610948585-16286-1-git-send-email-spujar@nvidia.com/
is now merged. Resending this now to appear in the top of the mail list.
Changelog
=========
v5 -> v6
--------
* Added ports or port description in YAML docs for Tegra AHUB
devices and graph card in patch 1/6 and 2/6. Reference of
audio-graph-port.yaml is used for AHUB devices.
* Dropped redundant NULL check return for of_device_get_match_data()
in patch 3/6.
* Added 'Reviewed-by' tag from Jon Hunter.
* No changes in remaining patches.
v4 -> v5
--------
* Audio graph related changes were sent in separate v5 series as
mentioned above and are dropped from current series.
* Graph and audio graph doc patches are dropped from this series
and are sent separately as mentioned above.
* Minor change with phandle label for TX1 and Nano platform DT files.
* No changes in other patches.
v3 -> v4
--------
* Added new patches to convert graph.txt and audio-graph-card.txt
to corresponding json-schema files. Later these references
are used in Tegra audio graph schema.
* AHUB component binding docs are updated to reflect the usage
of ports/port/endpoint
* More common stuff is moved into graph_parse_of() and this is
used by both generic and Tegra audio graph.
* DT binding for Tegra audio graph is updated to included "ports { }"
* As per the suggestion 'void *data' member is dropped from
'asoc_simple_priv' and instead container method is used to
maintain required custom data internal to Tegra audio graph.
v2 -> v3
--------
* Dropped new compatible addition in generic graph driver
after reviewing it with Morimoto-san. Instead added Tegra
audio graph driver and new compatibles are added in the same.
* Added new patches to expose new members for customization
in audio graph driver.
* Added new patch for Tegra audio graph driver and related
documentation.
* Minor change in below commit where mutex version of helper is used
"ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM"
* DT binding is updated to use the newly exposed compatibles
* No changes in other patches
v1 -> v2
--------
* Re-organized ports/endpoints description for ADMAIF and XBAR.
Updated DT patches accordingly.
* After above change, multiple Codec endpoint support is not
required and hence dropped for now. This will be considered
separately if at all required in future.
* Re-ordered patches in the series.
Sameer Pujar (6):
ASoC: dt-bindings: tegra: Add graph bindings
ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card
ASoC: tegra: Add audio graph based card driver
arm64: defconfig: Enable Tegra audio graph card driver
arm64: tegra: Audio graph header for Tegra210
arm64: tegra: Audio graph sound card for Jetson Nano and TX1
.../sound/nvidia,tegra-audio-graph-card.yaml | 187 +++++++++++++++
.../bindings/sound/nvidia,tegra186-dspk.yaml | 18 +-
.../bindings/sound/nvidia,tegra210-admaif.yaml | 13 +-
.../bindings/sound/nvidia,tegra210-ahub.yaml | 13 +-
.../bindings/sound/nvidia,tegra210-dmic.yaml | 18 +-
.../bindings/sound/nvidia,tegra210-i2s.yaml | 18 +-
.../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++++++++
arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 +++++++++++++++++++++
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++++++++
arch/arm64/configs/defconfig | 1 +
sound/soc/tegra/Kconfig | 9 +
sound/soc/tegra/Makefile | 2 +
sound/soc/tegra/tegra_audio_graph_card.c | 251 ++++++++++++++++++++
13 files changed, 1085 insertions(+), 6 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml
create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi
create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c
--
2.7.4
ASoC: cpcap: Implement set_tdm_slot for voice call support
For using cpcap for voice calls, we need to route audio directly from
the modem to cpcap for TDM (Time Division Multiplexing). The voice call
is direct data between the modem and cpcap with no CPU involvment. In
this mode, the cpcap related audio mixer controls work for the speaker
selection and volume though.
To do this, we need to implement standard snd_soc_dai_set_tdm_slot()
for cpcap. Then the modem codec driver can use snd_soc_dai_set_sysclk(),
snd_soc_dai_set_fmt(), and snd_soc_dai_set_tdm_slot() to configure a
voice call.
Let's add cpcap_voice_set_tdm_slot() for this, and cpcap_voice_call()
helper to configure the additional registers needed for voice call.
Let's also clear CPCAP_REG_VAUDIOC on init in case we have the bit for
CPCAP_BIT_VAUDIO_MODE0 set on init.
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Link: https://lore.kernel.org/r/20210112174704.GA13496@duo.ucw.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Tegra audio machine driver which is based on generic audio graph card
driver. It re-uses most of the common stuff from audio graph driver and
uses the same DT binding. Required Tegra specific customizations are done
in the driver and additional DT bindings are required for clock handling.
Details on the customizations done:
- Update PLL rates at runtime: Tegra HW supports multiple sample rates
(multiples of 8x and 11.025x) and both of these groups require different
PLL rates. Hence there is a requirement to update this at runtime.
This is achieved by providing a custom 'snd_soc_ops' and in hw_param()
callback PLL rate is updated as per the sample rate.
- Internal structure 'tegra_audio_graph_data' is used to maintain clock
handles of PLL.
- The 'force_dpcm' flag is set to use DPCM for all DAI links.
- The 'component_chaining' flag is set to use DPCM with component model.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/1611048496-24650-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The timeout for an individual transaction w/ the Cadence IP is the same as
the entire resume operation for codecs.
This doesn't make sense, we need to have at least one order of magnitude
between individual transactions and the entire resume operation.
Set the timeout on the Cadence side to 500ms and 5s for the codec resume.
Both ASoC and SoundWire trees are fine for this series.
Pierre-Louis Bossart (2):
ASoC: codecs: soundwire: increase resume timeout
soundwire: cadence: reduce timeout on transactions
drivers/soundwire/cadence_master.c | 2 +-
sound/soc/codecs/max98373-sdw.c | 4 +++-
sound/soc/codecs/rt1308-sdw.c | 2 +-
sound/soc/codecs/rt5682.h | 2 +-
sound/soc/codecs/rt700-sdw.c | 2 +-
sound/soc/codecs/rt711-sdw.c | 2 +-
sound/soc/codecs/rt715-sdw.c | 2 +-
7 files changed, 9 insertions(+), 7 deletions(-)
--
2.17.1
Here's some minor code cleanups for the lpass-cpu driver. I noticed that
it casts away const from the driver data from DT. That's not great but
fixing it is a little more involved. I'll get to it later. There's also
some questionable clk_get() usage that should probably be
clk_get_optional(). For now this should help a little.
Cc: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Cc: Srinivasa Rao <srivasam@codeaurora.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Stephen Boyd (4):
ASoC: qcom: Remove useless debug print
ASoC: qcom: Add some names to regmap configs
ASoC: qcom: Stop casting away __iomem for error pointers
ASoC: qcom: Remove duplicate error messages on ioremap
sound/soc/qcom/lpass-cpu.c | 17 ++++++-----------
1 file changed, 6 insertions(+), 11 deletions(-)
base-commit: 5c8fe583cc
--
https://chromeos.dev
Hi Mark
These are not so important, but for
soc-pcm cleanup patches.
Kuninori Morimoto (6):
ASoC: soc-pcm: move dpcm_set_fe_update_state()
ASoC: soc-pcm: add dpcm_set_be_update_state()
ASoC: soc-pcm: add soc_pcm_set_dai_params()
ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry()
ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()
ASoC: soc-pcm: setup pcm at one place in soc_new_pcm()
sound/soc/soc-pcm.c | 231 +++++++++++++++++---------------------------
1 file changed, 90 insertions(+), 141 deletions(-)
--
2.25.1
Thank you for your help !!
Best regards
---
Kuninori Morimoto
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20210106133650.13509-3-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20210106133650.13509-2-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20210106133650.13509-1-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment it is necessary to set up the DAPM routes between
front-end AIF<->DAI explicitly in the device tree, e.g. using
audio-routing =
"MM_DL1", "MultiMedia1 Playback",
"MM_DL3", "MultiMedia3 Playback",
"MM_DL4", "MultiMedia4 Playback",
"MultiMedia2 Capture", "MM_UL2";
This is prone to mistakes and (sadly) there is no clear error if one
of these routes is missing. :(
Actually, this should not be necessary because the ASoC core normally
automatically links AIF<->DAI within snd_soc_dapm_link_dai_widgets().
This is done using the "stname" parameter of SND_SOC_DAPM_AIF_IN/OUT.
For SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
it should create the route from above: MM_DL1 <-> MultiMedia1 Playback.
This does not work at the moment because the AIF widget (MM_DL1)
and the DAI widget (MultiMedia1 Playback) belong to different
DAPM contexts (q6routing / q6asm-dai).
Fix this by declaring the AIF widgets in the same driver as the DAIs
(q6asm-dai). Now the routes above are created automatically
and no longer need to be specified in the device tree.
This is also more consistent with the back-end AIFs which are already
declared in q6afe-dais instead of q6routing. q6routing should only link
the components together using mixers.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20201211203255.148246-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
This series adds support for devices with only a headphone jack
(no speakers/internal mic). Specifically this adds support for the
Mele PCG03 Mini PC. But the new no-speakers and no-internal-mic quirks
will likely be useful on other devices too.
Regards,
Hans
sound/soc/soc-core.c: soc_remove_component() unconditionally calls
snd_soc_component_set_jack(component, NULL, NULL); on any components
being removed.
This means that on machines where the machine-driver does not provide
a jack through snd_soc_component_set_jack() es8316_disable_jack_detect()
will still get called and at this time es8316->jack will be NULL and
the es8316->jack->status check in es8316_disable_jack_detect() will
lead to a NULL pointer deref.
Fix this by checking for es8316->jack bein NULL at the start of
es8316_disable_jack_detect() and turn the function into a no-op in
that case.
Cc: russianneuromancer <russianneuromancer@ya.ru>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210112101725.44200-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the snd_soc_dai_set_tdm_slot() call from cht_codec_init() to
cht_codec_fixup(). There are 2 reasons for doing this:
1. This aligns the cht_bsw_nau8824 with all the other BYT/CHT machine
drivers which also do this from their codec_fixup function.
2. When using the SOF driver, things like the TDM info is set from the
topology file. Moving the call to the codec_fixup function, which gets
skipped when using the SOF driver avoids the call interfering with the
settings when using the SOF driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210107115324.11602-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
