Jiao Zhou [Tue, 6 Dec 2022 18:53:11 +0000 (13:53 -0500)]
ALSA: hda/hdmi: Add HP Device 0x8711 to force connect list
HDMI audio is not working on the HP EliteDesk 800 G6 because the pin is
unconnected. This issue can be resolved by using the 'hdajackretask'
tool to override the unconnected pin to force it to connect.
Takashi Iwai [Tue, 6 Dec 2022 10:13:26 +0000 (11:13 +0100)]
Merge tag 'asoc-v6.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.2
This is a fairly sedate release for the core code, but there's been a
lot of driver work especially around the x86 platforms and device tree
updates:
- More cleanups of the DAPM code from Morimoto-san.
- Factoring out of mapping hw_params onto SoundWire configuration by
Charles Keepax.
- The ever ongoing overhauls of the Intel DSP code continue, including
support for loading libraries and probes with IPC4 on SOF.
- Support for more sample formats on JZ4740.
- Lots of device tree conversions and fixups.
- Support for Allwinner D1, a range of AMD and Intel systems, Mediatek
systems with multiple DMICs, Nuvoton NAU8318, NXP fsl_rpmsg and
i.MX93, Qualcomm AudioReach Enable, MFC and SAL, RealTek RT1318 and
Rockchip RK3588
There's more cross tree updates than usual, though all fairly minor:
- Some OMAP board file updates that were depedencies for removing their
providers in ASoC, as part of a wider effort removing the support for
the relevant OMAP platforms.
- A new I2C API required for updates to the new I2C probe API.
- A DRM update making use of a new API for fixing the capabilities
advertised via hdmi-codec.
Since this is being sent early I might send some more stuff if you've
not yet sent your pull request and there's more come in.
The Dell Inspiron Plus 16, in both laptop and 2in1 form factor, has top
speakers connected on NID 0x17, which the codec reports as unconnected.
These speakers should be connected to the DAC on NID 0x03.
ASoC: dt-bindings: maxim,max98504: Convert to DT schema
Convert the Maxim Integrated MAX98504 amplifier bindings to DT schema.
Few properties are made optional:
1. interrupts: current Linux driver implementation does not use them,
2. supplies: on some boards these might be wired to battery, for which
no regulator is provided.
ASoC: dt-bindings: maxim,max98357a: Convert to DT schema
Convert the Maxim Integrated MAX98357A/MAX98360A amplifier bindings to
DT schema. Add missing properties ('#sound-dai-cells' and
'sound-name-prefix' from common DAI properties).
ASoC: dt-bindings: Reference common DAI properties
Reference in all sound components which have '#sound-dai-cells' the
dai-common.yaml schema, which allows to use 'sound-name-prefix'
property.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Tested-by: Nicolas Frattaroli <frattaroli.nicolas@gmail.com> Acked-by: Nicolas Frattaroli <frattaroli.nicolas@gmail.com> Link: https://lore.kernel.org/r/20221203160442.69594-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: dt-bindings: Extend name-prefix.yaml into common DAI properties
Rename name-prefix.yaml into common DAI schema and document
'#sound-dai-cells' for completeness. The '#sound-dai-cells' cannot be
really constrained, as there are users with value of 0, 1 and 2, but at
least it brings definition to one common place.
Colin Ian King [Fri, 2 Dec 2022 17:14:50 +0000 (17:14 +0000)]
ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
Don't populate the read-only arrays capture_reg_H and capture_reg_L
on the stack but instead make them static const. Also makes the
object code a little smaller.
Colin Ian King [Fri, 2 Dec 2022 16:41:56 +0000 (16:41 +0000)]
ASoC: uniphier: aio-core: Make some read-only arrays static const
Don't populate the read-only arrays slotsel_2ch, slotsel_multi, v_pll
and v_div on the stack but instead make them static const. Also makes
the object code a little smaller.
Ajye Huang [Mon, 5 Dec 2022 12:06:48 +0000 (17:36 +0530)]
ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER
The Skyrim project and Whiterun met error when DSP
loading during device boot.
Ex, error in kernel log,
ERR kernel: [ 16.124537] snd_sof_amd_rembrandt
0000:04:00.5: PSP validation failed.
Use the snd_sof_dsp_read_poll_timeout function to successfully
read the FW_QUALIFIER register
Takashi Iwai [Mon, 5 Dec 2022 13:21:24 +0000 (14:21 +0100)]
ALSA: usb-audio: Workaround for XRUN at prepare
Under certain situations (typically in the implicit feedback mode),
USB-audio driver starts a playback stream already at PCM prepare call
even before the actual PCM trigger-START call. For implicit feedback
mode, this effectively starts two streams for data and sync
endpoints, and if a coupled sync stream gets XRUN at this point, it
results in an error -EPIPE.
The problem is that currently we return -EPIPE error as is from the
prepare. Then application tries to recover again via the prepare
call, but it'll fail again because the sync-stop is missing. The
sync-stop is missing because it's an internal trigger call (hence the
PCM core isn't involved).
Since we'll need to re-issue the prepare in anyway when trapped into
this pitfall, this patch attempts to address it in a bit different
way; namely, the driver tries to prepare once again after syncing the
stop manually by itself -- so applications don't see the internal
error. At the second failure, we report the error as is, but this
shouldn't happen in normal situations.
Takashi Iwai [Mon, 5 Dec 2022 13:21:23 +0000 (14:21 +0100)]
ALSA: pcm: Handle XRUN at trigger START
When the driver returns -EPIPE for indicating an XRUN already at PCM
trigger START, we should treat properly and set it to the XRUN state.
Otherwise the state is missing and the application would try to issue
trigger again without knowing that it's in an error state.
This is just for a theoretical bug, and it won't happen in most
cases.
Takashi Iwai [Mon, 5 Dec 2022 13:21:22 +0000 (14:21 +0100)]
ALSA: pcm: Set missing stop_operating flag at undoing trigger start
When a PCM trigger-start fails at snd_pcm_do_start(), PCM core tries
to undo the action at snd_pcm_undo_start() by issuing the trigger STOP
manually. At that point, we forgot to set the stop_operating flag,
hence the sync-stop won't be issued at the next prepare or other
calls.
This patch adds the missing stop_operating flag at
snd_pcm_undo_start().
Mark Brown [Sun, 4 Dec 2022 17:01:50 +0000 (17:01 +0000)]
ASoC/tda998x: Fix reporting of nonexistent capture streams
Merge series from Mark Brown <broonie@kernel.org>:
The recently added pcm-test selftest has pointed out that systems with
the tda998x driver end up advertising that they support capture when in
reality as far as I can see the tda998x devices are transmit only. The
DAIs registered through hdmi-codec are bidirectional, meaning that for
I2S systems when combined with a typical bidrectional CPU DAI the
overall capability of the PCM is bidirectional. In most cases the I2S
links will clock OK but no useful audio will be returned which isn't so
bad but we should still not advertise the useless capability, and some
systems may notice problems for example due to pinmux management.
This is happening due to the hdmi-codec helpers not providing any
mechanism for indicating unidirectional audio so add one and use it in
the tda998x driver. It is likely other hdmi-codec users are also
affected but I don't have those systems to hand.
Mark Brown (2):
ASoC: hdmi-codec: Allow playback and capture to be disabled
drm: tda99x: Don't advertise non-existent capture support
Mark Brown [Wed, 30 Nov 2022 18:46:43 +0000 (18:46 +0000)]
ASoC: hdmi-codec: Allow playback and capture to be disabled
Currently the hdmi-codec driver always registers both playback and capture
capabilities but for most systems there's no actual capture capability,
usually HDMI is transmit only. Provide platform data which allows the users
to indicate what is supported so that we don't end up advertising things
to userspace that we can't actually support.
In order to avoid breaking existing users the flags in platform data are
a bit awkward and specify what should be disabled rather than doing the
perhaps more expected thing and defaulting to not supporting capture.
Mark Brown [Thu, 1 Dec 2022 17:07:45 +0000 (17:07 +0000)]
kselftest/alsa: Add more coverage of sample rates and channel counts
Now that we can skip unsupported configurations add some more test cases
using that, cover 8kHz, 44.1kHz and 96kHz plus 8kHz mono and 48kHz 6
channel.
44.1kHz is a different clock base to the existing 48kHz tests and may
therefore show problems with the clock configuration if only 8kHz based
rates are really available (or help diagnose if bad clocking is due to
only 44.1kHz based rates being supported). 8kHz mono and 48Hz 6 channel
are real world formats and should show if clocking does not account for
channel count properly.
Mark Brown [Thu, 1 Dec 2022 17:07:44 +0000 (17:07 +0000)]
kselftest/alsa: Provide more meaningful names for tests
Rather than just numbering the tests try to provide semi descriptive names
for what the tests are trying to cover. This also has the advantage of
meaning we can add more tests without having to keep the list of tests
ordered by existing number which should make it easier to understand what
we're testing and why.
Mark Brown [Thu, 1 Dec 2022 17:07:43 +0000 (17:07 +0000)]
kselftest/alsa: Don't any configuration in the sample config
The values in the one example configuration file we currently have are the
default values for the two tests we have so there's no need to actually set
them. Comment them out as examples, with a rename for the tests so that we
can update the tests in the code more easily.
Mark Brown [Thu, 1 Dec 2022 17:07:42 +0000 (17:07 +0000)]
kselftest/alsa: Report failures to set the requested channels as skips
If constraint selection gives us a number of channels other than the one
that we asked for that isn't a failure, that is the device implementing
constraints and advertising that it can't support whatever we asked
for. Report such cases as a test skip rather than failure so we don't have
false positives.
Mark Brown [Thu, 1 Dec 2022 17:07:41 +0000 (17:07 +0000)]
kselftest/alsa: Report failures to set the requested sample rate as skips
If constraint selection gives us a sample rate other than the one that we
asked for that isn't a failure, that is the device implementing sample
rate constraints and advertising that it can't support whatever we asked
for. Report such cases as a test skip rather than failure so we don't have
false positives.
Mark Brown [Thu, 1 Dec 2022 17:07:40 +0000 (17:07 +0000)]
kselftest/alsa: Refactor pcm-test to list the tests to run in a struct
In order to help make the list of tests a bit easier to maintain refactor
things so we pass the tests around as a struct with the parameters in,
enabling us to add new tests by adding to a table with comments saying
what each of the number are. We could also use named initializers if we get
more parameters.
David Rau [Mon, 21 Nov 2022 05:07:44 +0000 (05:07 +0000)]
ASoC: da7219: Fix pole orientation detection on OMTP headsets when playing music
The OMTP pin define headsets can be mis-detected as line out
instead of OMTP, causing obvious issues with audio quality.
This patch is to put increased resistances within
the device at a suitable point.
To solve this issue better, the new mechanism setup
ground switches with conditional delay control
and these allow for more stabile detection process
to operate as intended. This conditional delay control
will not impact the hardware process
but use extra system resource.
This commit improves control of ground switches in the AAD logic.
Takashi Sakamoto [Wed, 30 Nov 2022 14:33:13 +0000 (23:33 +0900)]
ALSA: dice: add support for Focusrite Saffire Pro 40 with TCD3070 ASIC
TC Applied Technologies (TCAT) produces TCD3070 as final DICE ASIC for
communication in IEEE 1394 bus for IEC 61883-1/6 protocol. As long as I
know, latter model of Focusrite Saffire Pro 40 is an application of the
ASIC and only in the market for consumers.
This patchset adds support for the device. The device has several
remarkable points.
1. No support for extended synchronization information section in TCAT
general protocol. The value of GLOBAL_EXTENDED_STATUS register is always
zero. Additionally, NOTIFY_EXT_STATUS message is never emitted.
2. No support for TCAT protocol extension. Hard coding is required for
format of CIP payload.
3. During several seconds after changing sampling rate, the block to
process PCM frames is under disfunction. When starting packet streaming
during the state, the block is never function till configuring different
sampling rate and several seconds.
This commit adds support for the device. The item 1 and 2 can be
adaptable, while item 3 is not. It's not preferable that user process
is forced to sleep during the disfunction in the call of ioctl(2) with
SNDRV_PCM_IOCTL_HW_PARAMS or SNDRV_PCM_IOCTL_PREPARE request. It's
inconvenient but let user configure preferable sampling rate in advance
of starting PCM substream.
The content of configuration ROM in the device I used is available at:
* https://github.com/takaswie/am-config-roms/
I note that any mixer control operation is implemented by unique
transaction. The frame of request consists of 16 bytes header followed
by payload.
header (4 quadlets):
1st: the type of request, prefixed with 0x8000
2nd: counter at 2 bytes in MSB side, the length of data at 2 bytes in LSB
side
3rd: parameter 0
4th: parameter 1
payload (variable length if need):
5th-: data according to parameters
The request frame is sent by block write request to 0x'ffff'e040'01c0.
The frame of response is similar to the frame of request, but it is
header only, thus fixed to 16 bytes. The response frame is sent to the
address which is registered by lock transaction to 0x'ffff'e040'0008.
If the operation results in batch of data, the 2nd quadlet of header
includes the length of data like request. The data is itself readable
by read block request to 0x'ffff'e040'0030, which includes both
header and payload for data, thus the length to read should be the
length of data plus 16 bytes for header
The actual value of request, parameter 0, parameter 1, and data is
unclear yet.
Artem Lukyanov [Wed, 30 Nov 2022 08:52:47 +0000 (11:52 +0300)]
ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022 into DMI table
This model requires an additional detection quirk to enable the
internal microphone - BIOS doesn't seem to support AcpDmicConnected
(nothing in acpidump output).
Mark Brown [Tue, 29 Nov 2022 19:29:05 +0000 (19:29 +0000)]
ASoC: Intel: avs: rt5682: Refactor jack handling
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Leftover from recent series [1].
Following changes are proposed for the rt5682 sound card driver:
1) Move jack unassignment from platform_device->remove() to
dai_link->exit(). This is done to make jack init and deinit flows
symmetric
2) Remove platform_device->remove() function
3) Simplify card->suspend_pre() and card->resume_post() by making use of
snd_soc_card_get_codec_dai() helper
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
ASoC: pcm512x: Fix PM disable depth imbalance in pcm512x_probe
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context. We fix it by going to
err_pm instead of err_clk.
Fixes:f086ba9d5389c ("ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL")
Mark Brown [Tue, 29 Nov 2022 16:56:44 +0000 (16:56 +0000)]
ASoC: Intel: avs: Refactor jack handling
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
For all the boards included in this patchset, a similar set of changes
is proposed:
1) Move jack unassignment from platform_device->remove() to
dai_link->exit(). This is done to make jack init and deinit flows
symmetric
2) Remove platform_device->remove() function
3) Simplify card->suspend_pre() and card->resume_post() by making use of
snd_soc_card_get_codec_dai() helper
While bdw_rt286 board - which is utilized by the catpt-driver - is
definitely not part of "avs", same treatment applies. And thus decided
to make it part of this series instead of sending it separately.
Jaroslav Kysela [Tue, 29 Nov 2022 08:53:06 +0000 (09:53 +0100)]
selftests: alsa - move shared library configuration code to conf.c
The minimal alsa-lib configuration code is similar in both mixer
and pcm tests. Move this code to the shared conf.c source file.
Also, fix the build rules inspired by rseq tests. Build libatest.so
which is linked to the both test utilities dynamically.
Also, set the TEST_FILES variable for lib.mk.
Cc: linux-kselftest@vger.kernel.org Cc: Shuah Khan <shuah@kernel.org> Reported-by: Mark Brown <broonie@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Tested-by: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20221129085306.2345763-1-perex@perex.cz Signed-off-by: Takashi Iwai <tiwai@suse.de>
John Keeping [Tue, 29 Nov 2022 13:00:59 +0000 (13:00 +0000)]
ALSA: usb-audio: Add quirk for Tascam Model 12
Tascam's Model 12 is a mixer which can also operate as a USB audio
interface. The audio interface uses explicit feedback but it seems that
it does not correctly handle missing isochronous frames.
When injecting an xrun (or doing anything else that pauses the playback
stream) the feedback rate climbs (for example, at 44,100Hz nominal, I
see a stable rate around 44,099 but xrun injection sees this peak at
around 44,135 in most cases) and glitches are heard in the audio stream
for several seconds - this is significantly worse than the single glitch
expected for an underrun.
While the stream does normally recover and the feedback rate returns to
a stable value, I have seen some occurrences where this does not happen
and the rate continues to increase while no audio is heard from the
output. I have not found a solid reproduction for this.
This misbehaviour can be avoided by totally resetting the stream state
by switching the interface to alt 0 and back before restarting the
playback stream.
Add a new quirk flag which forces the endpoint and interface to be
reconfigured whenever the stream is stopped, and use this for the Tascam
Model 12.
Separate interfaces are used for the playback and capture endpoints, so
resetting the playback interface here will not affect the capture stream
if it is running. While there are two endpoints on the interface,
these are the OUT data endpoint and the IN explicit feedback endpoint
corresponding to it and these are always stopped and started together.
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Cezary Rojewski [Fri, 25 Nov 2022 18:40:22 +0000 (19:40 +0100)]
ASoC: Intel: bdw_rt286: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
ASoC: qcom: lpass-sc7180: Delete redundant error log from _resume()
sc7180_lpass_dev_resume() logs an error if clk_bulk_prepare_enable()
fails. The clock framework already generates error logs if anything
goes wrong, so the logging in _resume() is redundant, drop it.
ASoC: qcom: lpass-sc7280: Add system suspend/resume PM ops
Update lpass sc7280 platform driver with PM ops, such as
system supend and resume callbacks.
This update is required to disable clocks during supend and
avoid XO shutdown issue.
Mark Brown [Mon, 28 Nov 2022 16:38:18 +0000 (16:38 +0000)]
ASoC: dt-bindings: Rework Qualcomm APR/GPR Sound nodes for SM8450
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Adding sound support for Qualcomm SM8450 SoC (and later for SC8280XP) brought
some changes to APR/GPR services bindings. These bindings are part of
qcom,apr.yaml:
The schema for services (apr-gpr-service@[0-9]) was not complete and is still
quite not specific. It allows several incorrect combinations, like adding a
clock-controller to a APM device. Restricting it would complicate the schema
even more. Bringing new support for sound on Qualcomm SM8450 and SC8280XP SoC
would grow it as well.
Refactor the bindings before extending them for Qualcomm SM8450 SoC.
ASoC: qcom: lpass-sc7180: Add system suspend/resume PM ops
Update lpass sc7180 platform driver with PM ops, such as
system supend and resume callbacks.
This update is required to disable clocks during supend and
avoid XO shutdown issue.
Oder Chiou [Mon, 28 Nov 2022 07:08:25 +0000 (15:08 +0800)]
ASoC: rt5640: Fix Jack work after system suspend
We found an corner case in RT5640 codec driver which schedules jack work
after system suspend as IRQ was enabled. Due to this, hitting the error
as register access happening after suspend as jack worker thread getting
scheduled in irq handler. The patch disables the irq during the suspend
to prevent the corner case happening.
Jean Delvare [Sun, 27 Nov 2022 18:35:49 +0000 (19:35 +0100)]
ASoC: SOF: Drop obsolete dependency on COMPILE_TEST
Since commit 0166dc11be91 ("of: make CONFIG_OF user selectable"), it
is possible to test-build any driver which depends on OF on any
architecture by explicitly selecting OF. Therefore depending on
COMPILE_TEST as an alternative is no longer needed.
Pavel Dobias [Wed, 23 Nov 2022 15:38:18 +0000 (16:38 +0100)]
ASoC: max9867: Implement exact integer mode
For 8kHz and 16kHz sample rates and certain PCLK values
the codec can be programmed to operate in exact integer
mode. If available, use it to achieve the exact sample rate.
ASoC: rsnd: adg: use __clk_get_name() instead of local clk_name[]
Current rsnd_adg_clk_dbg_info() is using locak clk_name[] to ndicating
clk name, but we don't want to use local clk_name[] everywhere when we
support R-Car Gen4 sound to handling non compatible clk naming.
This patch uses __clk_get_name() instead of local clk_name[] for it.
Charles Keepax [Wed, 23 Nov 2022 16:54:32 +0000 (16:54 +0000)]
ASoC: sdw-mockup: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:31 +0000 (16:54 +0000)]
ASoC: rt715: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:30 +0000 (16:54 +0000)]
ASoC: rt711: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:29 +0000 (16:54 +0000)]
ASoC: rt700: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:28 +0000 (16:54 +0000)]
ASoC: rt5682-sdw: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:27 +0000 (16:54 +0000)]
ASoC: rt1316-sdw: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:26 +0000 (16:54 +0000)]
ASoC: rt1308-sdw: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:25 +0000 (16:54 +0000)]
ASoC: max98373-sdw: Switch to new snd_sdw_params_to_config helper
The conversion from hw_params to SoundWire config is pretty
standard as such most of the conversion can be handled by the new
snd_sdw_params_to_config helper function.
Charles Keepax [Wed, 23 Nov 2022 16:54:24 +0000 (16:54 +0000)]
sound: sdw: Add hw_params to SoundWire config helper function
The vast majority of the current users of the SoundWire framework
have almost identical code for converting from hw_params to SoundWire
configuration. Whilst complex devices might require more, it is very
likely that most new devices will follow the same pattern. Save a
little code by factoring this out into a helper function.
Jean Delvare [Sun, 27 Nov 2022 18:34:41 +0000 (19:34 +0100)]
ASoC: rsnd: Drop obsolete dependency on COMPILE_TEST
Since commit 0166dc11be91 ("of: make CONFIG_OF user selectable"), it
is possible to test-build any driver which depends on OF on any
architecture by explicitly selecting OF. Therefore depending on
COMPILE_TEST as an alternative is no longer needed.
ASoC: SOF: amd: Fix for selecting clock source as external clock.
By default clock source is selected as internal clock of 96Mhz
which is not configurable. Now we select the clock source to
external clock (ACLK) which can be configurable to different clock
ranges depending on usecase.
ASoC: SOF: amd: Fix for reading position updates from stream box.
By default the position updates are read from dsp box when streambox
size is not defined.if the streambox size is defined to some value
then position updates can be read from the streambox.
On SM8450 and SC8280XP, the Q6APM is a bit different:
1. It is used as a platform DAI link, so it needs #sound-dai-cells.
2. It has two DAI children, so add new "bedais" node.
ASoC: dt-bindings: qcom,q6apm-lpass-dais: Split to separate schema
The Qualcomm DSP LPASS Audio DAIs are a bit different than Qualcomm DSP
Audio FrontEnd (Q6AFE) DAIs - they do not use children nodes for each
DAI. None of other properties from qcom,q6dsp-lpass-ports.yaml apply
here as well, so move the qcom,q6apm-lpass-dais compatible to its own
binding.
ASoC: dt-bindings: qcom,q6core: Split to separate schema
The APR/GPR bindings with services got complicated so move out the
Q6Core service to its own binding. Previously the compatible was
documented in qcom,apr.yaml.
ASoC: dt-bindings: qcom,q6prm: Split to separate schema
The APR/GPR bindings with services got complicated so move out the Q6PRM
service to its own binding. Previously the compatible was documented in
qcom,apr.yaml.
ASoC: dt-bindings: qcom,q6asm: Split to separate schema
The APR/GPR bindings with services got complicated so move out the Q6ASM
service to its own binding. Previously the compatible was documented in
qcom,apr.yaml. Move most of the examples from its children to this new
file.
ASoC: dt-bindings: qcom,q6adm: Split to separate schema
The APR/GPR bindings with services got complicated so move out the Q6ADM
service to its own binding. Previously the compatible was documented in
qcom,apr.yaml. Move most of the examples from its children to this new
file.
ASoC: dt-bindings: qcom,q6apm: Split to separate schema
The APR/GPR bindings with services got complicated so move out the Q6APM
service to its own binding. Previously the compatible was documented in
qcom,apr.yaml. Move most of the examples from its children to this new
file.
ASoC: dt-bindings: qcom,q6afe: Split to separate schema
The APR/GPR bindings with services got complicated so move out the Q6AFE
service to its own binding. Previously the compatible was documented in
qcom,apr.yaml. Move most of the examples from its children to this new
file.
ASoC: dt-bindings: qcom,apr: Correct and extend example
Correct the APR/GPR example:
1. Use consistent 4-space indentation,
2. Add required properties to services nodes, so the binding check
passes once schema for these services is improved,
3. Add few other properties as APR/GPR is part of a GLINK edge:
qcom,glink-channels and qcom,intents.
4. Drop unnecessary services, to make the example compact.
The schema for services (apr-gpr-service@[0-9]) already grows
considerably and is still quite not specific. It allows several
incorrect combinations, like adding a clock-controller to a APM device.
Restricting it would complicate the schema even more. Bringing new
support for sound on Qualcomm SM8450 and SC8280XP SoC would grow it as
well.
Simplify the qcom,apr.yaml by splitting the services to a shared file
which will be:
1. Referenced by qcom,apr.yaml with additionalProperties:true,
2. Referenced by specific bindings for services with
additionalProperties:false (not yet in this commit).
While moving the code, add also required 'reg' and
'qcom,protection-domain' to further constrain the bindings.
Mark Brown [Fri, 25 Nov 2022 21:39:20 +0000 (21:39 +0000)]
RK3588 Audio Support
Merge series from Nicolas Frattaroli <frattaroli.nicolas@gmail.com>:
This patchset refactors the Rockchip I2S/TDM driver in order to
support the RK3588 SoC, and then adds the necessary compatible
string to load the driver for it.
Patch 1 rectifies a problem with the bindings where we were too
strict about requiring the rockchip,grf property. Most features
of this audio device don't need access to the GRF to function.
Patch 2 modifies the driver to adjust its behaviour to what the
changed bindings now allow, namely using most things without the
GRF.
Patch 3 and 4 are boring compatible string stuff that enables
RK3588 support. No special data is needed to initialise the
driver for this instance of the I2S/TDM IP.
Maarten Zanders [Fri, 28 Oct 2022 15:26:25 +0000 (17:26 +0200)]
ASoC: adau1372: correct PGA enable & mute bit
The DAPM control for PGAx uses the PGA mute bit for
power management. This bit is active high but is set to
non-inverted (ie when powering, it will mute).
The ALSA control "PGA x Capture Switch" uses the active
high PGA_ENx bit, but is set to inverted. So when
enabling this switch, the PGA gets disabled.
Maarten Zanders [Fri, 28 Oct 2022 15:26:23 +0000 (17:26 +0200)]
ASoC: adau1372: fix mclk
"mclk" is retrieved from the configuration and assigned to adau1372->clk.
However adau1372->mclk (==NULL) is used for clk_prepare_enable() and
clk_disable_unprepare() which don't have any effect.
Remove .clk from struct adau1372 and use .mclk throughout.
This change ensures that the input clock is switched on/off when the
bias level is changed.
ASoC: rockchip: i2s_tdm: Make the grf property optional
Only IO Multiplex and two TRCM modes need access to the GRF, so
making it a hard requirement is not a wise idea, as it complicates
support for newer SoCs which do not do these things.
projects / people / aperard / linux.git / log