Derek Fang [Fri, 1 Oct 2021 07:41:13 +0000 (15:41 +0800)]
ASoC: rt5682s: Fix HP noise caused by SAR mode switch when the system resumes
When the system resumes from S3, if the system plays a beep,
there is continuous "Zizi.." noise from HP that could be heard.
It is caused by the SAR mode switch during the combo jack
re-detection which be executed parallelly in a workqueue
after the system resumes.
This patch changes the behavior of SAR mode switch to
avoid this issue.
Derek Fang [Fri, 1 Oct 2021 07:41:12 +0000 (15:41 +0800)]
ASoC: rt5682s: Enable ASRC auto-disable to fix pop during jack plug-in while playback
When codec's ASRC was enabled, the DA and AD filters use
the reference clk which tracks the I2S clks.
And if the I2S clks' timing of open and close are not expected,
this patch allows the filters to switch to use sysclk if
ASRC's ref clks disappeared and could fix the below possible issues:
1. Avoid DA filter to keep surplus samples.
2. Avoid that AD filter works failed during dapm's power on.
For example,
if I2S clks were closed before dacdat during playback off
due to jack unplug, it causes ref clks disappeared and
DA filter remained some samples which will produce pop noise
on the next HP playback.
ASRC auto-disable could clear the samples during the playback off.
Trevor Wu [Fri, 1 Oct 2021 03:16:01 +0000 (11:16 +0800)]
ASoC: mediatek: mt8195: move of_node_put to remove function
platforms->of_node and codes->of_node are assigned in probe function,
and of_node_put is called at the end of probe function, because of_node
seems to be not used after probe functon
In this patch, of_node_put is moved to platform remove function in case
of_node is used at any occasions after probe function in the future.
Malik_Hsu [Fri, 1 Oct 2021 15:03:16 +0000 (23:03 +0800)]
ASoC: Intel: sof_rt5682: Add support for max98360a speaker amp
Add a board config adl_mx98360a_rt5682 to support alc5682 headset
codec and max98360a speaker amplifier. Follow Intel BT offload design
by connecting alc5682 to SSP0 and max98360a to SSP1.
Signed-off-by: Malik_Hsu <malik_hsu@wistron.corp-partner.google.com> Signed-off-by: Brent Lu <brent.lu@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211001150316.414141-1-brent.lu@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
The struct tegra_machine_snd_ops is only assigned to the ops field in the
snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops.
Make it const to allow the compiler to put it in read-only memory.
Simon Trimmer [Thu, 30 Sep 2021 14:21:16 +0000 (15:21 +0100)]
ASoC: soc-component: Remove conditional definition of debugfs data members
This simplification allows the use of the standard kernel pattern of
static inline dummy functions for debugfs code. Most systems will only
have a small number of snd_soc_components so the memory impact is
minimal.
Jack Yu [Wed, 29 Sep 2021 05:43:44 +0000 (13:43 +0800)]
ASoC: rt5682: move clk related code to rt5682_i2c_probe
The DAI clock is only used in I2S mode, to make it clear
and to fix clock resource release issue, we move CCF clock
related code to rt5682_i2c_probe to fix clock
register/unregister issue.
Peter Ujfalusi [Thu, 30 Sep 2021 07:04:38 +0000 (10:04 +0300)]
ASoC: SOF: Change SND_SOC_SOF_TOPLEVEL from config to menuconfig
We have growing number of options under SND_SOC_SOF_TOPLEVEL as SOF
adaptation is growing (Intel, NXP, AMD and Mediatek) and new features are
added.
It will make the menuconfig user experience much cleaner if we move the
SOF options under a separate page.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Bard Liao <bard.liao@intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20210930070438.16846-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
The struct mop500_ab8500_ops is only assigned to the ops field in the
snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops.
Make it const to allow the compiler to put it in read-only memory.
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.
Brent Lu [Mon, 27 Sep 2021 14:32:49 +0000 (22:32 +0800)]
ASoC: intel: sof_rt5682: update platform device name for Maxim amplifier
To follow 20-character length limitation of platform device name, we
have only 7 character space for amplifier. Therefore, the last
character of mx98357a and mx98360a is removed to save space.
Signed-off-by: Brent Lu <brent.lu@intel.com> Fixes: e224ef76fa8a ('ASoC: intel: sof_rt5682: support jsl_rt5682s_mx98360a board') Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210927143249.439129-1-brent.lu@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Tue, 28 Sep 2021 07:36:15 +0000 (10:36 +0300)]
ASoC: SOF: ipc: Make the error prints consistent in tx_wait_done()
If we get an error on reply (msg->reply_error) then we should print the
error value out.
At the same time extend the print to include the message size as well and
do the same in case of a timeout.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210928073615.29574-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Marc Herbert [Tue, 28 Sep 2021 10:26:35 +0000 (13:26 +0300)]
ASoC: SOF: prefix some terse and cryptic dev_dbg() with __func__
These helped troubleshoot some DMA issue in SOF.
Signed-off-by: Marc Herbert <marc.herbert@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://lore.kernel.org/r/20210928102635.26227-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: SOF: add error handling to snd_sof_ipc_msg_data()
If an invalid stream is passed to snd_sof_ipc_msg_data() it won't
fill the provided object with data. The caller has to be able to
recognise such cases to avoid handling invalid data. Make the
function return an error when failing.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi@intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://lore.kernel.org/r/20210928103516.8066-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Sparse (make C=2) complains about undeclared variables. Fix by adding
a real prototype instead of 'extern' in sof-of-dev.c
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://lore.kernel.org/r/20210928072807.27838-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Tue, 28 Sep 2021 08:17:44 +0000 (11:17 +0300)]
ASoC: SOF: Intel: hda-stream: Print stream name on STREAM_SD_OFFSET timeout
In order to provide more information in case of timeout observed while
reading STREAM_SD_OFFSET, print out the stream name or in case there is
no audio stream associated (like dma-trace), print "--"
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@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/20210928081744.4785-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Mon, 27 Sep 2021 17:32:10 +0000 (18:32 +0100)]
Merge series "add driver to support firmware loading on Cirrus Logic DSPs" from Simon Trimmer <simont@opensource.cirrus.com>:
This series of patches gradually separates the ASoC specific wm_adsp
code from that required to manage firmware in Cirrus Logic DSPs.
The series starts with renaming, progresses to splitting the
functionality before finally moving the independent functionality into
drivers/firmware so that it can be used by both the existing ASoC
wm_adsp and new non-audio parts.
Charles Keepax (3):
ASoC: wm_adsp: Move check for control existence
ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed
buffers
ASoC: wm_adsp: Move sys_config_size to wm_adsp
Simon Trimmer (13):
ASoC: wm_adsp: Remove use of snd_ctl_elem_type_t
ASoC: wm_adsp: Cancel ongoing work when removing controls
ASoC: wm_adsp: Rename generic DSP support
ASoC: wm_adsp: Introduce cs_dsp logging macros
ASoC: wm_adsp: Separate some ASoC and generic functions
ASoC: wm_adsp: Split DSP power operations into helper functions
ASoC: wm_adsp: Separate generic cs_dsp_coeff_ctl handling
ASoC: wm_adsp: Move check of dsp->running to better place
ASoC: wm_adsp: Pass firmware names as parameters when starting DSP
core
ASoC: wm_adsp: move firmware loading to client
ASoC: wm_adsp: Split out struct cs_dsp from struct wm_adsp
ASoC: wm_adsp: Separate wm_adsp specifics in cs_dsp_client_ops
firmware: cs_dsp: add driver to support firmware loading on Cirrus
Logic DSPs
Currently, The fixed 512KB prealloc buffer size is too larger for
tiny memory kernel (such as 16MB memory). This patch adds the module
option "prealloc_buffer_size_kbytes" to specify prealloc buffer size.
It's suitable for cards which use the generic dmaengine pcm driver
with no config.
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>
Simon Trimmer [Mon, 13 Sep 2021 16:00:57 +0000 (17:00 +0100)]
firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs
wm_adsp originally provided firmware loading on some audio DSP and was
implemented as an ASoC codec driver. However, the firmware loading now
covers a wider range of DSP cores and peripherals containing them,
beyond just audio. So it needs to be available to non-audio drivers. All
the core firmware loading support has been moved into a new driver
cs_dsp, leaving only the ASoC-specific parts in wm_adsp.
Simon Trimmer [Mon, 13 Sep 2021 16:00:55 +0000 (17:00 +0100)]
ASoC: wm_adsp: Split out struct cs_dsp from struct wm_adsp
In preparation for moving the generic DSP support out of ASoC split
struct wm_adsp into two parts, one will form the structure for the new
generic DSP code and embed that one into wm_adsp.
Simon Trimmer [Mon, 13 Sep 2021 16:00:54 +0000 (17:00 +0100)]
ASoC: wm_adsp: move firmware loading to client
This is preparation for moving the generic DSP support out of ASoC.
Passing the firmware as parameters into the power_up functions
simplifies the generic code that will be moved out of wm_adsp.
Simon Trimmer [Mon, 13 Sep 2021 16:00:53 +0000 (17:00 +0100)]
ASoC: wm_adsp: Pass firmware names as parameters when starting DSP core
As preparation for moving the generic DSP support out of ASoC pass the
firmware names used when loading files as parameters as the generic code
can't refer directly to the array specific to wm_adsp. The code
remaining in wm_adsp.c doesn't need to change, it can continue to use
the string arrays directly.
Simon Trimmer [Mon, 13 Sep 2021 16:00:52 +0000 (17:00 +0100)]
ASoC: wm_adsp: Move check of dsp->running to better place
In preparation for moving the generic DSP support out of ASoC, move
the check of dsp->running to a more appropriate place that will move
to the generic code.
Simon Trimmer [Mon, 13 Sep 2021 16:00:51 +0000 (17:00 +0100)]
ASoC: wm_adsp: Separate generic cs_dsp_coeff_ctl handling
This is preparation for moving the generic DSP support out of ASoC. The
majority of the handling of firmware controls is generic and this change
separates the generic and ASoC specific details into separate structures
and functions and renames the generic code named wm_* to cs_*.
Simon Trimmer [Mon, 13 Sep 2021 16:00:49 +0000 (17:00 +0100)]
ASoC: wm_adsp: Split DSP power operations into helper functions
This is preparation for moving the generic DSP support out of
ASoC. This change separates the generic handling of power and state
transitions from the DAPM API wrapper.
Simon Trimmer [Mon, 13 Sep 2021 16:00:48 +0000 (17:00 +0100)]
ASoC: wm_adsp: Separate some ASoC and generic functions
Split some functions into ASoC and generic portions so that existing
interfaces can be retained whilst allowing the implementation to be
moved out of ASoC.
Charles Keepax [Mon, 13 Sep 2021 16:00:44 +0000 (17:00 +0100)]
ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed buffers
When parsing a compressed buffer from the firmware the driver currently
open codes reading the firmware coefficient containing the buffer
description. Improve this slightly by using the coefficient read
functions already provided by the wm_adsp driver. It is worth noting
this change requires the running variable to be set before
wm_adsp_buffer_init is called, however this is safe, since its all still
under the power lock and nothing in the compressed code gates itself on
running.
Simon Trimmer [Mon, 13 Sep 2021 16:00:42 +0000 (17:00 +0100)]
ASoC: wm_adsp: Remove use of snd_ctl_elem_type_t
In preparation for moving the generic DSP support out of ASoC, remove
the use of the ALSA specific types for the control type. The use of an
ALSA type was unnecessary, the simplified code is easier to read and
avoids Sparse warnings.
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
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)
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