component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_set_params()
to soc-component as snd_soc_component_compr_set_params().
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_trigger()
to soc-component as snd_soc_component_compr_trigger().
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_free()
to soc-component as snd_soc_component_compr_free().
component related function should be implemented at
soc-component.c.
This patch moves soc-compress soc_compr_components_open()
to soc-component as snd_soc_component_compr_open().
ASoC: intel: SND_SOC_INTEL_KEEMBAY should depend on ARCH_KEEMBAY
The Intel Keem Bay audio module is only present on Intel Keem Bay SoCs.
Hence add a dependency on ARCH_KEEMBAY, to prevent asking the user about
this driver when configuring a kernel without Intel Keem Bay platform
support.
ASoC: fsl: SND_SOC_FSL_AUD2HTX should depend on ARCH_MXC
The Freescale/NXP AUDIO TO HDMI TX module is only present on NXP i.MX 8
Series SoCs. Hence add a dependency on ARCH_MXC, to prevent asking the
user about this driver when configuring a kernel without i.MX 8 platform
support.
Bogdan Togorean [Tue, 10 Nov 2020 15:22:13 +0000 (17:22 +0200)]
ASoc: adi: Kconfig: Remove depends on for ADI reference designs
Audio ADI reference designs are also used on some ZynqMP boards, and can
also be used on Intel FPGA boards and also on some more complex FPGA
combinations (FPGA cards connected through PCIe).
This change removes the dependency on Microblaze and Zynq architectures
to allow the usage of this driver for the systems described above.
Mark Brown [Mon, 16 Nov 2020 23:28:03 +0000 (23:28 +0000)]
Merge series "ASoC: Intel: catpt: Offload fixes and code optimization" from Cezary Rojewski <cezary.rojewski@intel.com>:
First two of the series address bugs connected mainly to offload streams:
- scenarios with very low buffer sizes: RESET_STREAM IPC timeouts
- fix lp clock selection when switching between PAUSE <-> RESUME states:
glitches on first offload when no additional stream is opened
simultaneously
Follow ups are: code reduction and optimization oriented patches.
This has been foretold in:
[PATCH v10 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point
https://www.spinics.net/lists/alsa-devel/msg116440.html
Note: LPT power up/down sequences might get aligned with WPT once
enough testing is done as capabilities are shared for both DSPs.
First, optimize applying of user settings - prevent redundand calls from
happening - and then as mentioned above, streamline power on/off sequence
for LPT and WPT.
Cezary Rojewski (5):
ASoC: Intel: catpt: Skip position update for unprepared streams
ASoC: Intel: catpt: Correct clock selection for dai trigger
ASoC: Intel: catpt: Optimize applying user settings
ASoC: Intel: catpt: Streamline power routines across LPT and WPT
ASoC: Intel: catpt: Cleanup after power routines streamlining
Mark Brown [Mon, 16 Nov 2020 23:23:48 +0000 (23:23 +0000)]
Merge series "ASoC: pcm512x: Patch series to set fmt from `set_fmt()`" from Kirill Marinushkin <kmarinushkin@birdec.com>:
Set format from `set_fmt()` func instead of `hw_params()`, plus supportive
commits
Kirill Marinushkin (4):
ASoC: pcm512x: Fix not setting word length if DAIFMT_CBS_CFS
ASoC: pcm512x: Rearrange operations in `hw_params()`
ASoC: pcm512x: Move format check into `set_fmt()`
ASoC: pcm512x: Add support for more data formats
ASoC: qcom: lpass-sc7180: Add 32 bit format support for capture
Add 32 bit format support for capture in lpass-sc7180
snd_soc_dai_driver capabilities. Need to add contstraints
in machine driver so that only specific format allowed.
Jerome Brunet [Mon, 16 Nov 2020 17:24:23 +0000 (18:24 +0100)]
ASoC: meson: fix COMPILE_TEST error
When compiled with CONFIG_HAVE_CLK, the kernel need to get provider for the
clock API. This is usually selected by the platform and the sound drivers
should not really care about this. However COMPILE_TEST is special and the
platform required may not have been selected, leading to this type of
error:
> aiu-encoder-spdif.c:(.text+0x3a0): undefined reference to `clk_set_parent'
Since we need a sane provider of the API with COMPILE_TEST, depends on
COMMON_CLK.
Fixes: 97799c939d89 ("ASoC: meson: add axg fifo base driver") Reported-by: kernel test robot <lkp@intel.com> Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20201116172423.546855-1-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie@kernel.org>
Cezary Rojewski [Mon, 16 Nov 2020 13:33:31 +0000 (14:33 +0100)]
ASoC: Intel: catpt: Streamline power routines across LPT and WPT
There is no need for separate power on/off routines for LPT and WPT as
as the protocol is shared for both platforms. Make WPT routines generic
and reuse them in LPT case too.
Cezary Rojewski [Mon, 16 Nov 2020 13:33:30 +0000 (14:33 +0100)]
ASoC: Intel: catpt: Optimize applying user settings
Initial user settings such as volume control need to be applied only
once after stream is allocated. As prepare() operation can be invoked
multiple times during the stream's lifetime, relocate
catpt_dai_apply_usettings() and call it directly within
catpt_dai_hw_params() rather than on every catpt_dai_prepare().
Cezary Rojewski [Mon, 16 Nov 2020 13:33:29 +0000 (14:33 +0100)]
ASoC: Intel: catpt: Correct clock selection for dai trigger
During stream start DSP firmware requires LPCS disabled as that moment in
time is resource heavy. Currently high-clock is selected on start of
second stream onwards while low-clock is re-selected before stream
actually leaves RESUME state i.e. PAUSE_STREAM call. Fix this by always
updating clock before RESUME_STREAM and directly after PAUSE_STREAM.
Cezary Rojewski [Mon, 16 Nov 2020 13:33:28 +0000 (14:33 +0100)]
ASoC: Intel: catpt: Skip position update for unprepared streams
Playing with very low period sizes may lead to timeouts when awaiting
RESET_STREAM reply for offload streams. This is caused by NOTIFY_POSITION
appearing in the middle of trigger(stop).
Stream is unprepared during trigger(stop) where PAUSE_STREAM IPC gets
invoked. However, all data that is already mixed in DSP firmware's mixer
stream will still be played regardless of the pause. For offload streams,
this means possibility for another NOTIFY_POSITION to process. Keep these
notifications in check by only handling them when stream is in prepared
state.
Currently, pcm512x driver supports only I2S data format.
This commit adds RJ, LJ, DSP_A and DSP_B as well.
I don't expect regression WRT existing sound cards, because:
* default value in corresponding register of pcm512x codec is 0 == I2S
* existing in-tree sound cards with pcm512x codec are configured for I2S
* i don't see how existing off-tree sound cards with pcm512x codec could be
configured differently - it would not work
* tested explicitly, that there is no regression with Raspberry Pi +
sound card `sound/soc/bcm/hifiberry_dacplus.c`
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com> Cc: Mark Brown <broonie@kernel.org> Cc: Takashi Iwai <tiwai@suse.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Matthias Reichl <hias@horus.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Peter Ujfalusi <peter.ujfalusi@ti.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20201115122306.18164-5-kmarinushkin@birdec.com Signed-off-by: Mark Brown <broonie@kernel.org>
I would like to describe the reasoning by quoting Peter Ujfalusi
<peter.ujfalusi@ti.com> from his review of this patch series v1 [1]:
> When you bind a link you will use set_fmt for the two sides to see if
> they can agree, that both can support what has been asked.
>
> The pcm512x driver just saves the fmt and say back to that card:
> whatever, I'm fine with it. But runtime during hw_params it can fail due
> to unsupported bus format, which it actually acked to be ok.
>
> This is the difference.
>
> Sure, some device have constraint based on the fmt towards the hw_params
> and it is perfectly OK to do such a checks and rejections or build
> rules/constraints based on fmt, but failing hw_params just because
> set_fmt did not checked that the bus format is not even supported is not
> a nice thing to do.
ASoC: pcm512x: Rearrange operations in `hw_params()`
This commit is a preparation for the next patch in the series.
It's goal is to make format check easy-to-move-out. Theoretically, more
butifications are possile in `hw_params()` func, but my intention in this
commit is to keep behaviour unchanged.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com> Cc: Mark Brown <broonie@kernel.org> Cc: Takashi Iwai <tiwai@suse.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Matthias Reichl <hias@horus.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Peter Ujfalusi <peter.ujfalusi@ti.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20201115122306.18164-3-kmarinushkin@birdec.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: pcm512x: Fix not setting word length if DAIFMT_CBS_CFS
In `pcm512x_hw_params()`, the following switch-case:
~~~~
switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
~~~~
returns 0, which was preventing word length from being written into codecs
register.
Fixed by writing it into register before checking
`SND_SOC_DAIFMT_MASTER_MASK`.
Tested with Raspberry Pi + sound card `hifiberry_dacplus` in CBS_CFS format
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com> Cc: Mark Brown <broonie@kernel.org> Cc: Takashi Iwai <tiwai@suse.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Matthias Reichl <hias@horus.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Peter Ujfalusi <peter.ujfalusi@ti.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20201115122306.18164-2-kmarinushkin@birdec.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Fri, 13 Nov 2020 16:04:13 +0000 (16:04 +0000)]
Merge series "ASoC: use inclusive language for bclk/fsync/topology" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
The SOF (Sound Open Firmware) tree contains a lot of references in
topology files to 'codec_slave'/'codec_master' terms, which in turn
come from alsa-lib and ALSA/ASoC topology support at the kernel
level. These terms are no longer compatible with the guidelines
adopted by the kernel community [1], standard organizations, and need
to change in backwards-compatible ways.
The main/secondary terms typically suggested in guidelines don't mean
anything for clocks, this patchset suggests instead the use of
'provider' and 'consumer' terms, with the 'codec' prefix kept to make
it clear that the codec is the reference. The CM/CS suffixes are also
replaced by CP/CC.
It can be argued that the change of suffix is invasive, but finding a
replacement that keeps the M and S shortcuts has proven difficult in
quite a few contexts.
The previous definitions are kept for backwards-compatibility so this
change should not have any functional impact. It is suggested that new
contributions only use the new terms but there is no requirement to
transition immediately to the new definitions for existing code. Intel
will however update all its past contributions related to bit
clock/frame sync configurations immediately.
This patchset contains the kernel changes only, the alsa-lib changes
were shared separately.
Feedback welcome
~Pierre
[1] https://lkml.org/lkml/2020/7/4/229
Pierre-Louis Bossart (4):
ASoC: topology: use inclusive language for bclk and fsync
ASoC: SOF: use inclusive language for bclk and fsync
ASoC: Intel: atom: use inclusive language for SSP bclk/fsync
ASoC: Intel: keembay: use inclusive language for bclk and fsync
Fabio Estevam [Tue, 10 Nov 2020 20:39:34 +0000 (17:39 -0300)]
ASoC: phycore-ac97: Remove unused driver
Since commit d2ca8eac9410 ("ARM: imx: Remove i.MX35 board files"), the
MACH_PCM043 and MACH_PCA100 non-DT platform are no longer supported,
so get rid of their machine audio driver too.
Fabio Estevam [Tue, 10 Nov 2020 20:39:33 +0000 (17:39 -0300)]
ASoC: mx27vis-aic32x4: Remove unused driver
Since commit 817e5c9e585f ("ARM: imx: Remove i.MX27 board files"), the
MACH_IMX27_VISSTRIM_M10 non-DT platform is no longer supported,
so get rid of its machine audio driver too.
Fabio Estevam [Tue, 10 Nov 2020 20:39:32 +0000 (17:39 -0300)]
ASoC: wm1133-ev1: Remove unused driver
Since commit 76d00d44de6a ("ARM: imx: Remove i.MX31 board files"), the
MACH_MX31ADS_WM1133_EV1 non-DT platform is no longer supported,
so get rid of its machine audio driver too.
ASoC: SOF: Intel: fix Kconfig dependency for SND_INTEL_DSP_CONFIG
SND_INTEL_DSP_CONFIG is selected by the HDaudio, Skylake and SOF
drivers. When the HDaudio link is not selected as a option, this
Kconfig option is not touched and will default to whatever other
drivers selected. In the case e.g. where HDaudio is compiled as
built-in, the linker will complain:
ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe':
sof-pci-dev.c:(.text+0x5c): undefined reference to
`snd_intel_dsp_driver_probe'
Adding the select for all HDaudio platforms, regardless of whether
they rely on the HDaudio link or not, solves the problem.
Reported-by: Randy Dunlap <rdunlap@infradead.org> Acked-by: Randy Dunlap <rdunlap@infradead.org> Fixes: 573ce1806025c ('ALSA: hda: add Intel DSP configuration / probe code') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20201112164425.25603-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Vehmanen [Wed, 11 Nov 2020 17:33:21 +0000 (19:33 +0200)]
ASoC: SOF: loader: do not warn about unknown firmware headers
The firmware extended data IPC and manifest structures are designed to
be extendable without breaking the driver-firmware ABI. Driver should
not raise a warning in case a new header type is detected at
firmware boot. There are already checks for IPC ABI compatibility in
snd_sof_ipc_valid() and if the versions are deemed compatible, extra
fields in IPC messages should not trigger warnings.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Karol Trzciński <karolx.trzcinski@linux.intel.com> Link: https://lore.kernel.org/r/20201111173321.1933452-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Dharageswari R [Mon, 9 Nov 2020 21:09:58 +0000 (13:09 -0800)]
ASoC: Intel: Boards: tgl_max98373: add dpcm_capture flag for speaker_smart_amp
Smart_amp_speaker device has the playback stream and capture stream
associated to it. Hence add the dpcm_capture = 1 flag while dailink
creation.
This patches fixes:
ERR kernel [timestamp] SSP1-Codec: ASoC: no backend capture stream
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Dharageswari R <dharageswari.r@intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20201109210958.84198-1-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck complains, use separate loop variable for sink and source ports
sound/soc/codecs/rt711-sdw.c:382:4: style: Variable 'i' is reassigned
a value before the old one has been used. [redundantAssignment]
i = 0;
^
sound/soc/codecs/rt711-sdw.c:371:4: note: i is assigned
i++;
^
sound/soc/codecs/rt711-sdw.c:382:4: note: i is overwritten
i = 0;
^
Zhang Qilong [Wed, 11 Nov 2020 13:09:20 +0000 (21:09 +0800)]
ASoC: arizona: Fix a wrong free in wm8997_probe
In the normal path, we should not free the arizona,
we should return immediately. It will be free when
call remove operation.
Fixes: 87ca3a948a4eb ("ASoC: arizona: Move request of speaker IRQs into bus probe") Reported-by: Richard Fitzgerald <rf@opensource.cirrus.com> Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com> Acked-by: Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://lore.kernel.org/r/20201111130923.220186-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Wed, 11 Nov 2020 15:44:29 +0000 (15:44 +0000)]
Merge series "Fix PM disable depth imbalance on error" from Zhang Qilong <zhangqilong3@huawei.com>:
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context. This series of patches
fixed it.
Zhang Qilong (3):
ASoC: wm8994: Fix PM disable depth imbalance on error
ASoC: wm8997: Fix PM disable depth imbalance on error
ASoC: wm8998: Fix PM disable depth imbalance on error
Jiaxin Yu [Wed, 11 Nov 2020 02:45:22 +0000 (10:45 +0800)]
ASoC: mediatek: mt8192: skip first time data at the beginning of DMIC recording
We can choose to drop away any length of data from the beginning according
to project needs. Some projects don't want to throw away any data, because
they want to use recorded data to do echo cancellation, so they have to
make sure that they are aligned with the reference data as much as
possible. Or there are other algorithms in the upper layer to eliminate
this noise. Or some projects want to eliminate this noise form the kernel
layer. However, the minimum recommended value is 50ms to skip pop noise.
Jon Hunter [Wed, 11 Nov 2020 10:32:45 +0000 (10:32 +0000)]
ASoC: tegra: Don't warn on probe deferral
Deferred probe is an expected return value for snd_soc_register_card().
Given that the driver deals with it properly, there's no need to output
a warning that may potentially confuse users.
Zhang Qilong [Wed, 11 Nov 2020 04:13:26 +0000 (12:13 +0800)]
ASoC: wm8998: Fix PM disable depth imbalance on error
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context.
Fixes: 87ca3a948a4eb ("ASoC: arizona: Move request of speaker IRQs into bus probe") Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com> Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://lore.kernel.org/r/20201111041326.1257558-4-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Zhang Qilong [Wed, 11 Nov 2020 04:13:24 +0000 (12:13 +0800)]
ASoC: wm8994: Fix PM disable depth imbalance on error
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context.
Fixes: 4e9644271c341 ("ASoC: wm8994: Move runtime PM init to platform device init") Signed-off-by: Zhang Qilong <zhangqilong3@huawei.com> Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://lore.kernel.org/r/20201111041326.1257558-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Tue, 10 Nov 2020 21:36:13 +0000 (21:36 +0000)]
Merge series "Audio Graph Updates" from Sameer Pujar <spujar@nvidia.com>:
This series is a prepraration for using generic graph driver for Tegra210
audio. Tegra audio graph series will be sent in a separate series because
it has some dependency over other series for documentation work. This
series can focus on the generic ASoC driver updates. Below are the summary
of changes done.
* Support multiple instances of a component. For example there can be
multiple I2S devices which can use the same component driver.
* Support open platforms with empty Codec endpoint. Customers can plug
their own HW and can populate codec endpoint.
* In a component model there can be many components which can be
connected together. In such cases Identify no-pcm DPCM DAI links which
can be used in BE<->BE connections.
* Expose structures or helpers to be re-used by other similar graph
drivers.
The series is based on following references where DPCM usgae for Tegra
Audio and simple-card driver proposal were discussed.
* https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra)
* https://lkml.org/lkml/2020/6/27/4 (simple-card driver)
Changelog
=========
v4 -> v5
--------
* No changes in the core/audio-graph driver patches which are
part of the current series.
* Dropped Tegra audio patches and doc patches. This will be
sent separately once the doc depdendencies are resolved.
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 (7):
ASoC: soc-core: Fix component name_prefix parsing
ASoC: soc-pcm: Get all BEs along DAPM path
ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names
ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM
ASoC: audio-graph: Support empty Codec endpoint
ASoC: audio-graph: Expose new members for asoc_simple_priv
ASoC: audio-graph: Expose helpers from audio graph
Dan Carpenter [Thu, 5 Nov 2020 12:51:54 +0000 (15:51 +0300)]
ASoC: qcom: common: Fix refcounting in qcom_snd_parse_of()
There are two issues in this function.
1) We can't drop the refrences on "cpu", "codec" and "platform" before
we take the reference. This doesn't cause a problem on the first
iteration because those pointers start as NULL so the of_node_put()
is a no-op. But on the subsequent iterations, it will lead to a use
after free.
2) If the devm_kzalloc() allocation failed then the code returned
directly instead of cleaning up.
Fixes: 80d0c04e02ea ("ASoC: qcom: common: Fix refcount imbalance on error") Fixes: e16fba78b33a ("ASoC: qcom: common: use modern dai_link style") Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Link: https://lore.kernel.org/r/20201105125154.GA176426@mwanda Signed-off-by: Mark Brown <broonie@kernel.org>
Sameer Pujar [Mon, 2 Nov 2020 15:10:14 +0000 (20:40 +0530)]
ASoC: audio-graph: Expose helpers from audio graph
This commit exposes following functions which can be used by a sound
card driver based on generic audio graph driver. Idea is vendors can
have a thin driver and re-use common stuff from audio graph driver.
- graph_card_probe()
- graph_parse_of()
In doing so a new header file is added for above. The graph_probe()
function is simplified by moving more common stuff to graph_parse_of().
Sameer Pujar [Mon, 2 Nov 2020 15:10:13 +0000 (20:40 +0530)]
ASoC: audio-graph: Expose new members for asoc_simple_priv
Add new members in struct 'asoc_simple_priv'. Idea is to leverage
simple or graph card driver as much as possible and vendor can
maintain a thin driver to control the behavior by populating these
newly exposed members.
Following are the members added in 'asoc_simple_priv':
- 'ops' struct: In some cases SoC vendor drivers may want to
implement 'snd_soc_ops' callbacks differently. In such cases
custom callbacks would be used.
- 'force_dpcm' flag: Right now simple or graph card drivers
detect DAI links as DPCM links if:
* The dpcm_selectable is set AND
* Codec is connected to multiple CPU endpoints or aconvert
property is used for rate/channels.
So there is no way to directly specify usage of DPCM alone. So a
flag is exposed to mark all links as DPCM. Vendor driver can
set this if required.
- 'dpcm_selectable': Currently simple or audio graph drivers
provide a way to enable this for specific compatibles. However
vendor driver may want to define some additional info. Thus
expose this variable where vendor drivers can set this if
required.
Audio graph driver is updated to consider above flags or callbacks.
Subsequent patches in the series illustrate usage for above.
Sameer Pujar [Mon, 2 Nov 2020 15:10:12 +0000 (20:40 +0530)]
ASoC: audio-graph: Support empty Codec endpoint
For open platforms, which can support pluggable audio cards, Codec
endpoint is not fixed always. It actually depends on the compatible
HW module that is going to be connected. From SoC side the given I/O
interface is always available. Hence such links have fixed CPU endpoint
but no Codec endpoint. This patch helps to support such links where
user can populate Codec endpoint only and its fields in Platform DT
depending on the plugged HW.
Sameer Pujar [Mon, 2 Nov 2020 15:10:11 +0000 (20:40 +0530)]
ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM
PCM devices are created for FE dai links with 'no-pcm' flag as '0'.
Such DAI links have CPU component which implement either pcm_construct()
or pcm_new() at component or dai level respectively. Based on this,
current patch exposes a helper function to identify such components
and populate 'no_pcm' flag for DPCM DAI link.
This helps to have BE<->BE component links where PCM devices need
not be created for CPU component involved in such links.
Sameer Pujar [Mon, 2 Nov 2020 15:10:10 +0000 (20:40 +0530)]
ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names
For multiple instances of components, using DAI name alone for DAI links
is causing conflicts. Components can define multiple DAIs and hence using
just a device name won't help either. Thus DT device node reference and
DAI names are used to uniquely represent DAI link names.
Sameer Pujar [Mon, 2 Nov 2020 15:10:09 +0000 (20:40 +0530)]
ASoC: soc-pcm: Get all BEs along DAPM path
dpcm_end_walk_at_be() stops the graph walk when first BE is found for
the given FE component. In a component model we may want to connect
multiple DAIs from different components. A new flag is introduced in
'snd_soc_card', which when set allows DAI/component chaining. Later
PCM operations can be called for all these listed components for a
valid DAPM path.
Sameer Pujar [Mon, 2 Nov 2020 15:10:08 +0000 (20:40 +0530)]
ASoC: soc-core: Fix component name_prefix parsing
The "prefix" can be defined in DAI link node or it can be specified as
part of the component node itself. Currently "sound-name-prefix" defined
in a component is not taking effect. Actually the property is not getting
parsed. It can be fixed by parsing "sound-name-prefix" property whenever
"prefix" is missing in DAI link Codec node.
Mark Brown [Tue, 10 Nov 2020 15:57:39 +0000 (15:57 +0000)]
Merge series "Modify documentation and machine driver for SC7180 sound card" from Ajye Huang <ajye.huang@gmail.com>:
Note:
- The patch is made by the collaboration of
Ajye Huang <ajye_huang@compal.corp-partner.google.com>
Cheng-Yi Chiang <cychiang@chromium.org>
v6:
- Documentation: Addressed suggestions from Rob Herring.
- Define "maxItems: 1" in dmic-gpios property.
- Only keep one example and add dmic-gpios property in.
v5:
- Machine driver:
- Fix a format string warning (Reported-by: kernel test robot <lkp@intel.com>).
detailed info at https://lore.kernel.org/patchwork/patch/1331087/
v4:
- Machine driver: Addressed suggestions from Tzung-Bi.
- Remove redundant judgments in dmic_set() and dmic_get().
- Remove 1 level indent of judgment of IS_ERR(data->dmic_sel).
v3:
- Machine driver: Addressed suggestions from Tzung-Bi.
- Move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data.
- Remove redundant judgments in dmic_set().
v2:
- Documentation: Modify the dimc-gpios property description and examples.
- Machine driver:
- Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible
- See gpio property and use anadditional control.
Thanks for the review!
Ajye Huang (2):
ASoC: google: dt-bindings: modify machine bindings for two MICs case
ASoC: qcom: sc7180: Modify machine driver for 2mic
Mark Brown [Tue, 10 Nov 2020 15:57:37 +0000 (15:57 +0000)]
Merge series "ASoC: soc-compress: tidyup STREAM vs COMPRESS" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Mark
These are v2 of tidyup patch-set for SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx.
soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx, but mixed use.
This is confusable, but no problem. Because these are defined as UAPI and
are using same value.
This patch-set make sure these are same value.
v1 -> v2
- checks COMPRESS vs PCM_STREAM by using BUILD_BUG_ON()
Essentially both COMPRESS and PCM_STREAM definitions are identical,
and can be never different because of ABI compatibility,
which means it's safe to mix both variants in the code.
This patch checks it by BUILD_BUG_ON(), and merge to use.
Shane.Chien [Tue, 10 Nov 2020 02:31:32 +0000 (10:31 +0800)]
dt-bindings: mediatek: mt6359: Add new property for mt6359
This patch add "LDO_VAUD18-supply" property to
control vaud18 regulator. It is labeled as required
due to mt6359 audio path always need to enable vaud18.
Shane.Chien [Tue, 10 Nov 2020 02:31:31 +0000 (10:31 +0800)]
ASoC: Fix vaud18 power leakage of mt6359
vaud18 is power of mt6359 audio path. It
should only enable when audio is used,
instead of in boot up stage.
Once mt6359 audio path is enabled or disabled,
vaud18 is controlled by regulator supply widget
"LDO_VAUD18". Due to vaud18 is controlled by
regulator dapm macro instead of regmap, the macro
MT6359_LDO_VAUD18_CON0 and variable avdd_reg
is no used and removed from mt6359.h.
Mark Brown [Mon, 9 Nov 2020 19:43:31 +0000 (19:43 +0000)]
Merge series "ASoC: ti: davinci-mcasp: Handle incomplete DT node gracefully" from Peter Ujfalusi <peter.ujfalusi@ti.com>:
Hi,
The series is inspired by the effort to standardize TI's arm64 dtsi files to keep
all nodes in 'okay' state and let the board dts files disable not needed
peripherals (and not properly configured):
https://lore.kernel.org/lkml/20201104224356.18040-1-nm@ti.com/
In the unlikely (or likely?) event when the dts misses to disable the McASP node
which is not configured we currenly and luckily just manage to not crash as we
had fixup code in place for legacy pdata boots.
This however prints out a message which does not really help to identify the
issue.
This series will reduce some of the noise during boot (first patch) then
adds the needed changes to handle the variations of 'okay':
A - have all required DT properties for audio
B - gpiochip is enabled
A && !B -> everything is OK for audio, no gpiochip registered
A && B -> everything is OK for audio, gpiochip is registered
!A && B -> audio is not OK, gpiochip is registered. dev_dbg() for audio and do
not register SOC DAI and PCM
!A && !B -> audio is not OK, no gpiochip. dev_err() and fail the probe
Regards,
Peter
---
Peter Ujfalusi (4):
ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional
ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing
ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling
ASoC: ti: davinci-mcasp: Handle missing required DT properties
McASP needs three required properties to be usable for audio:
op-mode, tdm-slots and the serial-dir array.
Instead of probing the driver even without the needed information we should
make sure that all the parameters are provided for operation.
The fact that McASP can act as a GPIO controller for it's pins complicates
this a bit, but as a general rule we can:
- we fail the probe if McASP is not configured to be used as gpiochip
- we will not register the DAI (and PCM) if gpiochip is defined
Peter Ujfalusi [Fri, 6 Nov 2020 07:25:50 +0000 (09:25 +0200)]
ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling
Replace the davinci_mcasp_set_pdata_from_of() function which returned a
pdata pointer with davinci_mcasp_get_config() to return an actual error
code and handle all pdata validation and private mcasp struct setup in
there.
Drop the unused ram-size-playback and sram-size-capture query from DT at
the same time.
The legacy dma_request (which was holding the DMA request number) is no
longer in use for a long time.
All legacy platforms has been converted to dma_slave_map.
Remove it along with the DT parsing to get tx_dma_channel and
rx_dma_channel.
Mark Brown [Fri, 6 Nov 2020 11:21:24 +0000 (11:21 +0000)]
Merge series "ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Rob, Mark Cc: Sameer
These are v6 of switch to yaml base Documentation for
audio-graph-card and renesas,rsnd which uses audio-graph-card.
v5 -> v6
- add Acked-by/Reviewed-by on each patch
- remove "audio-graph-card,xxx" property
- remove un-needed if "ok"
v4 -> v5
- move "endpoint" under "port"
v3 -> v4
- fixuped yamllint warnings/errors
v2 -> v3
- fixuped Rob's pointed points
v1 -> v2
- upgrade dt-schema, and fixup errors for [2/2] patch
Kuninori Morimoto (3):
dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation
dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation
dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support
Shengjiu Wang [Mon, 2 Nov 2020 01:52:27 +0000 (09:52 +0800)]
ASoC: fsl_aud2htx: Add aud2htx module driver
The AUD2HTX is a digital module that provides a bridge between
the Audio Subsystem and the HDMI RTX Subsystem. This module
includes intermediate storage to queue SDMA transactions prior
to being synchronized and passed to the HDMI RTX Subsystem over
the Audio Link.
The AUD2HTX contains a DMA request routed to the SDMA module.
This DMA request is controlled based on the watermark level in
the 32-entry sample buffer.
Xu Wang [Thu, 29 Oct 2020 09:01:04 +0000 (09:01 +0000)]
ASoC: pxa: pxa-ssp: Remove redundant null check before clk_prepare_enable/clk_disable_unprepare
ecause clk_prepare_enable() and clk_disable_unprepare() already checked
NULL clock parameter, so the additional checks are unnecessary, just
remove them.
Cezary Rojewski [Mon, 26 Oct 2020 10:01:29 +0000 (11:01 +0100)]
ASoC: pcm: DRAIN support reactivation
soc-pcm's dpcm_fe_dai_do_trigger() supported DRAIN commnad up to kernel
v5.4 where explicit switch(cmd) has been introduced which takes into
account all SNDRV_PCM_TRIGGER_xxx but SNDRV_PCM_TRIGGER_DRAIN. Update
switch statement to reactive support for it.
As DRAIN is somewhat unique by lacking negative/stop counterpart, bring
behaviour of dpcm_fe_dai_do_trigger() for said command back to its
pre-v5.4 state by adding it to START/RESUME/PAUSE_RELEASE group.
Fixes: 24463e83c5e3 ("ASoC: pcm: update FE/BE trigger order based on the command") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20201026100129.8216-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
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