We have multiple entries for the same codecs, use the new compatible
IDs to have a single entry.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Link: https://lore.kernel.org/r/20211029171409.611600-5-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use comp_ids field to enumerate rt5682/rt5682s headphone codec for
JSL/TGL/ADL devices and remove redundant entries in tables.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Link: https://lore.kernel.org/r/20211029171409.611600-4-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Detect whether the headphone codec is ALC5682I-VS or not in probe
function so we don't need to duplicate all board configs for this new
variant.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Link: https://lore.kernel.org/r/20211029171409.611600-3-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A machine driver needs to be enumerated by more than one ACPI HID if
it supports second headphone driver (i.e. rt5682 and rt5682s).
However, the id field in snd_soc_acpi_mach structure could contain
only one HID. By adding a 'comp_ids' field which can contain several
HIDs, we can enumerate a machine driver by multiple ACPI HIDs.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Link: https://lore.kernel.org/r/20211029171409.611600-2-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case that there are other components assigned to runtime device,
depending on order dummy component can override their params with its
own, which shouldn't happen. Check if there are any other components
assigned to rtd and if so, skip setting hwparams.
Occurs when using topology where 'snd-soc-dummy' gets assigned by
default as codec and platform component.
Alternative approach would be to copy whole dummy handling and rename it
to "snd-soc-null" or something similar. And remove hwparams assignment
to make it really do nothing.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20211015161257.27052-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology needs device for prints and resource allocation. So far,
component->dev is used. However, this may lead to high memory use in
model where card is an independent driver which can be reloaded and
topology is loaded from component's probe() method. Every time machine
driver is reloaded topology is being loaded anew, each time allocating
new memory. Said memory will only be freed when component itself is
being freed.
Address the problem by tying topology to component->card->dev instead,
so memory occupied by the topology is freed whenever related machine
device gets removed.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20211015161257.27052-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_tplg_add_dcontrol() passes device as argument which is later used to
print messages. Align it with all other prints in file to use tplg->dev.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20211015161257.27052-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add sanity checks to make sure the data is read within file boundary.
Helps in situations where file is only partially copied or malformed.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20211015161257.27052-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add sanity check to make sure the data is read within file boundary.
Helps in situations where file is only partially copied or malformed.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20211015161257.27052-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If snd_soc_component_set_jack() is called after
snd_soc_component_remove() it may operate on memory which is freed in
->remove handler.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20211015161257.27052-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
TFA9897 has an internal 'rcv' switch so that it can manage both
loudspeaker and earpiece modes with the same physical speaker.
Signed-off-by: Vincent Knecht <vincent.knecht@mailoo.org>
Link: https://lore.kernel.org/r/20211024085840.1536438-3-vincent.knecht@mailoo.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The acp-platform driver now needs the ACPI helpers:
ld.lld: error: undefined symbol: snd_soc_acpi_find_machine
>>> referenced by acp-platform.c
>>> soc/amd/acp/acp-platform.o:(acp_machine_select) in archive sound/built-in.a
ld.lld: error: undefined symbol: snd_soc_acpi_codec_list
>>> referenced by acp-renoir.c
>>> soc/amd/acp/acp-renoir.o:(snd_soc_acpi_amd_acp_machines) in archive sound/built-in.a
Other drivers using this interface, select SND_SOC_ACPI, so do the
same thing here.
Fixes: e646b51f5d ("ASoC: amd: acp: Add callback for machine driver on ACP")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20211029113714.966823-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Summary: The driver applied the opposite of the DT setting to the
wrong register bit.
The jack plug detect hardware in cs42l42 is somewhat confusing,
compounded by an unclear description in the datasheet. This is most
likely the reason that the driver implemented a DT property for the
wrong register bit, that had the opposite effect of what was
described in the binding.
Changing the meaning of the property values isn't feasible; the
driver dates from 2016 and the risk of breaking out-of-tree configs
is too high (the property is also available to ACPI systems).
So the fix is to make the binding doc match the actual behaviour and
then fix the driver to apply it to the correct register bit.
As a bonus, patch #3 converts the binding to yaml.
Richard Fitzgerald (3):
ASoC: dt-bindings: cs42l42: Correct description of ts-inv
ASoC: cs42l42: Correct configuring of switch inversion from ts-inv
ASoC: dt-bindings: cs42l42: Convert binding to yaml
.../devicetree/bindings/sound/cirrus,cs42l42.yaml | 225 +++++++++++++++++++++
.../devicetree/bindings/sound/cs42l42.txt | 114 -----------
MAINTAINERS | 1 +
sound/soc/codecs/cs42l42.c | 9 +-
4 files changed, 230 insertions(+), 119 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml
delete mode 100644 Documentation/devicetree/bindings/sound/cs42l42.txt
--
2.11.0
Detect the codec variant in probe function and update DAI link
accordingly. Also add an new entry in enumeration table for machine
driver enumeration.
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/20211028140909.496022-1-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The same codec seems to have different personalities. ESSX8316 was
used for Baytrail/CherryTrail, ESSX8336 seems to be used for AppoloLake,
GeminiLake, JasperLake and TigerLake devices.
BugLink: https://github.com/thesofproject/linux/issues/2955
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>-e
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20211029011109.23633-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The setting from the cirrus,ts-inv property should be applied to the
TIP_SENSE_INV bit, as this is the one that actually affects the jack
detect block. The TS_INV bit only swaps the meaning of the PLUG and
UNPLUG interrupts and should always be 1 for the interrupts to have
the normal meaning.
Due to some misunderstanding the driver had been implemented to
configure the TS_INV bit based on the jack switch polarity. This made
the interrupts behave the correct way around, but left the jack detect
block, button detect and analogue circuits always interpreting an open
switch as unplugged.
The signal chain inside the codec is:
SENSE pin -> TIP_SENSE_INV -> TS_INV -> (invert) -> interrupts
|
v
Jack detect,
button detect and
analog control
As the TIP_SENSE_INV already performs the necessary inversion the
TS_INV bit never needs to change. It must always be 1 to yield the
expected interrupt behaviour.
Some extra confusion has arisen because of the additional invert in the
interrupt path, meaning that a value applied to the TS_INV bit produces
the opposite effect of applying it to the TIP_SENSE_INV bit. The ts-inv
property has therefore always had the opposite effect to what might be
expected (0 = inverted, 1 = not inverted). To maintain the meaning of
the ts-inv property it must be inverted when applied to TIP_SENSE_INV.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20211028140902.11786-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Clang warns:
sound/soc/qcom/qdsp6/topology.c:465:3: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough]
default:
^
sound/soc/qcom/qdsp6/topology.c:465:3: note: insert 'break;' to avoid fall-through
default:
^
break;
1 warning generated.
Clang is a little more pedantic than GCC, which permits implicit
fallthroughs to cases that contain just break or return. Clang's version
is more in line with the kernel's own stance in deprecated.rst, which
states that all switch/case blocks must end in either break,
fallthrough, continue, goto, or return. Add the missing break to fix
the warning.
Link: https://github.com/ClangBuiltLinux/linux/issues/1495
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://lore.kernel.org/r/20211027190823.4057382-1-nathan@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When SND_SOC_AMD_RENOIR_MACH or SND_SOC_AMD_RV_RT5682_MACH
are selected, and GPIOLIB is not selected, Kbuild gives
the following warnings, respectively:
WARNING: unmet direct dependencies detected for SND_SOC_DMIC
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n]
Selected by [y]:
- SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y]
and
WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n]
Selected by [y]:
- SND_SOC_AMD_RV_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_ACP3x [=y] && I2C [=y] && CROS_EC [=y]
This is because SND_SOC_DMIC and SND_SOC_MAX98357A are
selected by SND_SOC_AMD_RV_RT5682_MACH and SND_SOC_AMD_RENOIR_MACH,
respectively. However, neither of the selectors depend on or select GPIOLIB,
despite their selectees depending on GPIOLIB.
These unmet dependency bugs were detected by Kismet,
a static analysis tool for Kconfig. Please advise if this
is not the appropriate solution.
Signed-off-by: Julian Braha <julianbraha@gmail.com>
Link: https://lore.kernel.org/r/20211027184835.112916-1-julianbraha@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The series of features will make general and simple for new sof machine driver.
David Lin (2):
ASoC: nau8825: add set_jack coponment support
ASoC: nau8825: add clock management for power saving
sound/soc/codecs/nau8825.c | 48 ++++++++++++++++++++++++++++++++++++--
1 file changed, 46 insertions(+), 2 deletions(-)
We have configurations for this codec on APL, GLK and TGL, somehow the
information that some designs rely on JasperLake was not shared.
BugLink: https://github.com/thesofproject/linux/issues/3210
Fixes: 790049fb66 ('ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20211027023311.25005-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of error, platform_device_register_data() returns ERR_PTR()
and never returns NULL. The NULL test in the return value check
should be replaced with IS_ERR().
Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: e646b51f5d ("ASoC: amd: acp: Add callback for machine driver on ACP")
Signed-off-by: Yang Yingliang <yangyingliang@huawei.com>
Link: https://lore.kernel.org/r/20211027065228.833825-1-yangyingliang@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck reports the following warning:
sound/soc/rockchip/rockchip_i2s_tdm.c:599:9: warning: Identical
condition and return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/rockchip/rockchip_i2s_tdm.c:594:6: note: If condition 'ret'
is true, the function will return/exit
if (ret)
^
sound/soc/rockchip/rockchip_i2s_tdm.c:599:9: note: Returning identical
expression 'ret'
return ret;
^
While the code is not wrong, it's clearer to return 0 directly.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211025185933.144327-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck reports the following warning:
sound/soc/mediatek/mt8195/mt8195-dai-etdm.c:1299:10: style: Variable
'ret' is assigned a value that is never used. [unreadVariable]
int ret = 0;
^
The suggested change aligns the implementation of
mt8195_afe_disable_etdm() with mt8195_afe_enable_etdm() - same
negative return value upon error.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211025185933.144327-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cppcheck warning:
sound/soc/mediatek/common/mtk-afe-fe-dai.c:353:8: style: Variable 'i'
is assigned a value that is never used. [unreadVariable]
int i = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211025185933.144327-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck reports a false positive
sound/soc/codecs/nau8821.c:390:17: error: Array 'dmic_speed_sel[4]'
accessed at index 4, which is out of bounds. [arrayIndexOutOfBounds]
dmic_speed_sel[i].param, dmic_speed_sel[i].val);
^
sound/soc/codecs/nau8821.c:378:2: note: After for loop, i has value 4
for (i = 0 ; i < 4 ; i++)
^
sound/soc/codecs/nau8821.c:390:17: note: Array index out of bounds
dmic_speed_sel[i].param, dmic_speed_sel[i].val);
^
While the code is not incorrect, we can deal with the out-of-bounds
check in a clearer way that makes static analysis happy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211025185933.144327-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
make W=1 reports warnings:
sound/soc/codecs/nau8821.c:1192: warning: Function parameter or member
'component' not described in 'nau8821_set_fll'
sound/soc/codecs/nau8821.c:1192: warning: Function parameter or member
'pll_id' not described in 'nau8821_set_fll'
sound/soc/codecs/nau8821.c:1192: warning: Function parameter or member
'source' not described in 'nau8821_set_fll'
sound/soc/codecs/nau8821.c:1192: warning: Excess function parameter
'codec' description in 'nau8821_set_fll'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211025185933.144327-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Sparse reports the following warnings:
sound/soc/codecs/rt5682s.c:44:12: error: symbol 'rt5682s_supply_names'
was not declared. Should it be static?
sound/soc/codecs/rt5682s.c:74:26: error: symbol 'rt5682s_reg' was not
declared. Should it be static?
sound/soc/codecs/rt5682s.c:2841:30: error: symbol
'rt5682s_aif1_dai_ops' was not declared. Should it be static?
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211025185933.144327-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver probe and remove were inconsistent in what they did to power-down
and neither did all steps. In addition to that, neither function
prevented the interrupt handler from running during and after power-down.
Richard Fitzgerald (2):
ASoC: cs42l42: Reset and power-down on remove() and failed probe()
ASoC: cs42l42: free_irq() before powering-down on probe() fail
sound/soc/codecs/cs42l42.c | 43 +++++++++++++++++++++++++++++++------------
1 file changed, 31 insertions(+), 12 deletions(-)
--
2.11.0
This patch set is to add support for lpass sc7280 based targets.
Upadate compatible name and change of bulk clock voting to optional
clock voting in digital codecs va, rx, tx macro drivers.
Changes Since V3:
-- Removed fixes tag.
-- Change signedoff by sequence.
Changes Since V2:
-- Add Tx macro deafults for lpass sc7280
Changes Since V1:
-- Removed individual clock voting and used bulk clock optional.
-- Removed volatile changes and fixed default values.
-- Typo errors.
Srinivasa Rao Mandadapu (5):
ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for
sc7280
ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital
codecs
ASoC: codecs: tx-macro: Enable tx top soundwire mic clock
ASoC: codecs: tx-macro: Update tx default values
ASoC: codecs: Change bulk clock voting to optional voting in digital
codecs
.../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +++-
.../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +++-
.../bindings/sound/qcom,lpass-va-macro.yaml | 4 +++-
.../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +++-
sound/soc/codecs/lpass-rx-macro.c | 3 ++-
sound/soc/codecs/lpass-tx-macro.c | 25 +++++++++++++++++++---
sound/soc/codecs/lpass-va-macro.c | 3 ++-
sound/soc/codecs/lpass-wsa-macro.c | 1 +
8 files changed, 39 insertions(+), 9 deletions(-)
--
Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc.,
is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.
Hi Mark,
This version is a respin of v10 fixing a build error in 12/17 patch.
QCOM SoC relevant non-audio patches in this series has been merged into
the Qualcomm drivers-for-5.16 tree, as this series depends those patches
an immutable tag is available at:
https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.gittags/20210927135559.738-6-srinivas.kandagatla@linaro.org
This patchset adds ASoC driver support to configure signal processing
framework ("AudioReach") which is integral part of Qualcomm next
generation audio SDK and will be deployed on upcoming Qualcomm chipsets.
It makes use of ASoC Topology to load graphs on to the DSP which is then
managed by APM (Audio Processing Manager) service to prepare/start/stop.
Here is simplified high-level block diagram of AudioReach:
___________________________________________________________
| CPU (Application Processor) |
| +---------+ +---------+ +----------+ |
| | q6apm | | q6apm | | q6apm | |
| | dais | <------> | | <-----> |lpass-dais| |
| +---------+ +---------+ +----------+ |
| ^ ^ |
| | | +---------+ |
| +---------+ v +---------->|topology | |
| | q6prm | +---------+ | | |
| | |<-------->| GPR | +---------+ |
| +---------+ +---------+ |
| ^ ^ |
| | | |
| +----------+ | |
| | q6prm | | |
| |lpass-clks| | |
| +----------+ | |
|____________________________|______________________________|
|
| RPMSG (IPC over GLINK)
____________________________|______________________________
| | |
| +-----------------------+ |
| | | |
| v v q6 (Audio DSP) |
|+-----+ +----------------------------------+ |
|| PRM | | APM (Audio Processing Manager) | |
|+-----+ | . Graph Management | |
| | . Command Handing | |
| | . Event Management | |
| | ... | |
| +----------------------------------+ |
| ^ |
|____________________________|______________________________|
|
| LPASS AIF
____________________________|______________________________
| | Audio I/O |
| v |
| +--------------------------------------------------+ |
| | Audio devices | |
| | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| |
| | | |
| +--------------------------------------------------+ |
|___________________________________________________________|
AudioReach has constructs of sub-graph, container and modules.
Each sub-graph can have N containers and each Container can have N Modules
and connections between them can be linear or non-linear.
An audio function can be realized with one or many connected
sub-graphs. There are also control/event paths between modules that can
be wired up while building graph to achieve various control mechanism
between modules. These concepts of Sub-Graph, Containers and Modules
are represented in ASoC topology.
Here is simple I2S graph with a Write Shared Memory and a
Volume control module within a single Subgraph (1) with one Container (1)
and 5 modules.
____________________________________________________________
| Sub-Graph [1] |
| _______________________________________________________ |
| | Container [1] | |
| | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| |
| |_______________________________________________________| |
|____________________________________________________________|
For now this graph is split into two subgraphs to achieve dpcm like below:
________________________________________________ _________________
| Sub-Graph [1] | | Sub-Graph [2] |
| ____________________________________________ | | _____________ |
| | Container [1] | | | |Container [2]| |
| | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | |
| |____________________________________________| | | |_____________| |
|________________________________________________| |_________________|
_________________
| Sub-Graph [3] |
| _____________ |
| |Container [3]| |
| | [DMA-EP] | |
| |_____________| |
|_________________|
This patchset adds very minimal support for AudioReach which includes
supporting sub-graphs containing CODEC DMA ports and simple PCM
Decoder/Encoder and Logger Modules. Additional capabilities will
be built over time to expose features offered by AudioReach.
This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5
and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via
VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio
via I2S.
First 10 Patches are mostly reorganization existing Old QDSP Audio
Framework code and bindings so that we could reuse them on AudioReach.
ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at
https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/
and Qualcomm AudioReach DSP headers are available at:
https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-headers
Note: There is one false positive warning in this patchset:
audioreach.c:80:45: warning: array of flexible structures
Thanks,
srini
Changes since v10:
- fix build error during arm64 defconfig build reported by Mark in 12/17 patch
for audioreach_tplg_init symbol
Srinivas Kandagatla (17):
ASoC: dt-bindings: move LPASS dai related bindings out of q6afe
ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe
ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h
ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file
ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file
ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible
ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible
ASoC: dt-bindings: add q6apm digital audio stream bindings
ASoC: qdsp6: audioreach: add basic pkt alloc support
ASoC: qdsp6: audioreach: add q6apm support
ASoC: qdsp6: audioreach: add module configuration command helpers
ASoC: qdsp6: audioreach: add Kconfig and Makefile
ASoC: qdsp6: audioreach: add topology support
ASoC: qdsp6: audioreach: add q6apm-dai support
ASoC: qdsp6: audioreach: add q6apm lpass dai support
ASoC: qdsp6: audioreach: add q6prm support
ASoC: qdsp6: audioreach: add support for q6prm-clocks
.../devicetree/bindings/sound/qcom,q6afe.txt | 181 ---
.../bindings/sound/qcom,q6apm-dai.yaml | 53 +
.../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++
.../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++
include/dt-bindings/sound/qcom,q6afe.h | 203 +--
.../sound/qcom,q6dsp-lpass-ports.h | 208 +++
include/uapi/sound/snd_ar_tokens.h | 208 +++
sound/soc/qcom/Kconfig | 22 +
sound/soc/qcom/qdsp6/Makefile | 11 +-
sound/soc/qcom/qdsp6/audioreach.c | 1130 +++++++++++++++++
sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++
sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +--
sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +---------
sound/soc/qcom/qdsp6/q6apm-dai.c | 416 ++++++
sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++
sound/soc/qcom/qdsp6/q6apm.c | 822 ++++++++++++
sound/soc/qcom/qdsp6/q6apm.h | 152 +++
sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++
sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 +
sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++
sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 +
sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++
sound/soc/qcom/qdsp6/q6prm.c | 202 +++
sound/soc/qcom/qdsp6/q6prm.h | 78 ++
sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++
25 files changed, 6664 insertions(+), 1227 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml
create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml
create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml
create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
create mode 100644 include/uapi/sound/snd_ar_tokens.h
create mode 100644 sound/soc/qcom/qdsp6/audioreach.c
create mode 100644 sound/soc/qcom/qdsp6/audioreach.h
create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c
create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
create mode 100644 sound/soc/qcom/qdsp6/q6apm.c
create mode 100644 sound/soc/qcom/qdsp6/q6apm.h
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c
create mode 100644 sound/soc/qcom/qdsp6/q6prm.c
create mode 100644 sound/soc/qcom/qdsp6/q6prm.h
create mode 100644 sound/soc/qcom/qdsp6/topology.c
--
2.21.0
All configuration symbols for AMD Audio ACP conponents depend on X86 &&
PCI, except for SND_SOC_AMD_ACP_COMMON. Add a dependency on X86 && PCI
to SND_SOC_AMD_ACP_COMMON, to prevent asking the user about AMD Audio
ACP support when configuring a kernel without X86 or PCI support.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/30fcedce513186bf89f1f2655b665298250fdc66.1635260849.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
If not all of CONFIG_X86, CONFIG_PCI, and CONFIG_I2C are set:
WARNING: unmet direct dependencies detected for SND_SOC_AMD_MACH_COMMON
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && X86 && PCI [=y] && I2C [=y]
Selected by [y]:
- SND_SOC_AMD_LEGACY_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y]
- SND_SOC_AMD_SOF_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y]
As SND_SOC_AMD_MACH_COMMON depends on X86 && PCI && I2C, all symbols
selecting it should depend on X86 && PCI && I2C, too.
Fixes: 9d8a7be88b ("ASoC: amd: acp: Add legacy sound card support for Chrome audio")
Fixes: 9f84940f50 ("ASoC: amd: acp: Add SOF audio support on Chrome board")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/dfb03bd33117e26f3f04ce227bb28095109b3d80.1635260849.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
The build only descends into sound/soc/amd/acp/ if
CONFIG_SND_SOC_AMD_ACP_COMMON=y. Hence all later config symbols should
depend on SND_SOC_AMD_ACP_COMMON, to prevent asking the user about
config symbols for driver code that won't be build anyway.
Fixes: 623621a9f9 ("ASoC: amd: Add common framework to support I2S on ACP SOC")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/53d1d63bed1865293e6f5085ead21cdbb068fb15.1635260849.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
Relying on devm to free the irq handler on probe failure leaves a
small window of opportunity for an interrupt to become pending and
then the handler to run after the chip has been reset and powered
off.
For safety cs42l42_probe() should free the irq in the error path.
As the irq is now disabled by the driver in probe() and remove()
there is no point allocating it as a devres-managed item, so
convert to plain non-devres.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211026125722.10220-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver remove() should assert RESET and disable the supplies.
probe() fail was disabling supplies but it didn't assert reset or
put the codec into a power-down state.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211026125722.10220-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to q6prm (Proxy Resource Manager) module used for clock resources
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-17-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to Audio port dais on LPASS Audio IP using
existing common q6dsp-lpass-ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-16-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to pcm dais in Audio Process Manager.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-15-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that all the code for audioreach and q6apm are in at this point to be
able to compile, start adding Kconfig and Makefile changes.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-13-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Audioreach module configuration helpers, which will be used by
the q6apm-dai driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-12-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to q6apm (Audio Process Manager) component which is
core Audioreach service running in the DSP.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-11-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Move common parts of q6afe-clocks to q6dsp-lpass-clocks so that we could
reuse most of the driver for new Q6DSP audio frameworks.
This is to make the code reuseable for new Q6DSP AudioReach framework.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-6-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Various Q6DSP frameworks will use LPASS Audio IP, so move all the hardware
specific details to a common file so that they could be reused across
multiple Q6DSP frameworks.
In this case all the audio ports definitions can be moved to a common file
to be able to reuse across multiple Q6DSP frameworks.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211026111655.1702-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Change bulk clock frequency voting to optional bulk voting in va, rx and tx macros
to accommodate both ADSP and ADSP bypass based lpass architectures.
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Co-developed-by: Venkata Prasad Potturu <potturu@codeaurora.org>
Signed-off-by: Venkata Prasad Potturu <potturu@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1635234188-7746-6-git-send-email-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Adjust dapm widget to manage clock from power event for power saving.
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Link: https://lore.kernel.org/r/20211025113857.3860951-3-CTLIN0@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use set_jack ops to set jack for new machine drivers. Meanwhile,
the old machine drivers can still call previous export function
"nau8825_enable_jack_detect".
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Link: https://lore.kernel.org/r/20211025113857.3860951-2-CTLIN0@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, the DAI link nodes in the device tree always have to be
specified completely in each device tree. However, the available
interfaces (e.g. Primary/Secondary/Tertiary/Quaternary MI2S) are common
for all devices of a SoC, so the majority of the definitions can be
placed in a common device tree include to reduce boilerplate.
Make it possible to define such stubs in device tree includes by
respecting the "status" property for the DAI link nodes. This is
a trivial change that just requires switching to the _available_
OF functions that check the "status" property additionally.
This allows defining a stub like:
sound_dai_quaternary: dai-link-quaternary {
link-name = "Quaternary MI2S";
status = "disabled"; /* Needs extra codec configuration */
cpu {
sound-dai = <&q6afedai QUATERNARY_MI2S_RX>;
};
platform {
sound-dai = <&q6routing>;
};
};
where the codec would be filled in by the device-specific device tree.
For existing device trees this change does not make any difference.
A missing "status" property is treated like status = "okay".
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211025105503.49444-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The interrupt handling code was getting the struct device* from a
struct snd_soc_component* stored in struct cs42l42_private. If the
interrupt was asserted before ASoC calls component_probe() the
snd_soc_component* will be NULL.
The stored snd_soc_component* is not actually used for anything other
than indirectly getting the struct device*. Remove it, and store the
struct device* in struct cs42l42_private.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211025112258.9282-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the existing devm_clk_get_optional() helper instead of building a
similar construct on top of devm_clk_get() that fails to handle all
errors but -EPROBE_DEFER.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/c2a8a1a628804a4439732d02847e25c227083690.1634565564.git.geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
When SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH is selected,
and GPIOLIB is not selected, Kbuild gives the
following warnings:
WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n]
Selected by [y]:
- SND_SOC_INTEL_DA7219_MAX98357A_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y]
WARNING: unmet direct dependencies detected for SND_SOC_DMIC
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n]
Selected by [y]:
- SND_SOC_INTEL_DA7219_MAX98357A_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y]
WARNING: unmet direct dependencies detected for SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && GPIOLIB [=n]
Selected by [y]:
- SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && SND_SOC_INTEL_KBL [=y] && I2C [=y] && ACPI [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n])
This is because SND_SOC_DMIC and SND_SOC_MAX98357A are
selected by SND_SOC_INTEL_DA7219_MAX98357A_GENERIC, which
is also selected by SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH.
However, the selectors do not depend on or select GPIOLIB,
despite SND_SOC_DMIC and SND_SOC_MAX98357A depending on GPIOLIB.
These unmet dependency bugs were detected by Kismet,
a static analysis tool for Kconfig. Please advise if this
is not the appropriate solution.
Signed-off-by: Julian Braha <julianbraha@gmail.com>
Link: https://lore.kernel.org/r/20211025010615.10070-1-julianbraha@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Up to now cs35l41_remove() returns zero unconditionally. Make it
return void instead which makes it easier to see in the callers that
there is no error to handle.
Also the return value of i2c, platform and spi remove callbacks is
ignored anyway.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Link: https://lore.kernel.org/r/20211020132416.30288-1-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
A driver with a remove callback that just returns 0 behaves identically
to a driver with no remove callback at all. So simplify accordingly.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211020125726.22946-1-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
The default card name for Trimslice device should be "tegra-trimslice".
It got lost by accident during unification of machine sound drivers,
fix it.
Cc: <stable@vger.kernel.org>
Fixes: cc8f70f560 ("ASoC: tegra: Unify ASoC machine drivers")
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20211024192853.21957-2-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The device-tree of AC97 codecs need to be parsed differently from I2S
codecs, plus codec device may need to be created. This was missed by the
patch that unified machine drivers into a single driver, fix it. It should
restore audio on Toradex Colibri board.
Cc: <stable@vger.kernel.org>
Fixes: cc8f70f560 ("ASoC: tegra: Unify ASoC machine drivers")
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20211024192853.21957-1-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A driver with a remove callback that just returns 0 behaves identically
to a driver with no remove callback at all. So simplify accordingly.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211020125803.23117-1-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
On the Amlogic AXG series, the TODDR FIFO may get out of sync with the TDM
decoder if the decoder is started before the FIFO. The channel appears
shifted in memory in an unpredictable way.
To fix this, the trick is to start the FIFO before the TDM decoder. This
way the FIFO is already waiting when the 1st channel is produced and it is
correctly placed in memory.
Jerome Brunet (2):
ASoC: meson: axg-card: make links nonatomic
ASoC: meson: axg-tdm-interface: manage formatters in trigger
sound/soc/meson/axg-card.c | 1 +
sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++++++-----
2 files changed, 22 insertions(+), 5 deletions(-)
--
2.33.0
So far, the formatters have been reset/enabled using the .prepare()
callback. This was done in this callback because walking the formatters use
a mutex so it could not be done in .trigger(), which is atomic by default.
It turns out there is a problem on capture path of the AXG series.
The FIFO may get out of sync with the TDM decoder if the IP are not enabled
in a specific order. The FIFO must be enabled before the formatter starts
producing data. IOW, we must deal with FE before the BE. The .prepare()
callback is called on the BEs before the FE so it is not OK for the AXG.
The .trigger() callback order can be configured, and it deals with the FE
before the BEs by default. To solve our problem, we just need to start and
stop the formatters from the .trigger() callback. It is OK do so now that
the links have been made 'nonatomic' in the card driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211020114217.133153-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Non atomic operations need to be performed in the trigger callback
of the TDM interfaces. Those are BEs but what matters is the nonatomic
flag of the FE in the DPCM context. Just set nonatomic for everything so,
at least, it is clear.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211020114217.133153-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has runtime_suspend and runtime_resume callbacks, but
pm_runtime is never enabled so these functions won't be called. They
could not be used anyway because the runtime_suspend would cause jack
detect to stop working.
These functions are unused - delete them.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211018164431.5871-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When SND_SOC_SC7180 or SND_SOC_STORM is selected,
and GPIOLIB is not selected, Kbuild gives the following
warning:
WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n]
Selected by [y]:
- SND_SOC_STORM [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y]
- SND_SOC_SC7180 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && I2C [=y]
This is because SND_SOC_MAX98357A is selected
by SND_SOC_STORM and SND_SOC_SC7180, but
these config options do not select or depend on
GPIOLIB, despite SND_SOC_MAX98357A depending on
GPIOLIB.
These unmet dependency bugs were detected by Kismet,
a static analysis tool for Kconfig. Please advise if this
is not the appropriate solution.
Signed-off-by: Julian Braha <julianbraha@gmail.com>
Link: https://lore.kernel.org/r/20211010215627.17869-1-julianbraha@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use clk_prepare_enable()/clk_disable_unprepare() in preparation for switch
to Common Clock Framework, otherwise the following is visible:
WARNING: CPU: 0 PID: 97 at drivers/clk/clk.c:1011 clk_core_enable+0x9c/0xbc
Enabling unprepared mclk
...
Hardware name: Cirrus Logic EDB9302 Evaluation Board
...
clk_core_enable
clk_core_enable_lock
ep93xx_i2s_hw_params
snd_soc_dai_hw_params
soc_pcm_hw_params
snd_pcm_hw_params
snd_pcm_ioctl
...
Signed-off-by: Alexander Sverdlin <alexander.sverdlin@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20211018103105.146380-2-alexander.sverdlin@gmail.com'
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
This patchset adds support for testing WCD938X connected via TX and RX Macros
on SM8250 MTP.
Srinivas Kandagatla (2):
ASoC: qcom: sm8250: add support for TX and RX Macro dais
ASoC: qcom: sm8250: Add Jack support
sound/soc/qcom/sm8250.c | 79 +++++++++++++++++++++++++++++++++++++++++
1 file changed, 79 insertions(+)
--
2.21.0
In newer variants primary codec is rt5682vs. Add support for newer
codec variants in generic machine driver module and define driver
data to register SOF sound card.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-9-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In newer chrome boards we have max98360a as an amplifier codec.
Add support for max98360a in generic machine driver and configure
driver data to enable SOF sound card support on newer boards .
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-8-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Chrome board has RT5682 codec and RT1019 amp connected to I2S SP
controller on ACP hw. Also it support DMIC capture endpoints with
inbuilt pdm controller on ACP hw block. Add driver module to create
backend dai links for sof dsp core. We pass driver data with audio
end points configuration to register sound cards and create device
nodes for all audio endpoints.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-7-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Renoir based Chrome board has RT5682 as primary headset codec and
RT1019 amp device connected to I2SSP ACP i2s controller. Add driver
to register legacy sound card devices on Chrome board.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-6-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have machines with different audio endpoints configurations
across various distributions. We need to support multiple sound
cards for different combinations of I2S instance and codecs hw.
Now we also need to support SOF-DSP endpoints based sound cards.
All such card combinations slightly differs in terms of machine
ops callback. This patch adds ACP generic machine driver module
that exposes method to create ACP cards dai links and define new
ops for audio endpoints configurations. Initially we have added
dailink support for RT5682 and RT1019 codec connection with ACP
I2S_SP instance. We will add newer codecs in this module to use
this for all AMD's ACP block sound cards supports in future.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-5-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add I2S dai driver for Renoir platform and register with common
acp framework to support non dsp I2S use case on Renoir.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-3-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We are using legacy way of exposing dais and DMA configuration that
requires separate driver modules for various ACP SOC with almost
similar hw configuration. Moreover the legacy approach requires
separate I2S and DMA module platform devices registration and need
machine specific quirk to control various I2S endpoints. Add generic
dai driver and platform driver for I2S controller on ACP hw block.
This common framework can be used by various ACP platform devices
that shares common specs.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211019070938.5076-2-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 75b31192fe.
The original purpose of customized pcm was to config prealloc buffer size
flexibly. but, we can do the same thing by soc-generic-dmaengine-pcm.
And the generic one can generated the better config by querying DMA
capabilities from dmaengine driver rather than the Hard-Coded one.
e.g.
the customized one:
static const struct snd_pcm_hardware snd_rockchip_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED,
...
the generic one:
ret = dma_get_slave_caps(chan, &dma_caps);
if (ret == 0) {
if (dma_caps.cmd_pause && dma_caps.cmd_resume)
hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
hw.info |= SNDRV_PCM_INFO_BATCH;
...
So, let's revert back to use the generic dmaengine pcm.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Reviewed-by: John Keeping <john@metanate.com>
Link: https://lore.kernel.org/r/1632792957-80428-1-git-send-email-sugar.zhang@rock-chips.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The soc_intel_is_foo() helpers from
sound/soc/intel/common/soc-intel-quirks.h are useful outside of the
sound subsystem too.
Move these to include/linux/platform_data/x86/soc.h, so that
other code can use them too.
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211018143324.296961-2-hdegoede@redhat.com
WCD938X on SM8250 MTP is connected via TX macro which has MBHC support,
So add this jack support in the soundcard driver too.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211006172745.22103-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
On SM8250 MTP boards WCD938x codec is connected via TX and RX Macros,
so add support for this dais in the soundcard driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211006172745.22103-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Yellow Carp ACP6x drivers can be built by selecting necessary
kernel config option.
The patch enables build support of the same.
Signed-off-by: Vijendar Mukunda<Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211018112044.1705805-11-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add ACP6x irq handler for handling irq events for ACP IP.
Add pdm irq events handling.
Whenever audio data equal to the PDM watermark level are consumed,
interrupt is generated. Acknowledge the interrupt.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211018112044.1705805-7-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
PDM platform driver binds to the platform device created by
ACP6x PCI device. PDM driver registers ALSA DMA and CPU DAI
components with ASoC framework.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211018112044.1705805-6-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ACP6.x IP has PDM decoder block.
Create a platform device for it, so that the PDM platform driver
can be bound to this device.
Pass PCI resources like MMIO to this platform device.
Create a platform device for generic dmic codec driver.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211018112044.1705805-5-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During development of V5 of the i2s-tdm patch series, I replaced
the atomic refcount with a regular integer, as it was only ever
accessed within a spinlock.
Foolishly, I got the semantics of atomic_dec_and_test wrong, which
resulted in a test for 0 actually becoming a test for >0.
The result was that setting the audio frequency broke; switching
from 44100 Hz audio playback to 96000 Hz audio playback would
garble the sound most unpleasantly.
Fix this by checking for --refcount == 0, which is what it should
have been all along.
Fixes: 081068fd64 ("ASoC: rockchip: add support for i2s-tdm controller")
Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas@gmail.com>
Link: https://lore.kernel.org/r/20211015210730.308946-1-frattaroli.nicolas@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement driver_name to provide an alternative to card_name for userspace
configuration of Amlogic audio cards.
Suggested-by: Matthias Reichl <hias@horus.com>
Signed-off-by: Christian Hewitt <christianshewitt@gmail.com>
Acked-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211017160028.23318-1-christianshewitt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Up to now aic32x4_remove() returns zero unconditionally. Make it return
void instead which makes it easier to see in the callers that there is
no error to handle.
Also the return value of i2c and spi remove callbacks is ignored anyway.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Link: https://lore.kernel.org/r/20211015071113.2795767-1-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Do nothing if format was zero at snd_soc_runtime_set_dai_fmt().
soc-core.c can be more simple code by this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ee8jt7d3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAI active count is not exchanged during for_each_rtd_dais()
loops. We don't need to keep snd_soc_dai_stream_active() as
"active" on soc_pcm_hw_clean(). This patch avoid verbose code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ilxvt7e6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_hw_clean() is using "continue" during for_each_rtd_dais(),
but it is very verbose. This patch cleanup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k0ibt7ej.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In cases where both rx and tx lrck are synced to the same source,
the resets for rx and tx need to be triggered simultaneously,
according to the downstream driver.
As there is no reset API to atomically bulk (de)assert two resets
at once, what the driver did was implement half a reset controller
specific to Rockchip, which tried to write the registers for the
resets within one write ideally or several writes within an irqsave
section.
This of course violates abstractions quite badly. The driver should
not write to the CRU's registers directly.
In practice, for the cases I tested the driver with, which is audio
playback, replacing the synchronised asserts with just individual
ones does not seem to make any difference.
If it turns out that this breaks something in the future, it should
be fixed through the specification and implementation of an atomic
bulk reset API, not with a CRU hack.
Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas@gmail.com>
Reviewed-by: Heiko Stuebner <heiko@sntech.de>
Message-Id: <20211016105354.116513-2-frattaroli.nicolas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi Mark
We already have Audio-Graph-Card which is Of-Graph base general sound
card driver. Basically it supports basic CPU-Codec connection, and is
also supporting DPCM connection. Because it was forcibly expanded to
DPCM, DT parsing is very limited and very difficult to add new features
on it, for example Multi-CPU/Codec support, Codec2Codec support, etc.
This patch adds more flexible new Audio-Graph-Card2 driver for it.
Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have
full compatibility.
The reason why I need Audio-Graph-Card2 instead of updating Audio-Graph-Card
is that it is very difficult to keep compatibility.
Audio-Graph-Card2 supports Normal/DPCM/Codec2Codec Connection wich
Single/Multi DAIs. And it is possible to Customizing.
This patch-set adds Audio-Graph-Card2 driver and its custom driver
sample, and DT settings sample which can be used for testing.
To enable testing/debuging, this patch-set also adds Test-Component
driver. We already have Dummy Component and/or Dummy DAI on soc-utils,
but 1) we can't use it from DT, 2) it do nothing.
Added new Test-Component can be used from DT, and it can indicate called
function name. We can use it to trace callback order, understanding
ALSA SoC behavior, etc, etc...
Sample DT settings of Audio Graph Card2 is using Test-Component as CPU/Codec DAI.
You can easily try to use/test it if you added below line to your DT file.
Your .config needs to have below CONFIGs to use/test it.
It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec
connections.
#include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi"
CONFIG_SND_AUDIO_GRAPH_CARD2
CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE
CONFIG_SND_TEST_COMPONENT
Because Audio Graph Card2 is still under experimental stage, it will
indicate such warning when probing, and the DT might be updated/exchanged.
It can use Codec2Codec, but it will start automatically when probed,
and can't stop it so far. It should be updated.
Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87tuitusy4.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87a6jn56x0.wl-kuninori.morimoto.gx@renesas.com
v1 -> v2
- don't use "port" base for_each loop
v2 -> v3
- Rename audio-graph-card2 to rich-graph-card
- Rename DSP to DPCM not to confuse
- Normal/DPCM/Codec2Codec can use Single/Multi DAIs.
- use dpcm/multi/codec2codec node instead of using extra compatible
- Sample DTSI patch is separated to Single/Multi.
v3 -> v4
- Rename rich-graph-card to audio-graph-card2
- fixup custom sample driver's connection bug
- test-component compatible uses "verbose" instead of "vv"
v4 -> v5
- tidyup git-log comment at
- tidyup Custom Sample comment
Kuninori Morimoto (16):
ASoC: test-component: add Test Component YAML bindings
ASoC: test-component: add Test Component for Sound debug/test
ASoC: simple-card-utils: add asoc_graph_is_ports0()
ASoC: simple-card-utils: add codec2codec support
ASoC: add Audio Graph Card2 driver
ASoC: audio-graph-card2: add Multi CPU/Codec support
ASoC: audio-graph-card2: add DPCM support
ASoC: audio-graph-card2: add Codec2Codec support
ASoC: add Audio Graph Card2 Yaml Document
ASoC: add Audio Graph Card2 Custom Sample
ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Single)
ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Nulti)
ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Single)
ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Multi)
ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Single)
ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Multi)
.../bindings/sound/audio-graph-card2.yaml | 57 +
.../bindings/sound/test-component.yaml | 33 +
include/sound/graph_card.h | 21 +
include/sound/simple_card_utils.h | 4 +
sound/soc/generic/Kconfig | 20 +
sound/soc/generic/Makefile | 6 +
.../generic/audio-graph-card2-custom-sample.c | 183 +++
.../audio-graph-card2-custom-sample.dtsi | 227 +++
sound/soc/generic/audio-graph-card2.c | 1281 +++++++++++++++++
sound/soc/generic/simple-card-utils.c | 46 +-
sound/soc/generic/test-component.c | 659 +++++++++
11 files changed, 2536 insertions(+), 1 deletion(-)
create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml
create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml
create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.c
create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi
create mode 100644 sound/soc/generic/audio-graph-card2.c
create mode 100644 sound/soc/generic/test-component.c
--
2.25.1
A colletion of smallish mostly driver specific fixes, the biggest thing
here is fixing some of the core code to generate change notifications
properly when writing to controls which will fix issues with UIs not
showing the correct values.
There's one build fix here with a slightly misleading changelog saying
it's adding IRQ config support, it's adding a missing select of the
regmap-irq code rather than adding a feature.
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Merge tag 'asoc-fix-v5.15-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.15
A colletion of smallish mostly driver specific fixes, the biggest thing
here is fixing some of the core code to generate change notifications
properly when writing to controls which will fix issues with UIs not
showing the correct values.
There's one build fix here with a slightly misleading changelog saying
it's adding IRQ config support, it's adding a missing select of the
regmap-irq code rather than adding a feature.
The headset type detection must run to set the analogue switches
correctly for the attached headset type. Without this only headsets
with wiring matching the chip default will have a functioning mic.
commit c26a5289e8 ("ASoC: cs42l42: Add support for set_jack calls")
moved the interrupt unmasking to the component set_jack() callback.
But it's not mandatory for a machine driver to register a struct
snd_soc_jack handler. Without a registered handler the type detection
would not have run and so the mic would not work on some types of
headset.
This patch restores the unmasking of TS_PLUG and TS_UNPLUG interrupts
during probe.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-17-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver must free the IRQ in remove() to prevent the potential race
where an IRQ starts to be handled while the driver is being removed but
devres has not yet called free_irq(). However, the driver can run without
an interrupt but devm_free_irq() will hit a WARN() if no devres-managed
interrupt was ever created.
Fix this by only attempting to create the interrupt handler if the hardware
config specified an interrupt, and failing probe() if the interrupt could
not be created. This means that in cs42l42_remove() an interrupt must have
been registered if the irq number is valid and therefore it is safe to call
devm_free_irq().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-16-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the datasheet the SRC MCLK must be as near as possible to
(125 * sample rate). This means it should be ~6MHz for rates up to 48k
and ~12MHz for rates above that. As per datasheet table 4-21.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-14-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
After enabling the HP or ADC by writing the corresponding PDN=0,
it takes around 20 milliseconds for it to power up and the midrail
supply to be stable. Add this wait into a DAPM widget callback.
If HP and ADC are both powering up in a DAPM sequence, there's no
need to do the wait twice. The widget will perform one wait in the
POST_PMU if there was a PRE_PMU for one or both.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-13-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It isn't possible to switch MCLK between 12MHz and 24MHz rate groups
on-the-fly - this can only be done when cs42l42 is powered-down.
All "normal" SCLK rates use an MCLK in the 12MHz group, so change the
configs for SCLK > 12.288 MHz to use the PLL to generate an MCLK in
the 12MHz group.
As this means MCLK_DIV is always 0 it can be removed from the pll
configuration setup.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-12-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver currently only supports configuring for sample rates <= 96k
and it isn't possible to setup a configuration that will support all
sample rates up to 192k.
For sample rates up to 96k MCLK is in the 12MHz group.
However, although 192k only requires an I2S clock in the 12MHz group,
the cs42l42 audio path is not natively 192k so the audio must be
resampled. But for 192k the SRC requires a 24MHz MCLK.
It is not possible to switch MCLK between 12MHz and 24MHz groups
on-the-fly. The 12MHz group supports all sample rates up to 96k.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-11-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver can run without an interrupt so if devm_request_threaded_irq()
failed, the probe() just carried on. But if this was EPROBE_DEFER the
driver would continue without an interrupt instead of deferring to wait
for the interrupt to become available.
Fixes: 2c394ca796 ("ASoC: Add support for CS42L42 codec")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
An I2S frame always has two slots (left and right) even when sending
mono. The right channel (channel 2) of ASP TX will always have the
same bit width as the left channel and will always be on the high
phase of LRCLK.
The previous implementation always passed the field masks for both
channels to snd_soc_component_update_bits() but for mono the written value
only contained the settings for channel 1. The result was that for mono
channel 2 was set to 8-bit (which is an invalid configuration) with both
channels on the low phase of LRCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 585e7079de ("ASoC: cs42l42: Add Capture Support")
Link: https://lore.kernel.org/r/20211015133619.4698-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When capture and playback substreams are both running at the same time,
cs42l42_pcm_hw_params() would be called for each direction. The first
call will configure the PLL. The second call must not write the PLL
configuration registers again if the first substream is already running,
as this could destabilize the PLL.
The DAI is marked symmetric sample bits and sample rate, so the two
directions will always have the same SCLK (I2S always has 2 channel slots
so the DAI does not need to require symmetric channels to guarantee the
same SCLK). However, since cs42l42_pll_config() is checking for an active
stream it may as well test that the requested SCLK is the same as the
currently active configuration.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently errors on register read/write/update are reported with
an error code and the corresponding function but does not provide
any details on the which register number did it actually fail.
register number can give better clue and it should be easy to
locate the code and fix.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211014161330.26645-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Codec2Codec-Multi sample to audio-graph-card2-custom-sample.dtsi.
Because it can use very basic connection only for now,
it can use only
- 2channels
- S32_LE format
Test-Component driver has "IN" and "OUT" widget. Thus the route is
+--+ +-+
| | | |- Codec8 <- IN
| | <- | |- Codec9 <- IN
| | +-+
| |
| | +-+
| | -> | |- Codec10 -> OUT
| | | |- Codec11 -> OUT
+--+ +-+
One note here is that it will start works when it boot.
In other words we can't stop it so far.
We need to update driver for it in the future.
...
asoc-audio-graph-card2-custom-sample: multicodec <-> multicpu mapping ok
test-component test_codec: test_dai_startup() : test_codec.9
test-component test_codec: test_dai_startup() : test_codec.8
test-component test_codec: test_dai_startup() : test_codec.11
test-component test_codec: test_dai_startup() : test_codec.10
...
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87mtnelu2k.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Codec2Codec-Single sample to audio-graph-card2-custom-sample.dtsi.
Because it can use very basic connection only for now,
it can use only
- 2channels
- S32_LE format
Test-Component driver has "IN" and "OUT" widget. Thus the route is
+--+
| | <-- Codec6 <-- IN
| | --> Codec7 --> OUT
+--+
One note here is that it will start works when it boot.
In other words we can't stop it so far.
We need to update driver for it in the future.
...
asoc-audio-graph-card2-custom-sample: test_codec.7 <-> test_codec.6 mapping ok
test-component test_codec: test_dai_startup() : test_codec.6
test-component test_codec: test_dai_startup() : test_codec.7
...
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o87ulu2o.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds DPCM link Multi-CPU/Codec sample to
audio-graph-card2-custom-sample.dtsi.
This sample is assuming MIXer connection.
One note is that Multi-FE is not supported on ASoC
FE BE
**** +-+
CPU5 -- * * -- | | -- Codec4
CPU6 -- * * | | -- Codec5
**** +-+
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pmsalu2s.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds DPCM link Single-CPU/Codec sample to
audio-graph-card2-custom-sample.dtsi.
This sample is assuming MIXer connection.
FE BE
****
CPU3 -- * * -- Codec3
CPU4 -- * *
****
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r1cqlu2w.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio Graph Card2 settings is a little bit difficult for beginner,
and Customizing it also difficult/confusable too.
So, this patch adds sample for it.
You can easily use it by adding below line on your DT file,
and select CONFIGs to your .config.
#include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi"
CONFIG_SND_AUDIO_GRAPH_CARD2
CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE
CONFIG_SND_TEST_COMPONENT
This patch uses audio-graph-card2 base custom sample driver.
You can directly use audio-graph-card2 instead of custom sample driver
by modifing compatible.
- compatible = "audio-graph-card2-custom-sample";
+ compatible = "audio-graph-card2";
Sample custom driver will indicate customized print.
It is using Test-Component driver for CPU/Codec.
It can indicate more detail print of each behavior if user want to.
In such case, you need to update compatible to "xxx-nv" or "xxx-vv".
- compatible = "test-cpu";
+ compatible = "test-cpu-verbose";
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tuhmlu35.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card2 has customizing support.
This means user can re-use audio-graph-card2 DT parsing, and possible
to expand to own special handling.
This patch adds Audio Graph Card2 Customize Sample Driver.
It can re-use audio-graph-card2 parsing by calling
audio_graph2_parse_of(...), and user can expand each functions by
using hooks.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v922lu3c.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Codec2Codec support to audio-graph-card2.
It can use Codec2Codec but very simple case only for now.
It doesn't have "SWITCH" control yet, thus it start automatically
when it was probed, and can't stop, so far.
Thus it needs to be updated around widgets/routing handling,
and you need to understand that it is under experimental.
Codec has SND_SOC_DAPM_INPUT() (= IN) / SND_SOC_DAPM_OUTPUT(= OUT)
widgets in below case.
It is assuming 2channel, S32_LE format for now.
It needs to be updated, too.
It needs "codec2codec" node (= B), needs to have routing (= A),
need to indicate CPU side at links (= X).
ports@0 is for CPU side (= X), port@1 is Codec side (= Y).
It needs to have "rate" (= C)
+--+
| |<-- Codec0 <-- IN
| |--> Codec1 --> OUT
+--+
sound {
compatible = "audio-graph-card2";
(A) routing = "OUT" ,"DAI1 Playback",
"DAI0 Capture", "IN";
(X) links = <&c2c>;
(B) codec2codec {
ports {
(C) rate = <48000>;
(X) c2c: port@0 { c2cf_ep: endpoint { remote-endpoint = <&codec0_ep>; }; };
(Y) port@1 { c2cb_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
};
};
Codec {
ports {
port@0 {
bitclock-master;
frame-master;
codec0_ep: endpoint { remote-endpoint = <&c2cf_ep>; }; };
port@1 { codec1_ep: endpoint { remote-endpoint = <&c2cb_ep>; }; };
};
};
Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y26ylu4a.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card2 will support DPCM/Multi/Codec2Codec,
and these will use almost same DT settings which uses
ports0 and ports1.
This patch adds asoc_graph_is_ports0() which checks
port is under port0 or not.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875yu2n8ra.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pierre-Louis Bossart
Brent Lu
Trevor Wu
Amadeusz Sławiński
Cezary Rojewski
Vincent Knecht
Arnd Bergmann
Mark Brown
Julian Braha
Richard Fitzgerald
Nathan Chancellor
AngeloGioacchino Del Regno
Yang Yingliang
George Song
Geert Uytterhoeven
Srinivas Kandagatla
Srinivasa Rao Mandadapu
David Lin
Stephan Gerhold
Uwe Kleine-König
Dmitry Osipenko
Ajit Kumar Pandey
Derek Fang
Jerome Brunet
Alexander Sverdlin
Sugar Zhang
Hans de Goede
Vijendar Mukunda
Nicolas Frattaroli
Christian Hewitt
Kuninori Morimoto
Greg Kroah-Hartman
Takashi Iwai