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>
Now UAPI of ALSA firewire stack got enough functions to interact with
userspace for protocol of MOTU FireWire series. Let's remove the relevant
TODO.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211029012847.11839-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A user reports functional regression for Mackie Onyx 1640i that the device
generates slow sound with ALSA oxfw driver which supports media clock
recovery. Although the device is based on OXFW971 ASIC, it does not
transfer isochronous packet with own event frequency as expected. The
device seems to adjust event frequency according to events in received
isochronous packets in the beginning of packet streaming. This is
unknown quirk.
This commit fixes the regression to turn the recovery off in driver
side. As a result, nominal frequency is used in duplex packet streaming
between device and driver. For stability of sampling rate in events of
transferred isochronous packet, 4,000 isochronous packets are skipped
in the beginning of packet streaming.
Reference: https://github.com/takaswie/snd-firewire-improve/issues/38
Fixes: 029ffc4294 ("ALSA: oxfw: perform sequence replay for media clock recovery")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211028130325.45772-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
In command DSP models, one meter information consists of 4 bytes for
IEEE 764 floating point (binary32). In previous patch, it is exported
to userspace as 32 bit storage since the storage is also handled in
ALSA firewire-motu driver as well in kernel space in which floating point
arithmetic is not preferable. On the other hand, ALSA firewire-motu driver
doesn't perform floating point calculation. The driver just gather meter
information from isochronous packets and fill structure fields for
userspace.
In 'header' target of Kbuild, UAPI headers are processed before installed.
In this timing, #ifdef macro with __KERNEL__ is removed. This mechanism
is useful in the case so that the 32 bit storage can be accessible as u32
type in kernel space and float type in user space. We can see the same
usage in ''struct acct_v3' in 'include/uapi/linux/acct.h'.
This commit is for the above idea. Additionally, due to message
protocol, meter information is filled with 0xffffffff in the end of
period but 0xffffffff is invalid as binary32. To avoid confusion in
userspace application, the last two elements are left without any
assignment.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211027125529.54295-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After further investigation, I realize that the total number of elements
in array is not enough to store all of related messages from device.
This commit refines meter array and message parser.
In terms of channel identifier, register DSP models are classified to
two categories:
1. the target of output is selectable
828mk2, 896hd, and Traveler are in the category. They transfer messages
with channel identifier between 0x00 and 0x13 for input meters,
therefore 20 elements are needed to store.
On the other hand, they transfer messages with channel identifier for one
pair of output meters. The selection is done by asynchronous write
transaction to offset 0x'ffff'f000'0b2c. The table for relationship
between written value and available identifiers is below:
============= ===============
written value identifier pair
============= ===============
0x00000b00 0x80/0x81
0x00000b01 0x82/0x83
... ...
0x00000b0b 0x96/0x97
... ...
0x00000b10 0xa0/0xa1
... ...
0x00000b3f 0xfe/0xff
... ...
greater 0xfe/0xff
============= ===============
Actually in the above three models, 0x96/0x97 pair is the maximum. Thus
the number of available output meter is 24.
2. all of output is available
8 pre, Ultralite, Audio Express, and 4 pre are in the category. They
transfer messages for output meters without any selection. The table for
available identifier for each direction is below:
============== ========= ==========
model input output
============== ========= ==========
8 pre 0x00-0x0f 0x82-0x8d
Ultralite 0x00-0x09 0x82-0x8f
Audio Express 0x00-0x09 0x80-0x8d
4 pre 0x00-0x09 0x80-0x8d
============== ========= ==========
Some of available identifiers might not be used for actual output meters.
Anyway, 24 plus 24 elements accommodate the input/output meters.
I note that isochronous packet from V3HD/V4HD deliver no message.
Notification by asynchronous transaction to registered address seems to be
used for the purpose as well as for change of mixer parameter.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211027125529.54295-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA firewire-motu driver recently got support for event notification via
ALSA HwDep interface for register DSP models. However, when polling ALSA
HwDep cdev, the driver can cause null pointer dereference for the other
models due to accessing to unallocated memory or uninitialized memory.
This commit fixes the bug by check the type of model before accessing to
the memory.
Reported-by: kernel test robot <lkp@intel.com>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Fixes: 634ec0b290 ("ALSA: firewire-motu: notify event for parameter change in register DSP model")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211027125529.54295-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Add the missing endpoint max-packet sanity check to probe() to avoid
division by zero in alloc_stream_buffers() in case a malicious device
has broken descriptors (or when doing descriptor fuzz testing).
Note that USB core will reject URBs submitted for endpoints with zero
wMaxPacketSize but that drivers doing packet-size calculations still
need to handle this (cf. commit 2548288b4f ("USB: Fix: Don't skip
endpoint descriptors with maxpacket=0")).
Fixes: 63978ab3e3 ("sound: add Edirol UA-101 support")
Cc: stable@vger.kernel.org # 2.6.34
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211026095401.26522-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
USB control and bulk message timeouts are specified in milliseconds and
should specifically not vary with CONFIG_HZ.
Fixes: c6d43ba816 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB")
Cc: stable@vger.kernel.org # 2.6.39
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211025121142.6531-2-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pointer cs_desc return from snd_usb_find_clock_source could
be null, so there is a potential null pointer dereference issue.
Fix this by adding a null check before dereference.
Signed-off-by: Chengfeng Ye <cyeaa@connect.ust.hk>
Link: https://lore.kernel.org/r/20211024111736.11342-1-cyeaa@connect.ust.hk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pointer block return from snd_gf1_dma_next_block could be
null, so there is a potential null pointer dereference issue.
Fix this by adding a null check before dereference.
Signed-off-by: Chengfeng Ye <cyeaa@connect.ust.hk>
Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
On the 'HP Spectre x360 Convertible 14-ea0xx' the microphone mute led is
controlled by GPIO 0x04. The speaker mute LED does not seem to be
exposed by GPIO and is there not set.
[ a slight coding-style fix by tiwai ]
Fixes: c3bb2b5219 ("ALSA: hda/realtek: Quirk for HP Spectre x360 14 amp setup")
Signed-off-by: Johnathon Clark <john.clark@cantab.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211020131253.35894-1-john.clark@cantab.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
When a Jieli Technology USB Webcam is connected, the video part works
well, but the mic sound is speeded up. On dmesg there are messages
about different rates from the runtime rates, warnings about volume
resolution and lastly, the log is filled, every 5 seconds, with
retire_capture_urb error messages.
The mic works only when ep packet size is set to wMaxPacketSize (normal
sound and no more retire_capture_urb error messages). Skipping reading
sample rate, fixes the messages about different rates and forcing a volume
resolution, fixes warnings about volume range. I have arbitrarily choosed
the value (16): I read in a comment that there should be no more than 255
levels, so 4096 (max volume) / 16 = 0-255.
Signed-off-by: Marco Giunta <giun7a@gmail.com>
Link: https://lore.kernel.org/r/20211018162552.12082-1-giun7a@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
We've had an x86-specific SG-buffer handling code, but now it can be
merged gracefully with the standard non-contiguous DMA pages.
After the migration, SNDRV_DMA_TYPE_DMA_SG becomes identical with
SNDRV_DMA_TYPE_NONCONTIG on x86, while others still fall back to
SNDRV_DMA_TYPE_DEV.
The remaining problem is about the SG-buffer with WC pages: the DMA
core stuff on x86 doesn't treat it well, so we still need some special
handling to manipulate the page attribute manually. The mmap handler
for SNDRV_DMA_TYPE_DEV_SG_WC still returns -ENOENT intentionally for
the fallback to the default handler.
Link: https://lore.kernel.org/r/20211017074859.24112-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Following to the addition of non-contiguous pages, this patch adds the
new contiguous non-coherent page allocation to the standard memalloc
helper. Like the previous non-contig type, this non-coherent type is
also directional and requires the explicit sync, too. Hence the
driver using this type of buffer may need to set
SNDRV_PCM_INFO_EXPLICIT_SYNC flag to the PCM hardware.info as well,
unless it's set up in the managed mode.
Link: https://lore.kernel.org/r/20211017074859.24112-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support for allocation of non-contiguous DMA pages
in the common memalloc helper. It's another SG-buffer type, but
unlike the existing one, this is directional and requires the explicit
sync / invalidation of dirty pages on non-coherent architectures.
For this enhancement, the following points are changed:
- snd_dma_device stores the DMA direction.
- snd_dma_device stores need_sync flag indicating whether the explicit
sync is required or not.
- A new variant of helper functions, snd_dma_alloc_dir_pages() and
*_all() are introduced; the old snd_dma_alloc_pages() and *_all()
kept as just wrappers with DMA_BIDIRECTIONAL.
- A new helper snd_dma_buffer_sync() is introduced; this gets called
in the appropriate places.
- A new allocation type, SNDRV_DMA_TYPE_NONCONTIG, is introduced.
When the driver allocates pages with this new type, and it may require
the SNDRV_PCM_INFO_EXPLICIT_SYNC flag set to the PCM hardware.info for
taking the full control of PCM applptr and hwptr changes (that implies
disabling the mmap of control/status data). When the buffer
allocation is managed by snd_pcm_set_managed_buffer(), this flag is
automatically set depending on the result of dma_need_sync()
internally. Otherwise, if the buffer is managed manually, the driver
has to set the flag explicitly, too.
The explicit sync between CPU and device for non-coherent memory is
performed at the points before and after read/write transfer as well
as the applptr/hwptr syncptr ioctl. In the case of mmap mode,
user-space is supposed to call the syncptr ioctl with the hwptr flag
to update and fetch the status at first; this corresponds to CPU-sync.
Then user-space advances the applptr via syncptr ioctl again with
applptr flag, and this corresponds to the device sync with flushing.
Other than the DMA direction and the explicit sync, the usage of this
new buffer type is almost equivalent with the existing
SNDRV_DMA_TYPE_DEV_SG; you can get the page and the address via
snd_sgbuf_get_page() and snd_sgbuf_get_addr(), also calculate the
continuous pages via snd_sgbuf_get_chunk_size().
For those SG-page handling, the non-contig type shares the same ops
with the vmalloc handler. As we do always vmap the SG pages at first,
the actual address can be deduced from the vmapped address easily
without iterating the SG-list.
Link: https://lore.kernel.org/r/20211017074859.24112-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On m68k, compiling drivers under SND_ISA causes build errors:
../sound/core/isadma.c: In function 'snd_dma_program':
../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration]
33 | flags = claim_dma_lock();
| ^~~~~~~~~~~~~~
../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration]
41 | release_dma_lock(flags);
| ^~~~~~~~~~~~~~~~
../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare':
../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function)
253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT);
| ^~~~~~~~~~~~
../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in
../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare':
../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function)
322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT);
| ^~~~~~~~~~~~
and more...
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-m68k@lists.linux-m68k.org
Cc: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
This commit copies queued event for change of register DSP into
userspace when application operates ALSA hwdep character device.
The notification occurs only when packet streaming is running.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-12-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to notify parameter change of register DSP
to userspace application. A simple queue is added to store encoded data
for the change as long as ALSA hwdep character device is opened by
application.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-11-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds new ioctl command for userspace applications to read
cached parameters of register DSP.
The structured data includes model-dependent parameters. Userspace
application should be carefully programmed so that what parameter is
common and specific.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-10-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit parses message and cache current parameters of input function,
available for MOTU Ultralite, 4 pre, and Audio Express.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-9-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit parses message and cache current parameters of line input
function, available for MOTU 828 mk2 and Traveler.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-8-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit parses message and cache current parameters of output
function, commonly available for all of register DSP model.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit parses message and cache current parameters of mixer output
function, commonly available for all of register DSP model
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In register DSP models, current parameters of DSP are always reported by
messages in isochronous packet. When user operates hardware component such
as rotary knob, corresponding message is changed.
This commit parses the message and cache current parameters of mixer
source function, commonly available for all of register DSP models.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds new ioctl commands for userspace applications to read
cached image about hardware meters in register DSP and command DSP models.
The content of image differs depending on models. Model-specific parser
should be implemented in userspace.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of MOTU models allows software to configure their DSP parameters by
command included in asynchronous transaction. The models multiplex messages
for hardware meters into isochronous packet as well as PCM frames. For
convenience, I call them as 'command DSP' model.
This patch adds message parser for them to gather hardware meter
information.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of MOTU models allows software to configure their DSP parameters by
accessing to their registers. The models multiplex messages for status of
DSP into isochronous packet as well as PCM frames. The message includes
information of hardware metering, MIDI message, current parameters of DSP.
For my convenience, I call them as 'register DSP' model.
This patch adds message parser for them to gather hardware meter
information.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20211015080826.34847-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
We already have dummy-codec, dummy-platform.
But its issues are
1) we don't have dummy-cpu,
2) we can't select it via DeviceTree
3) It do nothing
Sometimes we want to have Dummy Sound Component for debugging,
for testing, for learning Framework behavior, etc, etc...
This patch adds Test-Component driver for it.
User can select CPU Component by using "test-cpu" compatible,
and can select Codec Component by using "test-codec" compatible.
It doesn't support Platform so far, but is easy to add.
We can verbose print to know its progress if user selected
xxx-verbose compatible driver.
for example,
test-cpu : silent Component, silent DAI
test-cpu-verbose-component : verbose Component, silent DAI
test-cpu-verbose-dai : silent Component, verbose DAI
test-cpu-verbose : verbose Component, verbose DAI
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877dein8rx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and
rear speakers are silent, this patch fixes that by overriding the pin
layout and by initializing the amplifier which needs a GPIO pin to be
set to 1 then 0, similar to the existing HP Spectre x360 14 model.
In order to have volume control, both front and rear speakers were
forced to use the DAC1.
This patch also correctly map the mute LED but since there is no
microphone on/off switch exposed by the alsa subsystem it never turns
on by itself.
There are still known audio issues in this laptop: headset microphone
doesn't work, the button to mute/unmute microphone is not yet mapped,
the LED of the mute/unmute speakers doesn't seems to be exposed via
GPIO and never turns on.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213953
Signed-off-by: Davide Baldo <davide@baldo.me>
Link: https://lore.kernel.org/r/20211015072121.5287-1-davide@baldo.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As per discussion at: https://github.com/szszoke/sennheiser-gsp670-pulseaudio-profile/issues/13
The GSP670 has 2 playback and 1 recording device that by default are
detected in an incompatible order for alsa. This may have been done to make
it compatible for the console by the manufacturer and only affects the
latest firmware which uses its own ID.
This quirk will resolve this by reordering the channels.
Signed-off-by: Brendan Grieve <brendan@grieve.com.au>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211015025335.196592-1-brendan@grieve.com.au
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both snd_pcm_delay() and snd_pcm_hwsync() do the almost same thing.
The only difference is that the former calculate the delay, so unify
them as a code cleanup, and treat NULL delay argument only for hwsync
operation.
Also, the patch does a slight code refactoring in snd_pcm_delay().
The initialization of the delay value is done in the caller side now.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211014145323.26506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we used to read the current value of the mixer element
dynamically at the first access, and the error from a GET_CUR message
is treated as a fatal error (unless QUIRK_IGNORE_CTL_ERROR is set).
It's rather inconvenient, as most of GET_CUR errors are no fatal, and
we can continue operation with assumption of some fixed value.
This patch makes the USB-audio driver to change the behavior at probe
time; now it tries to initialize the current value of each mixer
element that is built from a feature unit (those for typically for
mixer volumes and switches). When a read failure happens, it tries to
set the known minimum value. After that point, a cached value is used
always, hence we won't hit GET_CUR message error any longer.
The error from GET_CUR message is still shown as a warning normally,
but only once at the probe time, and it'll keep operating. If the
message is confirmed to be harmless, it can be shut up by
QUIRK_IGNORE_CTL_ERROR quirk flag, too.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error from snd_usb_lock_shutdown() indicates that the device is
disconnected, hence it makes no sense to show any further control
message error in get_ctl_value_v2(). Return the error directly
instead.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error message in get_ctl_value_v2() (for UAC2/3) is shown via
KERN_ERR level but it was intended to be rather a debug message as
seen in get_ctl_value_v1() (for UAC1). This patch downgrade the
printk level.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A back-merge of 5.15 branch into 5.16-devel branch for further
development of USB and ALSA core stuff that depends on 5.15 fixes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply existing PCI quirk to the Clevo PC50HS and related models to fix
audio output on the built in speakers.
Signed-off-by: Steven Clarkson <sc@lambdal.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211014133554.1326741-1-sc@lambdal.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Shciit Hel device responds to the ctl message for the mic capture
switch with a timeout of -EPIPE:
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
This seems safe to ignore as the device works properly with the control
message quirk, so add it to the quirk table so all is good.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-usb@vger.kernel.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/YWgR3nOI1osvr5Yo@kroah.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A little pop can be heard obviously from HP while playing a silent.
This patch fixes it by using two functions:
1. Enable HP 1bit output mode.
2. Change the charge pump switch size during playback on and off.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/20211014094054.811-1-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There are several things the patch adding the support for 'I2S Reference'
got wrong:
- "None" selection is in fact equals to last selected reference
- The custom put overrides RX/TX len, TDM slot sizes, etc
- the enum is useless in most part for the reference tracking
- there is no need for EXT control as there is a single bit in
RT1011_TDM1_SET_1 register (bit 7) which selects the reference
- it was using ucontrol->value.integer.value[0] in the put/get callbacks
which causesed access to 'I2S Reference' enum with alsamixer to fail
Complements: c3de683c4d ("ASoC: rt1011: Fix 'I2S Reference' enum control caused error")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20211013123300.11095-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a noise issue for 8kHz sample rate on slave mode.
Compared with master mode, the difference is the DACDIV
setting, after correcting the DACDIV, the noise is gone.
There is no noise issue for 48kHz sample rate, because
the default value of DACDIV is correct for 48kHz.
So wm8960_configure_clocking() should be functional for
ADC and DAC function even if it is slave mode.
In order to be compatible for old use case, just add
condition for checking that sysclk is zero with
slave mode.
Fixes: 0e50b51aa2 ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1634102224-3922-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ChiYuan Huang <cy_huang@richtek.com>:
From: ChiYuan Huang <cy_huang@richtek.com>
This patch series Add the Richtek RT9120 support.
In v4:
- Add 'classd_tlv' for 'SPK Gain Volume' control item.
- Unify the tlv declaration to the postfix '_tlv'.
- Fix 'digital_tlv' mute as 1 to declare the minimum is muted.
In v3:
- Add dvdd regulator binding to check the dvdd voltage domain.
- Refine sdo_select_text.
- Use switch case in 'internal_power_event' function.
- Remove the volume and mute initially write in component probe.
- Remove the mute API. It's no need by HW design.
In v2:
- Add missing #sound-dai-cells property.
ChiYuan Huang (2):
ASoC: dt-bindings: rt9120: Add initial bindings
ASoC: rt9120: Add rt9210 audio amplifier support
.../devicetree/bindings/sound/richtek,rt9120.yaml | 59 +++
sound/soc/codecs/Kconfig | 10 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/rt9120.c | 495 +++++++++++++++++++++
4 files changed, 566 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml
create mode 100644 sound/soc/codecs/rt9120.c
--
2.7.4
The only usage of acp5x_i2s_dai_ops is to assign its address to the ops
field in the snd_soc_dai_driver struct, which is a pointer to const.
Make it const to allow the compiler to put it in read-only memory.
Signed-off-by: Rikard Falkeborn <rikard.falkeborn@gmail.com>
Link: https://lore.kernel.org/r/20211012211506.21159-1-rikard.falkeborn@gmail.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.
Signed-off-by: Rikard Falkeborn <rikard.falkeborn@gmail.com>
Link: https://lore.kernel.org/r/20211012205521.14098-1-rikard.falkeborn@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This drops the rt9210 support due to a race with a new version being
sent out for some incremental changes.
Signed-off-by: Mark Brown <broonie@kernel.org>
The device advertises 8 formats, but only a rate of 48kHz is honored
by the hardware and 24 bits give chopped audio, so only report the
one working combination. This fixes out-of-the-box audio experience
with PipeWire which otherwise attempts to choose S24_3LE (while
PulseAudio defaulted to S16_LE).
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211012200906.3492-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_hdac_bus_reset_link() contains logic to clear STATESTS register
before performing controller reset. This code dates back to an old
bugfix in commit e8a7f136f5 ("[ALSA] hda-intel - Improve HD-audio
codec probing robustness"). Originally the code was added to
azx_reset().
The code was moved around in commit a41d122449 ("ALSA: hda - Embed bus
into controller object") and ended up to snd_hdac_bus_reset_link() and
called primarily via snd_hdac_bus_init_chip().
The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is
called when controller is not in reset. In this case, STATESTS can be
cleared. This can be useful e.g. when forcing a controller reset to retry
codec probe. A normal non-power-on reset will not clear the bits.
However, this old logic is problematic when controller is already in
reset. The HDA specification states that controller must be taken out of
reset before writing to registers other than GCTL.CRST (1.0a spec,
3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost
if the controller is already in reset per the HDA specification mentioned.
This has been harmless on older hardware. On newer generation of Intel
PCIe based HDA controllers, if configured to report issues, this write
will emit an unsupported request error. If ACPI Platform Error Interface
(APEI) is enabled in kernel, this will end up to kernel log.
Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if
the function is called when controller is not in reset. Otherwise
clearing the bits is not possible and should be skipped.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent support for the improved low-latency playback mode applied
the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this
was a slight overkill. The use of the flag above disables effectively
both PCM status and control mmaps, while basically what we want to
track is only about the appl_ptr update.
For less restriction, use a more proper flag,
SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control
mmap.
Fixes: d5f871f89e ("ALSA: usb-audio: Improved lowlatency playback support")
Link: https://lore.kernel.org/r/20211011103650.10182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to define the codec pin 0x1b to be the mic, but somehow
the mic doesn't support hot plugging detection, and Windows also has
this issue, so we set it to phantom headset-mic.
Also the determine_headset_type() often returns the omtp type by a
mistake when we plug a ctia headset, this makes the mic can't record
sound at all. Because most of the headset are ctia type nowadays and
some machines have the fixed ctia type audio jack, it is possible this
machine has the fixed ctia jack too. Here we set this mic jack to
fixed ctia type, this could avoid the mic type detection mistake and
make the ctia headset work stable.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214537
Reported-and-tested-by: msd <msd.mmq@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20211012114748.5238-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Access to 'I2S Reference' enum causes alsamixer to fail to load:
$ alsamixer
cannot load mixer controls: Invalid argument
cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow
The reason is that the original patch adding the code was using
ucontrol->value.integer.value[0]
instead the correct
ucontrol->value.enumerated.item[0]
for an ENUM control.
Fixes: 87f40af26c ("ASoC: rt1011: add i2s reference control for rt1011")
Reported-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20211011144518.2518-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Ensure the default 0dB playback path is always used.
The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was
spurious, and resulted in a -6dB attenuation being accidentally
inserted into the playback path.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211011144903.28915-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The small set of cleanups against bytcr_rt5651 board file.
In v2:
- added commit message to patch 2 (Joe, Pierre)
- added cover letter (Pierre)
- added Hans to Cc list (Hans)
Andy Shevchenko (4):
ASoC: Intel: bytcr_rt5651: Get platform data via dev_get_platdata()
ASoC: Intel: bytcr_rt5651: Use temporary variable for struct device
ASoC: Intel: bytcr_rt5651: use devm_clk_get_optional() for mclk
ASoC: Intel: bytcr_rt5651: Utilize dev_err_probe() to avoid log
saturation
sound/soc/intel/boards/bytcr_rt5651.c | 118 +++++++++++---------------
1 file changed, 50 insertions(+), 68 deletions(-)
--
2.33.0
Michael Forney reported an incorrect padding type that was defined in
the commit 80fe7430c7 ("ALSA: add new 32-bit layout for
snd_pcm_mmap_status/control") for PCM control mmap data.
His analysis is correct, and this caused the misplacements of PCM
control data on 32bit arch and 32bit compat mode.
The bug is that the __pad2 definition in __snd_pcm_mmap_control64
struct was wrongly with __pad_before_uframe, which should have been
__pad_after_uframe instead. This struct is used in SYNC_PTR ioctl and
control mmap. Basically this bug leads to two problems:
- The offset of avail_min field becomes wrong, it's placed right after
appl_ptr without padding on little-endian
- When appl_ptr and avail_min are read as 64bit values in kernel side,
the values become either zero or corrupted (mixed up)
One good news is that, because both user-space and kernel
misunderstand the wrong offset, at least, 32bit application running on
32bit kernel works as is. Also, 64bit applications are unaffected
because the padding size is zero. The remaining problem is the 32bit
compat mode; as mentioned in the above, avail_min is placed right
after appl_ptr on little-endian archs, 64bit kernel reads bogus values
for appl_ptr updates, which may lead to streaming bugs like jumping,
XRUN or whatever unexpected.
(However, we haven't heard any serious bug reports due to this over
years, so practically seen, it's fairly safe to assume that the impact
by this bug is limited.)
Ideally speaking, we should correct the wrong mmap status control
definition. But this would cause again incompatibility with the
existing binaries, and fixing it (e.g. by renumbering ioctls) would be
really messy.
So, as of this patch, we only correct the behavior of 32bit compat
mode and keep the rest as is. Namely, the SYNC_PTR ioctl is now
handled differently in compat mode to read/write the 32bit values at
the right offsets. The control mmap of 32bit apps on 64bit kernels
has been already disabled (which is likely rather an overlook, but
this worked fine at this time :), so covering SYNC_PTR ioctl should
suffice as a fallback.
Fixes: 80fe7430c7 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control")
Reported-by: Michael Forney <mforney@mforney.org>
Reviewed-by: Arnd Bergmann <arnd@arndb.de>
Cc: <stable@vger.kernel.org>
Cc: Rich Felker <dalias@libc.org>
Link: https://lore.kernel.org/r/29QBMJU8DE71E.2YZSH8IHT5HMH@mforney.org
Link: https://lore.kernel.org/r/20211010075546.23220-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev_err_probe() avoids printing into log when the deferred probe is invoked.
This is possible when clock provider is pending to appear.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007170250.27997-5-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The devm_clk_get_optional() helper returns NULL when devm_clk_get()
returns -ENOENT. This makes things slightly cleaner. The added benefit
is mostly cosmetic.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007170250.27997-4-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dev_err_probe() avoids printing into log when the deferred probe is invoked.
This is possible when clock provider is pending to appear.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007165715.27463-5-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The devm_clk_get_optional() helper returns NULL when devm_clk_get()
returns -ENOENT. This makes things slightly cleaner. The added benefit
is mostly cosmetic.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007165715.27463-4-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use temporary variable for struct device to make code neater.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007165715.27463-3-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Access to platform data via dev_get_platdata() getter to make code cleaner.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007165715.27463-2-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We are using fch clock controller as parent mclk source for rt5682
codec. Add config to enable clock framework support for 48MHz fixed
clock when machine driver config is selected.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20211011055354.67719-1-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
'component' is allocated in snd_soc_register_component(), but component->list
is not initalized, this may cause snd_soc_del_component_unlocked() deref null
ptr in the error handing case.
KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007]
RIP: 0010:__list_del_entry_valid+0x81/0xf0
Call Trace:
snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core]
snd_soc_add_component.cold+0x54/0x6c [snd_soc_core]
snd_soc_register_component+0x70/0x90 [snd_soc_core]
devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core]
tas2552_probe+0x265/0x320 [snd_soc_tas2552]
? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552]
i2c_device_probe+0xa31/0xbe0
Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize().
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Yang Yingliang <yangyingliang@huawei.com>
Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
