The only usage of these is to assign their address to the ops field in
the snd_soc_dai_driver struct, which is a pointer to const. Make them
const to allow the compiler to put them in read-only memory.
Stefan Binding [Mon, 27 Sep 2021 11:14:37 +0000 (12:14 +0100)]
ASoC: cs42l42: Use two thresholds and increased wait time for manual type detection
Some headsets require very different comparator thresholds for type detection,
as well as longer settling times. In order to detect a larger number of headsets,
use 2 thresholds to give maximum coverage (1.25V and 1.75V), as well as a longer
settling time of 100ms. This will not affect default audotodetect mode
and applies to manual mode type detection only.
In file included from include/linux/io.h:13,
from sound/soc/samsung/s3c-i2s-v2.c:16:
sound/soc/samsung/s3c-i2s-v2.c: In function 's3c2412_i2s_trigger':
arch/arm/include/asm/io.h:92:22: error: this statement may fall through [-Werror=implicit-fallthrough=]
#define __raw_writel __raw_writel
^
arch/arm/include/asm/io.h:299:29: note: in expansion of macro '__raw_writel'
#define writel_relaxed(v,c) __raw_writel((__force u32) cpu_to_le32(v),c)
^~~~~~~~~~~~
arch/arm/include/asm/io.h:307:36: note: in expansion of macro 'writel_relaxed'
#define writel(v,c) ({ __iowmb(); writel_relaxed(v,c); })
^~~~~~~~~~~~~~
sound/soc/samsung/s3c-i2s-v2.c:398:3: note: in expansion of macro 'writel'
writel(0x0, i2s->regs + S3C2412_IISFIC);
^~~~~~
sound/soc/samsung/s3c-i2s-v2.c:400:2: note: here
case SNDRV_PCM_TRIGGER_RESUME:
^~~~
From all I can tell, this was indeed meant to fall through, so
add "fallthrough;" statement to avoid the warning.
ASoC: wcd9335: Use correct version to initialize Class H
The versioning scheme was changed in an earlier patch, which caused the version
being used to initialize WCD9335 to be interpreted as if it was WCD937X, which
changed code paths causing broken headphones output. Pass WCD9335 instead of
WCD9335_VERSION_2_0 to wcd_clsh_ctrl_alloc to fix it.
Fixes: 19c5d1f6a0c3 ("ASoC: codecs: wcd-clsh: add new version support") Signed-off-by: Yassine Oudjana <y.oudjana@protonmail.com> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210925022339.786296-1-y.oudjana@protonmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Fix warning related to 'sound-name-prefix' binding
commit 82d3ec1d89fa ("ASoC: Use schema reference for sound-name-prefix")
added name-prefix.yaml schema and the same reference was used in couple
of other schemas. But this is causing following warning and the same is
fixed in current patch.
Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml:
audio-codec@34: 'sound-name-prefix' does not match any of the regexes:
'pinctrl-[0-9]+'
From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml
Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml:
audio-codec@36: 'sound-name-prefix' does not match any of the regexes:
'pinctrl-[0-9]+'
From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml
Fixes: 82d3ec1d89fa ("ASoC: Use schema reference for sound-name-prefix") Reported-by: Rob Herring <robh+dt@kernel.org> Suggested-by: Rob Herring <robh+dt@kernel.org> Signed-off-by: Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/1632238860-16947-1-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie@kernel.org>
These are only assigned to the ops field in the snd_soc_dai_link struct
which is a pointer to const struct snd_soc_ops. Make them const to allow
the compiler to put them in read-only memory.
Mark Brown [Mon, 20 Sep 2021 15:21:19 +0000 (16:21 +0100)]
ASoC: Drop mistakenly applied SPI patch
Revert 6e8cc4ddce828 ("spi: tegra20-slink: Declare runtime suspend and
resume functions conditionally") which was mistakenly applied to the
ASoC tree not the SPI tree (where it was also applied.
Mark Brown [Mon, 20 Sep 2021 14:46:54 +0000 (15:46 +0100)]
Merge series "ASoC: compress: Support module_get on stream open" from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
Hi,
SOF is marking all componet drivers with module_get_upon_open = 1 which works
fine with normal PCM streams, however on compressed side the module get upon
open is not supported. The module_get works when module_get_upon_open is not set
becasue the snd_soc_component_module_get_when_probe() will pass NULL for the
substream parameter of snd_soc_component_module_get().
In order to re-use the existing infrastructure for module_get, the proposal is
to convert the mark_module to void pointer (like the pm mark) and implement
matching code for the compressed open/free to pcm open/close.
Regards,
Peter
---
Peter Ujfalusi (2):
ASoC: soc-component: Convert the mark_module to void*
ASoC: compress/component: Use module_get_when_open/put_when_close for
cstream
Mark Brown [Mon, 20 Sep 2021 14:46:53 +0000 (15:46 +0100)]
Merge series "Extend AHUB audio support for Tegra210 and later" from Sameer Pujar <spujar@nvidia.com>:
Earlier as part of series [0], support for ADMAIF and I/O modules (such
as I2S, DMIC and DSPK) was added. This series aims at exposing some of
the AHUB internal modules (listed below), which can be used for audio
pre or post processing.
These modules can be plugged into audio paths and relevant processing
can be done. The MUX routes are extended to allow add or remove above
modules in the path via mixer controls. This is similar to how specific
ADMAIF channels are connected to relevant I/O module instances at the
moment.
Some of these modules can alter PCM parameters. Consider example of
resampler (44.1 -> 48 kHz) in the path.
The modules following SFC should be using converted sample rate and DAIs
need to be configured accordingly. The audio-graph driver provides a
mechanism to fixup the new parameters which can be specified in DT for a
given DAI. Then core uses these new values via fixup callback and then
pass it to respective DAIs hw_param() callback. The "convert-rate",
described in [1], property can be used when there is rate conversion in
the audio path. Similarly "convert-channels" can be used when there is
channel conversion in the path. There is no "convert-xxx" property for
sample size conversions. It can be added if necessary.
v1 -> v2
--------
* Put comments for soft reset application in the drivers.
* Split out mute/volume control logic in put() calls of MVC driver and
use separate callbacks for the respective kcontrols.
* Update kcontrol put() callback in MVC driver to return 1 whenever
there is change. Similar change is done in other drivers too.
* Use name-prefix.yaml reference for the driver documentation now.
* Add sound-name-prefix pattern for MIXER driver and use prefix
accordingly in DT.
Sameer Pujar (13):
ASoC: soc-pcm: Don't reconnect an already active BE
ASoC: simple-card-utils: Increase maximum DAI links limit to 512
ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link
ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules
ASoC: tegra: Add routes for few AHUB modules
ASoC: tegra: Add Tegra210 based MVC driver
ASoC: tegra: Add Tegra210 based SFC driver
ASoC: tegra: Add Tegra210 based AMX driver
ASoC: tegra: Add Tegra210 based ADX driver
ASoC: tegra: Add Tegra210 based Mixer driver
arm64: defconfig: Enable few Tegra210 based AHUB drivers
arm64: tegra: Add few AHUB devices for Tegra210 and later
arm64: tegra: Extend APE audio support on Jetson platforms
Krzysztof Kozlowski [Mon, 20 Sep 2021 11:21:06 +0000 (13:21 +0200)]
ASoC: dt-bindings: rt5682s: correct several errors
Correct several errors in rt5682s dtschema:
1. The examples should be under "examples":
'example' is not one of ['$id', '$schema', 'title', 'description', 'examples', ...
2. Missing type for vendor properties
3. clock-names should be an array:
properties:clock-names:items: {'const': 'mclk'} is not of type 'array'
4. Example DTS should include headers:
[scripts/Makefile.lib:386: Documentation/devicetree/bindings/sound/realtek,rt5682s.example.dt.yaml] Error 1
5. Node name in example DTS misses unit address and does not match DT
convention (generic name):
Warning (reg_format): /example-0/rt5682s:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1)
6. Node address should be in size-cells:0 block in example DTS:
Warning (reg_format): /example-0/codec@1a:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1)
The Mixer supports mixing of up to ten 7.1 audio input streams and
generate five outputs (each of which can be any combination of the
ten input streams)
This patch registers Mixer driver with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device.
The DAI driver exposes Mixer interfaces, which can be used to connect
different components in the ASoC layer. Makefile and Kconfig support
is added to allow build the driver. It can be enabled in the DT via
"nvidia,tegra210-amixer" compatible binding.
The Audio Demultiplexer (ADX) block takes an input stream with up to
16 channels and demultiplexes it into four output streams of up to 16
channels each. A byte RAM helps to form output frames by any combination
of bytes from the input frame. Its design is identical to that of byte
RAM in the AMX except that the data flow direction is reversed.
This patch registers ADX driver with ASoC framework. The component driver
exposes DAPM widgets, routes and kcontrols for the device. The DAI driver
exposes ADX interfaces, which can be used to connect different components
in the ASoC layer. Makefile and Kconfig support is added to allow build
the driver. It can be enabled in the DT via "nvidia,tegra210-adx"
compatible binding.
The Audio Multiplexer (AMX) block can multiplex up to four input streams
each of which can have maximum 16 channels and generate an output stream
with maximum 16 channels. A byte RAM helps to form an output frame by
any combination of bytes from the input frames.
This patch registers AMX driver with ASoC framework. The component driver
exposes DAPM widgets, routes and kcontrols for the device. The DAI driver
exposes AMX interfaces, which can be used to connect different components
in the ASoC layer. Makefile and Kconfig support is added to allow build
the driver. It can be enabled in the DT via "nvidia,tegra210-amx" for
Tegra210 and Tegra186. For Tegra194 and later, "nvidia,tegra194-amx" can
be used.
The Sampling Frequency Converter (SFC) converts the sampling frequency
of the input signal from one frequency to another. It supports sampling
frequency conversions of streams of up to two channels (stereo).
This patch registers SFC driver with ASoC framework. The component driver
exposes DAPM widgets, routes and kcontrols for the device. The DAI driver
exposes SFC interfaces, which can be used to connect different components
in the ASoC layer. Makefile and Kconfig support is added to allow build
the driver. It can be enabled in the DT via "nvidia,tegra210-sfc"
compatible binding.
The Master Volume Control (MVC) provides gain or attenuation to a digital
signal path. It can be used in input or output signal path for per-stream
volume control or it can be used as master volume control. The MVC block
has one input and one output. The input digital stream can be mono or
multi-channel (up to 7.1 channels) stream. An independent mute control is
also included in the MVC block.
This patch registers MVC driver with ASoC framework. The component driver
exposes DAPM widgets, routes and kcontrols for the device. The DAI driver
exposes MVC interfaces, which can be used to connect different components
in the ASoC layer. Makefile and Kconfig support is added to allow build
the driver. It can be enabled in the DT via "nvidia,tegra210-mvc"
compatible binding.
Add routing support for following modules of AHUB:
* SFC (Sampling Frequency Converter)
* MVC (Master Volume Control)
* AMX (Audio Multiplexer)
* ADX (Audio Demultiplexer)
* Mixer
These modules can be plugged into audio path as per the need using
routing controls similar to the already existing routes to I/O modules
such as I2S, DMIC and DSPK.
ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules
This patch adds YAML schema for DT bindings of few AHUB modules.
These devices will be registered as ASoC components and bindings
will be used on Tegra210 and later chips. The bindings for below
mentioned modules are added:
ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link
When multiple components are connected back to back in an audio path,
hw_param fixup may be required for CPU or Codec endpoint of BE<->BE
DAI links. Currently fixup support is available for Codec and this
commit adds similar feature for CPU endpoint of a BE<->BE link.
For example a resampler component can be plugged into an audio path.
[ FE -> BE1 -> ... -> resampler -> ... BEn ]
The resampler DAI links can be:
BEx (CPU) -> resampler input (Codec)
resampler output (CPU) -> BEy (Codec)
Thus input and output sample rate parameters for resampler can be
fixed up as per the resample requirement.
ASoC: simple-card-utils: Increase maximum DAI links limit to 512
The current limit of 128 is not sufficient when more components are
added to the audio map on Tegra210 and later platforms. Thus it is
resulting in probe failure.
The requirement is of nearly ~200 DAI links. To give sufficient room
for future additions the maximum limit is increased to 512 DAI links.
This is a preparatory patch to add more components like resampler,
mixer, multiplexers, demultiplexers and volume controllers to Tegra210
and later platforms.
ASoC: soc-pcm: Don't reconnect an already active BE
In some cases, multiple FE components have the same BE component in their
respective DPCM paths. One such example would be a mixer component, which
can receive two or more inputs and sends a mixed output. In such cases,
to avoid reconfiguration of already active DAI (mixer output DAI in this
case), check the BE stream state to filter out the redundancy.
In summary, allow connection of BE if the respective current stream state
is either NEW or CLOSED.
Peter Ujfalusi [Wed, 1 Sep 2021 09:52:55 +0000 (12:52 +0300)]
ASoC: compress/component: Use module_get_when_open/put_when_close for cstream
Currently the try_module_get() and module_put() is not possible for
compressed streams if the module_get_upon_open is set to 1 which means that\
the components are not protected in a same way as components when normal
audio is used.
SOF is setting module_get_upon_open to 1 for component drivers which works
correctly for audio stream but when compressed stream is used then the
module is not protected.
Convert the compress open and free operation to mimic the steps of it's
pcm counterpart to fix this issue.
Peter Ujfalusi [Wed, 1 Sep 2021 09:52:54 +0000 (12:52 +0300)]
ASoC: soc-component: Convert the mark_module to void*
The mark_module of the snd_soc_component is strict snd_pcm_substream type
which prevents it to be used by compressed streams.
Change the type to void* along with the snd_soc_component_module_get()
and snd_soc_component_module_put() to allow the same mark to be used by
compressed when it's module_get_upon_open is set to 1.
Mark Brown [Fri, 17 Sep 2021 13:56:43 +0000 (14:56 +0100)]
Merge series "ASoC: SOF: ipc: Small cleanups for message handler functions" from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
Hi,
Rename the parameter for ipc_trace_message() to match it's content and use
%#x" for hexadecimal prints in remaining places.
Regards,
Peter
---
Peter Ujfalusi (2):
ASoC: SOF: ipc: Clarify the parameter name for ipc_trace_message()
ASoC: SOF: ipc: Print 0x prefix for errors in
ipc_trace/stream_message()
Peter Ujfalusi [Thu, 16 Sep 2021 13:03:08 +0000 (16:03 +0300)]
ASoC: SOF: Rename sof_arch_ops to dsp_arch_ops
From the name sof_arch_ops one can not decipher that these ops are DSP
architecture ops.
Rename it to dsp_arch_ops and change also the macro to retrieve the DSP
architecture specific ops as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210916130308.7969-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Fri, 17 Sep 2021 08:58:22 +0000 (11:58 +0300)]
ASoC: SOF: ipc: Clarify the parameter name for ipc_trace_message()
ipc_trace_message() receives the type not the ID.
Use the same naming as the ipc_stream_message() function: msg_type to
help the reader to follow the code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Bard Liao <bard.liao@intel.com> Link: https://lore.kernel.org/r/20210917085823.27222-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Thu, 16 Sep 2021 15:06:47 +0000 (16:06 +0100)]
Merge series "ASoC: cs42l42: Implement Manual Type detection as fallback" from Vitaly Rodionov <vitalyr@opensource.cirrus.com>:
For some headsets CS42L42 autodetect mode is not working correctly.
They will be detected as unknown types or as headphones. According
to the CS42L42 datasheet, if the headset autodetect failed,
then the driver should switch to manual mode and perform a manual steps sequence.
These steps were missing in the current driver code. This patch will add manual
mode fallback steps in case autodetect failed. The default behavior is not affected,
manual mode runs only when autodetect failed.
Tested for regression with autodetect with all known headsets - no regression.
Tested with all headsets customers reported as false detected:
Gumdrop DropTech B1 - detected as headset OK
HUAWEI AM115 - detected as headset OK
UGREEN EP103 - detected as headset OK
HONOR AM116 - detected as headset OK
Stefan Binding (1):
ASoC: cs42l42: Implement Manual Type detection as fallback
Mark Brown [Thu, 16 Sep 2021 15:06:45 +0000 (16:06 +0100)]
Merge series "ASoC: SOF: Clean up the probe support" from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
Hi,
The probe debug feature of SOF can be used to extract streams of data from a
given point of a pipeline for analysis.
The support is implemented by using the ALSA/ASoC compress support for the
capture stream, but the code can not be used by/for a normal compressed data
stream. It is a debug feature.
Merge the probe implementation in the core (compress.c/h and probe.c/h) into
one file: sof-probes.c/h
Rename the Intel HDA specific probe implementation from hda-compressc.c to
hda-probes.c
We also need to add IPC logging support for the probes messages and drop the
unused references to SOF compress to have reasonably clean code.
Regards,
Peter
---
Peter Ujfalusi (5):
ASoC: SOF: ipc: Add probe message logging to ipc_log_header()
ASoC: SOF: pcm: Remove non existent CONFIG_SND_SOC_SOF_COMPRESS
reference
ASoC: SOF: probe: Merge and clean up the probe and compress files
ASoC: SOF: Intel: Rename hda-compress.c to hda-probes.c
ASoC: SOF: sof-probes: Correct the function names used for
snd_soc_cdai_ops
Ranjani Sridharan (1):
ASoC: SOF: compress: move and export sof_probe_compr_ops
Mark Brown [Wed, 15 Sep 2021 17:30:23 +0000 (18:30 +0100)]
ASoC: atmel: Convert to new style DAI format definitions
Convert the Atmel drivers to use the new style defines for clocking in DAI
formats.
Signed-off-by: Mark Brown <broonie@kernel.org> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Acked-by: Peter Rosin <peda@axentia.se>
David Rhodes [Wed, 15 Sep 2021 19:14:22 +0000 (14:14 -0500)]
ASoC: cs35l41: Binding fixes
Fix warnings and errors in DT bindings
Add newline at end of file
Replace 'unevaluatedProperties' with 'additionalProperties'
Add spi context to DT example
Add #sound-dai-cells to DT example
Rename to 'cirrus,cs35l41.yaml'
Stefan Binding [Thu, 16 Sep 2021 10:27:50 +0000 (11:27 +0100)]
ASoC: cs42l42: Implement Manual Type detection as fallback
Some headsets are not detected correctly by Automatic Type Detection
on cs42l42. Instead, Manual Type Detection can be used to give a
more accurate value.
Peter Ujfalusi [Thu, 16 Sep 2021 10:32:09 +0000 (13:32 +0300)]
ASoC: SOF: probe: Merge and clean up the probe and compress files
The probe debug functionality is implemented via compress support and it
was spread across two set of files:
probe.c/h
compress.c/h
Merge the two files into sof-probes.s/h and clean them up by removing
unused struct definitions, functions. We can also move most of the
functions static as they are only used internally.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.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/20210916103211.1573-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: SOF: compress: move and export sof_probe_compr_ops
sof_probe_compr_ops are not platform-specific. So move
it to common compress code and export the symbol. The
compilation of the common compress code is already dependent
on the selection of CONFIG_SND_SOC_SOF_DEBUG_PROBES, so no
need to check the Kconfig section for defining sof_probe_compr_ops
again.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://lore.kernel.org/r/20210916103211.1573-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Wed, 15 Sep 2021 12:21:12 +0000 (15:21 +0300)]
ASoC: SOF: debug: Add generic API and ops for DSP regions
Add new debugfs_add_region_item along with a generic wrapper
snd_sof_debugfs_add_region_item() to abstract away the DSP regions related
debugfs support.
At the same commit add iomem based generic implementation for the new ops
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210915122116.18317-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Wed, 15 Sep 2021 12:21:11 +0000 (15:21 +0300)]
ASoC: SOF: core: Do not use 'bar' as parameter for block_read/write
The use of bar in the core poses limits on the portability of the code
to other, non iomapped platforms.
To make the API more generic, remove the use of 'bar' as parameter
for the block copy API.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210915122116.18317-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Wed, 15 Sep 2021 12:21:05 +0000 (15:21 +0300)]
ASoC: SOF: Intel: bdw: Set the mailbox offset directly in bdw_probe
To align with other platforms, set only the sdev->dsp_box.offset in
bdw_probe().
The mailbox offset must be set in order to be able to receive the firmware
ready message.
The offsets and sizes will be re-configured after the FW ready message
based on the window information.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-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/20210915122116.18317-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Wed, 15 Sep 2021 15:12:29 +0000 (16:12 +0100)]
Merge series "ASoC: SOF: Remove unused members from struct sof_dev_desc" from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
Hi,
dma_engine, dma_size and resindex_dma_base is unused from sof_dev_desc, drop
them.
resindex_dma_base is initialized to -1 for Intel platforms, but it is not used.
Regards,
Peter
---
Peter Ujfalusi (2):
ASoC: SOF: intel: Do no initialize resindex_dma_base
ASoC: SOF: Drop resindex_dma_base, dma_engine, dma_size from
sof_dev_desc
projects / users / jedix / linux-maple.git / log