The HDAudio ASoC support relies on the set_tdm_slots() helper to store
the HDaudio stream tag in the tx_mask. This only works because of the
pre-existing order in soc-pcm.c, where the hw_params() is handled for
codec_dais *before* cpu_dais. When the order is reversed, the
stream_tag is used as a mask in the codec fixup functions:
/* fixup params based on TDM slot masks */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
codec_dai->tx_mask)
soc_pcm_codec_params_fixup(&codec_params,
codec_dai->tx_mask);
As a result of this confusion, the codec_params_fixup() ends-up
generating bad channel masks, depending on what stream_tag was
allocated.
We could add a flag to state that the tx_mask is really not a mask,
but it would be quite ugly to persist in overloading concepts.
Instead, this patch suggests a more generic get/set 'stream' API based
on the existing model for SoundWire. We can expand the concept to
store 'stream' opaque information that is specific to different DAI
types. In the case of HDAudio DAIs, we only need to store a stream tag
as an unsigned char pointer. The TDM rx_ and tx_masks should really
only be used to store masks.
Rename get_sdw_stream/set_sdw_stream callbacks and helpers as
get_stream/set_stream. No functionality change beyond the rename.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
This patch provides both a simplification of the suspend flows and a
better balanced operation during suspend/resume transition, as part of
the transition of Sound Open Firmware (SOF) to dynamic pipelines: the
DSP resources are only enabled when required instead of enabled on
startup.
The exiting code relies on a convoluted way of dealing with suspend
signals. Since there is no .suspend DAI callback, we used the
component .suspend and marked all the component DAI dmas as
'suspended'. The information was used in the .prepare stage to
differentiate resume operations from xrun handling, and only
reinitialize SHIM registers and DMA in the former case.
While this solution has been working reliably for about 2 years, there
is a much better solution consisting in trapping the TRIGGER_SUSPEND
in the .trigger DAI ops. The DMA is still marked in the same way for
the .prepare op to run, but in addition the callbacks sent to DSP
firmware are now balanced.
Normal operation:
hw_params -> intel_params_stream
hw_free -> intel_free_stream
This balanced operation was not required with existing SOF firmware
relying on static pipelines instantiated at every boot. With the
on-going transition to dynamic pipelines, it's however a requirement
to keep the use count for the DAI widget balanced across all
transitions.
The component suspend is not removed but instead modified to deal with
a corner case: when a substream is PAUSED, the ALSA core does not
throw the TRIGGER_SUSPEND. This is problematic since the refcount for
all pipelines and widgets is not balanced, leading to issues on
resume. The trigger callback keeps track of the 'paused' state with a
new flag, which is tested during the component suspend called later to
release the remaining DSP resources. These resources will be
re-enabled in the .prepare step.
The IPC used in the TRIGGER_SUSPEND to release DSP resources is not a
problem since the BE dailink is already marked as non-atomic.
ASoC/soundwire: intel: simplify callbacks for params/hw_free
We don't really need to pass a substream to the callback, we only need
the direction. No functionality change, only simplification to enable
improve suspend with paused streams.
Mark Brown [Thu, 23 Dec 2021 19:38:13 +0000 (19:38 +0000)]
ASoC: qcom: Parse "pin-switches" and "widgets" from DT
Merge series from Stephan Gerhold <stephan@gerhold.net>:
Some sound card setups might require extra pin switches to allow
turning off certain audio components. simple-card supports this
already using the "pin-switches" and "widgets" device tree property.
This series makes it possible to use the same properties for the Qcom
sound cards.
To implement that, the function that parses the "pin-switches" property
in simple-card-utils.c is first moved into the ASoC core. Then two
simple function calls are added to the common Qcom sound card DT parser.
Finally there is a small patch for the msm8916-wcd-analog codec to make
it possible to model sound card setups used in some MSM8916 smartphones.
(See PATCH 2/4 for an explanation of some real example use cases.)
Using pin switches rather than patching codec drivers with switches was
originally suggested by Mark Brown on a patch for the tfa989x codec:
https://lore.kernel.org/alsa-devel/YXaMVHo9drCIuD3u@sirena.org.uk/
Stephan Gerhold [Tue, 14 Dec 2021 14:20:49 +0000 (15:20 +0100)]
ASoC: msm8916-wcd-analog: Use separate outputs for HPH_L/HPH_R
The analog codec has separate output paths for the left headphone channel
(HPH_L) and the right headphone channel (HPH_R). While they are usually
used together for actual headphones output, some devices also have an
analog speaker amplifier connected to one of the headphone channels.
To allow modelling that properly (and to avoid powering on the unneeded
output path), HPH_L and HPH_R should be represented by separate outputs
rather than a shared HEADPHONE output that always activates both paths.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:48 +0000 (15:20 +0100)]
ASoC: qcom: common: Parse "pin-switches" and "widgets" from DT
Use the DT helpers in the ASoC core to parse the "pin-switches" and
"widgets" properties from the device tree. This allows adding extra
mixers to disable e.g. an extra speaker amplifier that would be
normally powered on automatically because it is connected to a shared
output pin.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:47 +0000 (15:20 +0100)]
ASoC: dt-bindings: qcom: sm8250: Document "pin-switches" and "widgets"
Some sound card setups might require extra pin switches to allow
turning off certain audio components. There are two real examples for
this in smartphones/tablets based on MSM8916:
1. Analog speaker amplifiers connected to headphone outputs.
The MSM8916 analog codec does not have a separate "Line Out" port
so some devices have an analog speaker amplifier connected to one
of the headphone outputs. A pin switch is necessary to allow
playback on headphones without also activating the speaker.
2. External speaker codec also used as earpiece.
Some smartphones have two front-facing (stereo) speakers that can
be also configured to act as an earpiece during voice calls. A pin
switch is needed to allow disabling the second speaker during
voice calls.
There are existing bindings that allow setting up such pin switches in
simple-card.yaml. Document the same for Qcom sound cards.
One variant of example 1 above is added to the examples in the DT
schema: There is an analog speaker amplifier connected to the HPH_R
(right headphone channel) output. Adding a "Speaker" pin switch and
widget allows turning off the speaker when audio should be only played
via the connected headphones.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:46 +0000 (15:20 +0100)]
ASoC: core: Add snd_soc_of_parse_pin_switches() from simple-card-utils
The ASoC core already has several helpers to parse card properties
from the device tree. Move the parsing code for "pin-switches" from
simple-card-utils to a shared snd_soc_of_parse_pin_switches() function
so other drivers can also use it to set up pin switches configured in
the device tree.
Mark Brown [Thu, 23 Dec 2021 17:15:58 +0000 (17:15 +0000)]
ASoC: SOF: Re-visit firmware state and panic tracking/handling
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
this series will improve how we are tracking the firmware's state to be able to
avoid communication with it when it is not going to answer due to a panic and
we will attempt to force power cycle the DSP to recover at the next runtime
suspend time.
The state handling brings in other improvements on the way the kernel reports
errors and DSP panics to reduce the printed lines for normal users, but at the
same time allowing developers (or for bug reports) to have more precise
information available to track down the issue.
We can now place messages easily in the correct debug level and not bound to the
static ERROR for some of the print chains, causing excess amount or partial
information to be printed, confusing users and machines (CI).
I would have prefered to split this series up, but it was developed together to
achieve a single goal to reduce the noise, but also provide the details we need
to be able to rootcause issues.
This is used in meson-gx. Add the property to the binding.
This fixes the dtschema warning:
audio-controller@5400: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
This is used in meson-axg, meson-g12 and meson-gx. Add the property to
the binding.
This fixes the dtschema warning:
audio-codec-0: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:28 +0000 (13:36 +0200)]
ASoC: SOF: Intel: hda: Use DEBUG log level for optional prints
If the user requested to see all dumps (even the optional ones) then use
KERN_DEBUG level for the optional dumps as they are only for debugging
purposes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Chao Song <chao.song@intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-21-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:27 +0000 (13:36 +0200)]
ASoC: SOF: debug: Use DEBUG log level for optional prints
If the user requested to see all dumps (even the optional ones) then use
KERN_DEBUG level for the optional dumps as they are only for debugging
purposes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Chao Song <chao.song@intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-20-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:26 +0000 (13:36 +0200)]
ASoC: SOF: Add clarifying comments for sof_core_debug and DSP dump flags
Update the comment for the global SOF level debug flags and add one for
the flags used to control the DSP dump functionality.
Document the expected behavior when the SOF_DBG_DUMP_OPTIONAL is passed
for the DSP dump:
Only print the dump if SOF_DBG_PRINT_ALL_DUMPS is set
Print must use KERN_DEBUG log level
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Chao Song <chao.song@intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-19-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:25 +0000 (13:36 +0200)]
ASoC: SOF: Rename snd_sof_get_status() and add kernel log level parameter
The snd_sof_get_status() is not the best name for a function which in fact
is tasked to print out DSP oops and stack. Rename it to
sof_print_oops_and_stack().
At the same time add a new parameter to specify the desired kernel log
level to be used for the prints.
When updating the users of the function, pass KERN_ERR for now to make sure
that there is no functional change happens.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Chao Song <chao.song@intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-18-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:23 +0000 (13:36 +0200)]
ASoC: SOF: ops: Always print DSP Panic message but use different message
Never suppress the DSP panic dump as it is always originates from an
assert() or panic() call within the firmware.
Use different message for DSP panics when there will be recovery attempt
going to be done compared to a definitive DSP panic.
Suggested-by: Chao Song <chao.song@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Chao Song <chao.song@intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:21 +0000 (13:36 +0200)]
ASoC: SOF: pm: Force DSP off on suspend in BOOT_FAILED state also
Try to force the DSP to be turned off next time if the fw_state is either
CRASHED or BOOT_FAILED when a suspend happens in order to attempt a clean
boot to recover.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:19 +0000 (13:36 +0200)]
ASoC: SOF: ipc: Only allow sending of an IPC in SOF_FW_BOOT_COMPLETE state
If the state of the firmware is not BOOT_COMPLETE, it means that the
firmware is not functioning, thus it is not capable of handling IPC
messages.
Do not try to send IPC if the state is not BOOT_COMPLETE
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20211223113628.18582-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:13 +0000 (13:36 +0200)]
ASoC: SOF: Add 'non_recoverable' parameter to snd_sof_dsp_panic()
Some platforms use retries during firmware boot to overcome DSP startup
issues.
In these cases we might receive a DSP panic message which should not be
treated as fatal if it happens during boot.
Pass this information to snd_sof_dsp_panic() and omit the panic print if
it is not fatal or the user does not want to see all dumps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-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/20211223113628.18582-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 23 Dec 2021 11:36:09 +0000 (13:36 +0200)]
ASoC: SOF: ops: Use dev_warn() if the panic offsets differ
Catch the cases when the stored sdev->dsp_oops_offset and the offset
received via the panic message differs and print a warning, but keep using
the dsp_oops_offset for the oops query.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-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/20211223113628.18582-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
This is used in meson-sm1 and meson-g12 .dtsi. Add the property to
the binding.
This fixes the dtschema warning:
audio-controller@740: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
This is used in meson-gxl and meson-g12-common .dtsi. Add the property to
the binding.
This fixes the dtschema warning:
audio-controller@32000: 'sound-name-prefix' does not match any of the
regexes: 'pinctrl-[0-9]+'
Vincent Knecht [Mon, 20 Dec 2021 19:37:25 +0000 (20:37 +0100)]
ASoC: Add AK4375 support
AK4375 is a 32-bit stereo DAC with headphones amplifier.
There's no documentation for it on akm.com, and only a brief
datasheet can be found floating on the internets [1].
Thanks to Oriane BAYERD <obayerd@eurocomposant.fr>
for finally answering my inquiries through akm.com, if only to tell
me that this chip is EOL following AKM factory burning in october 2020
and thus no detailed documentation is available anymore...
AK4331 is advertised [2] as pin and register compatible with AK4375
so some scraps of its datasheet were used and this driver might be
used as a base for it, but this is totally untested.
So this driver is mainly based on downstream code [3] and [4]
by Hu Jin from AKM (no known email).
Tested on msm8916-alcatel-idol347 and msm8939-alcatel-idol3,
which both use PLL driven clock with bypass of SRC (sample rate
converter), so only this setup is supported for now.
Lad Prabhakar [Tue, 21 Dec 2021 17:01:00 +0000 (17:01 +0000)]
ASoC: bcm: Use platform_get_irq() to get the interrupt
platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static
allocation of IRQ resources in DT core code, this causes an issue
when using hierarchical interrupt domains using "interrupts" property
in the node as this bypasses the hierarchical setup and messes up the
irq chaining.
In preparation for removal of static setup of IRQ resource from DT core
code use platform_get_irq().
While at it also drop "r_irq" member from struct bcm_i2s_priv as there
are no users of it.
Lad Prabhakar [Tue, 21 Dec 2021 17:00:59 +0000 (17:00 +0000)]
ASoC: xlnx: Use platform_get_irq() to get the interrupt
platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static
allocation of IRQ resources in DT core code, this causes an issue
when using hierarchical interrupt domains using "interrupts" property
in the node as this bypasses the hierarchical setup and messes up the
irq chaining.
In preparation for removal of static setup of IRQ resource from DT core
code use platform_get_irq().
Dan Carpenter [Fri, 17 Dec 2021 15:00:07 +0000 (18:00 +0300)]
ASoC: qdsp6: fix a use after free bug in open()
This code frees "graph" and then dereferences to save the error code.
Save the error code first and then use gotos to unwind the allocation.
Fixes: 4d78df72c8c1 ("ASoC: qdsp6: Fix an IS_ERR() vs NULL bug") Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Link: https://lore.kernel.org/r/20211217150007.GB16611@kili Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/sof/amd/acp.c:222:9: warning: Identical condition and return
expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/sof/amd/acp.c:213:6: note: If condition 'ret' is true, the
function will return/exit
if (ret)
^
sound/soc/sof/amd/acp.c:222:9: note: Returning identical expression 'ret'
return ret;
^
Dmitry Osipenko [Thu, 16 Dec 2021 16:02:29 +0000 (19:02 +0300)]
ASoC: tegra-audio-rt5677: Correct example
Remove non-existent properties from the example of the binding. These
properties were borrower from the old txt binding, but they were never
used in practice and aren't documented in the new binding. They aren't
reported by the binding checker because dtschema needs extra patch that
hasn't been upstreamed yet to make unevaluatedProperties work properly.
ASoC: amd: acp-config: Update sof_tplg_filename for SOF machines
SOF machines support different codec end points and hence required
different topologies configuration. Update tplg filename in machine
struct to load different topology files for SOF machines.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com> Reviewed-by: Curtis Malainey <curtis@malainey.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211221161814.236318-3-AjitKumar.Pandey@amd.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: amd: acp-config: Enable SOF audio for Google chrome boards.
We need to support sof audio on different variants of Google boards.
Add new entry in dmi table to enable SOF flag on Google chrome boards.
Also add newer machines to sof_machines list with codecs and amps acpi
id check to register sof sound cards on different variants.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@intel.com> Link: https://lore.kernel.org/r/20211221161814.236318-2-AjitKumar.Pandey@amd.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Fri, 17 Dec 2021 17:32:48 +0000 (17:32 +0000)]
ASoC: Intel: catpt: Dma-transfer fix and couple
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Set is made of one fix for dma-transfer so that result of
dmaengine_submit() is tested before moving on, and few cleanups:
- two non-impactful, where catpt_component_open() layout gets improved
slightly as well as relocation of couple of locals found in
PCM-functions so that they look more cohesive
- no need to expose catpt-driver board-matching information globally.
Most fields are not by it and it's the sole user of haswell_machines
table. By having them locally it is clear what is actually being used
Cezary Rojewski (5):
ASoC: Intel: catpt: Test dmaengine_submit() result before moving on
ASoC: Intel: catpt: Reduce size of catpt_component_open()
ASoC: Intel: catpt: Streamline locals declaration for PCM-functions
ASoC: Intel: catpt: Drop SND_SOC_ACPI_INTEL_MATCH dependency
ASoC: Intel: Drop legacy HSW/BDW board-match information
Mark Brown [Fri, 17 Dec 2021 17:32:46 +0000 (17:32 +0000)]
Support HDMI audio on NVIDIA Tegra20
Merge series from Dmitry Osipenko <digetx@gmail.com>:
This series revives Tegra20 S/PDIF driver which was upstreamed long time
ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI
audio CODEC provider. Finally, HDMI audio is enabled in device-trees.
For now the audio is enable only for Acer A500 tablet and Toshiba AC100
netbook because they're already supported by upstream, later on ASUS TF101
tablet will join them.
I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1]
that removes obsolete slave_id. This eases merging of the patches by
removing the merge conflict. This is a note for Mark Brown.
I also based this series on top of power management series [2]. I.e. [2]
should be applied first, otherwise "Add S/PDIF node to Tegra20 device-tree"
patch should have merge conflict. This is a note for Thierry.
v4: - Added patches that update multi_v7_defconfig with the enabled S/PDIF
and APB DMA drivers.
v3: - Renamed S/PDIF device-tree clocks as was suggested by Rob Herring.
- Added r-bs and acks that were given by Rob Herring to v2.
v2: - Corrected I2S yaml problem that was reported by the DT bot for v1
by removing the non-existent required clock-names property.
- Removed assigned-clocks property from S/PDIF yaml since this property
is now inherited from the clocks property.
- Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making
it the first sound/soc patch in the series, like it was suggested by
Mark Brown in the comment to v1. Also reworded commit message of this
patch to *not* make it looks like it should be backported to stable
kernels.
Mark Brown [Fri, 17 Dec 2021 17:32:45 +0000 (17:32 +0000)]
ASoC: SOF: couple of cleanups
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Guennadi spotted inconsistencies with our 'const' handling, Ajit Kumar
flagged a missing check for a null pointer and we missed the
definition of debug zones.
Heiner Kallweit [Fri, 17 Dec 2021 09:22:31 +0000 (10:22 +0100)]
ASoC: sh: rz-ssi: Check return value of pm_runtime_resume_and_get()
The return value of pm_runtime_resume_and_get() needs to be checked to
avoid a usage count imbalance in the error case. This fix is basically
the same as 6305ea2f20a8 ("reset: renesas: Fix Runtime PM usage"),
and the last step before pm_runtime_resume_and_get() can be annotated
as __must_check.
Cezary Rojewski [Thu, 16 Dec 2021 11:57:39 +0000 (12:57 +0100)]
ASoC: Intel: catpt: Test dmaengine_submit() result before moving on
After calling dmaengine_submit(), the submitted transfer descriptor
belongs to the DMA engine. Pointer to that descriptor may no longer be
valid after the call and should be tested before awaiting transfer
completion.
Reported-by: Kevin Tian <kevin.tian@intel.com> Suggested-by: Dave Jiang <dave.jiang@intel.com> Fixes: 6cb91f87afb5 ("ASoC: Intel: Add catpt base members") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20211216115743.2130622-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Dmitry Osipenko [Sat, 4 Dec 2021 14:37:16 +0000 (17:37 +0300)]
ASoC: tegra20: i2s: Filter out unsupported rates
Support new nvidia,fixed-parent-rate device-tree property which instructs
I2S that board wants parent clock rate to stay at a fixed rate. This allows
to play audio over S/PDIF and I2S simultaneously. The root of the problem
is that audio components on Tegra share the same audio PLL, and thus, only
a subset of rates can be supported if we want to play audio simultaneously.
Filter out audio rates that don't match parent clock rate if device-tree
has the nvidia,fixed-parent-rate property.
Dmitry Osipenko [Sat, 4 Dec 2021 14:37:15 +0000 (17:37 +0300)]
ASoC: tegra20: spdif: Filter out unsupported rates
SPDIF and other SoC components share audio PLL on Tegra, thus only one
component may set the desired base clock rate. This creates problem for
HDMI audio because it uses SPDIF and audio may not work if SPDIF's clock
doesn't exactly match standard audio rate since some receivers may reject
audio in that case. Filter out audio rates which SPDIF output can't
support, assuming that other components won't change rate at runtime.
Dmitry Osipenko [Sat, 4 Dec 2021 14:37:14 +0000 (17:37 +0300)]
ASoC: tegra20: spdif: Support system suspend
Support system suspend by enforcing runtime PM suspend/resume.
Now there is no doubt that h/w is indeed stopped during suspend
and that h/w state will be properly restored after resume.
Dmitry Osipenko [Sat, 4 Dec 2021 14:37:08 +0000 (17:37 +0300)]
ASoC: tegra20: spdif: Set FIFO trigger level
FIFO trigger level must be not less than the size of DMA burst, otherwise
audio will be played x4 faster that it should be because part of the DMA
data will be dropped on FIFO input buffer overflow.
Dmitry Osipenko [Sat, 4 Dec 2021 14:37:06 +0000 (17:37 +0300)]
ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property
Document new nvidia,fixed-parent-rate property which instructs that this
board wants parent clock to stay at a fixed rate. It allows to prevent
conflicts between audio components that share same parent PLL. For
instance, this property allows to have HDMI audio, speaker and headphones
in the system playing audio simultaneously, which is a common pattern for
consumer devices.
Mark Brown [Fri, 17 Dec 2021 11:13:39 +0000 (11:13 +0000)]
Merge tag 'dmaengine_topic_slave_id_removal_5.17' of git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/dmaengine into v4_20211204_digetx_support_hdmi_audio_on_nvidia_tegra20
dmaengine_topic_slave_id_removal_5.17
Tag for dmaengine slave_id removal topic branch which should be merged
into v5.17
Peter Ujfalusi [Thu, 16 Dec 2021 23:03:50 +0000 (17:03 -0600)]
ASoC: SOF: Kconfig: Make the SOF_DEVELOPER_SUPPORT depend on SND_SOC_SOF
SND_SOC_SOF_DEVELOPER_SUPPORT contains options affecting how the built
SOF driver stack will behave, enables debug options and other features.
These options have no meaning if the SND_SOC_SOF is not even enabled.
If we have SOF client options under developer_support and debug they can
be selected to be built even without the core, but they do need symbols
from the core (the sof-client API) which can result build failure.
In Kconfig we can have
SND_SOC_SOF_TOPLEVEL=y
SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST=y
for example, which will make the flood client to be built, but the SOF core
is not as SND_SOC_SOF is not selected.
Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com> Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211216230350.343857-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master
snd_soc_daifmt_parse_clock_provider_raw() is handling both
bitclock/frame-master, and is supporting both flag/phandle.
Current DT is assuming it is flag style.
This patch allows both case.
ASoC: SOF: ipc: Add null pointer check for substream->runtime
When pcm stream is stopped "substream->runtime" pointer will be set
to NULL by ALSA core. In case host received an ipc msg from firmware
of type IPC_STREAM_POSITION after pcm stream is stopped, there will
be kernel NULL pointer exception in ipc_period_elapsed(). This patch
fixes it by adding NULL pointer check for "substream->runtime".
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