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struct sdw_slave_prop is zero-initialized by the SoundWire core so there
is no need to clear clk_stop_mode1 to false. Removing this also avoids
having an unnecessary build dependency on a struct member.
Signed-off-by: Richard Fitzgerald <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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No product was ever released with A1 silicon so there is no
need for the driver to include support for it.
Signed-off-by: Richard Fitzgerald <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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This patch reverts a series of commits that allowed for the ASP
registers to be owned by either the driver or the firmware. Nothing
currently depends on the functionality that is being reverted, so
it is safe to remove.
The commits being reverted are (last 3 are bugfixes to the first 2):
commit 72a77d7631c6
("ASoC: cs35l56: Fix to ensure ASP1 registers match cache")
commit 07f7d6e7a124
("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
commit 4703b014f28b
("ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()")
commit c14f09f010cc
("ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization")
commit dfd2ffb37399
("ASoC: cs35l56: Prevent overwriting firmware ASP config")
These reverts have been squashed into a single commit because there
would be no reason to revert only some of them (which would just
reintroduce bugs).
The changes introduced by the commits were well-intentioned but
somewhat misguided. ACPI does not provide any information about how
audio hardware is linked together, so that information has to be
hardcoded into drivers. On Windows the firmware is customized to
statically setup appropriate configuration of the audio links,
and the intent of the commits was to re-use this information if the
Linux host drivers aren't taking control of the ASP. This would
avoid having to hardcode the ASP config into the machine driver on
some systems.
However, this added complexity and race conditions into the driver.
It also complicates implementation of new code.
The only case where the ASP is used but the host is not taking
ownership is when CS35L56 is used in SoundWire mode with the ASP
as a reference audio interconnect. But even in that case it's not
necessarily required even if the firmware initialized it. Typically
it is used to avoid the host SDCA drivers having to be capable of
aggregating capture paths from multiple SoundWire peripherals. But
the SOF SoundWire support is capable of doing that aggregation.
Reverting all these commits significantly simplifies the driver.
Let's just use the normal Linux mechanisms of the machine driver and
ALSA controls to set things up instead of trying to use the firmware
to do use-case setup.
Signed-off-by: Richard Fitzgerald <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Merge series from Piotr Wojtaszczyk <[email protected]>:
This pach set is to bring back audio to machines with a LPC32XX CPU.
The legacy LPC32XX SoC used to have audio spport in linux 2.6.27.
The support was dropped due to lack of interest from mainaeners.
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Add support for 705.6kHz and 768kHz sample rates
Signed-off-by: Jerome Brunet <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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On i.MX8MP, there are 3 interrupts, the third interrupt is
SPDIF wakeup interrupt from PHY
Add ref: dai-common.yaml for #sound-dai-cells is needed and
XCVR is a DAI device.
Signed-off-by: Shengjiu Wang <[email protected]>
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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clk_prepare_enable() may fail, so we should better check its return
value and propagate it in the case of error.
Fixes: 62a7fc32a628 ("ASoC: max98088: Add master clock handling")
Signed-off-by: Chen Ni <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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./sound/soc/codecs/ak4619.c:757:2-3: Unneeded semicolon
Reported-by: Abaci Robot <[email protected]>
Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=9442
Signed-off-by: Yang Li <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The dummy DAI should allow any (reasonable) rates possible.
Make the rate continuous for dummy and set range from 5512Hz to 768kHz
The change is mostly cosmetic as dummy is skipped when setting
the hwparams.
Signed-off-by: Jerome Brunet <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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This driver was ported from an old version in linux 2.6.27 and adjusted
for the new ASoC framework and DMA API.
Signed-off-by: Piotr Wojtaszczyk <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add nxp,lpc3220-i2s DT binding documentation.
Signed-off-by: Piotr Wojtaszczyk <[email protected]>
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The Qualcomm LPASS WSA macro codec driver uses now parts of common
module, so it has to select SND_SOC_LPASS_MACRO_COMMON.
sound/soc/codecs/lpass-wsa-macro.o: in function `wsa_macro_probe':
sound/soc/codecs/lpass-wsa-macro.c:2767:(.text+0x1c9c): undefined reference to `lpass_macro_get_codec_version'
Reported-by: kernel test robot <[email protected]>
Closes: https://lore.kernel.org/oe-kbuild-all/[email protected]/
Fixes: 5dcf442bbbca ("ASoC: codecs: lpass-wsa-macro: Prepare to accommodate new codec versions")
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Merge series from Shengjiu Wang <[email protected]>:
The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately in fsl_sai driver.
There will be three devices for audiomix sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output.
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Merge series from Kuninori Morimoto <[email protected]>:
Introduce the ability for sound cards to manually order the startup of
the various components in the card.
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Inside of test_pcm_time() arguments are printed via printf
but '%d' is used to print @flags (of type unsigned int).
Use '%u' instead, just like we do everywhere else.
Signed-off-by: Zhu Jun <[email protected]>
Reviewed-by: Mark Brown <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Signed-off-by: Takashi Iwai <[email protected]>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.10
A relatively large batch of updates, largely due to the long interval
since I last sent fixes due to various travel and holidays. There's a
lot of driver specific fixes and quirks in here, none of them too major,
and also some fixes for recently introduced memory safety issues in the
topology code.
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The version B will support the multi-lane function and integrate the DMIC function
in one SoundWire interface.
Due to some registers having different default values between version A and B,
this patch also removes the redundant default registers to avoid confusion.
Signed-off-by: Shuming Fan <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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This patch doesn't have any change of functionality.
Signed-off-by: Shuming Fan <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Like "audio-routing" drop the minItems: 2 from the "audio-widgets", because
any limit here - lower or upper- is rather meaningless.
This will also fix `dtbs_check` warnings like:
sound: audio-widgets: ['Speaker', '7J4-14 LEFT', 'Speaker', '7J4-11 RIGHT'] is too long
Signed-off-by: Neil Armstrong <[email protected]>
Acked-by: Rob Herring (Arm) <[email protected]>
Link: https://patch.msgid.link/20240605-topic-amlogic-upstream-bindings-fixes-audio-widgets-v1-1-65bd7cc2e09b@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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asoc_qcom_lpass_cpu_platform_probe()
If IORESOURCE_MEM "lpass-rxtx-cdc-dma-lpm" or "lpass-va-cdc-dma-lpm"
resources is not provided in Device Tree due to any error,
platform_get_resource_byname() will return NULL which is later
dereferenced. According to sound/qcom,lpass-cpu.yaml, these resources
are provided, but DT can be broken due to any error. In such cases driver
must be able to protect itself, since the DT is external data for the
driver.
Adjust this issues by adding NULL return check.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: b138706225c9 ("ASoC: qcom: Add regmap config support for codec dma driver")
Signed-off-by: Aleksandr Mishin <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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There will be three devices for this sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output. then capture device and playback device can be configured
with different master/slave mode.
Signed-off-by: Shengjiu Wang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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As audmix requires playback and capture stream in different
master/slave mode, so separate playback and capture stream to
different DAI. There are three DAIs required, two DAIs for playback
one DAI for capture.
Signed-off-by: Shengjiu Wang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.
+-------+ +-----------+
| SAI1 | --TX--> | |
| | <--RX-- | |
+-------+ | |
| AUDIOMIX |
+-------+ | |
| SAI2 | --TX--> | |
+-------+ +-----------+
The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately. but only
defined fsl_sai_set_dai_fmt_tx() and fsl_sai_set_dai_fmt_rx()
ops function for current case, in the future, the other ops
function for TX and RX can be defined if required.
Signed-off-by: Shengjiu Wang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Audio Graph
Card2 still not yet support it. Let's support it.
Cc: Maxim Kochetkov <[email protected]>
Signed-off-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Audio Graph
Card still not yet support it. Let's support it.
Cc: Maxim Kochetkov <[email protected]>
Signed-off-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Simple Audio
Card still not yet support it. Let's support it.
Cc: Maxim Kochetkov <[email protected]>
Signed-off-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Some Sound Card might need special trigger ordering which is based on
CPU/Codec connection. It is already supported on ASoC, but Simple Audio
Card / Audio Graph Card still not support it. Let's support it.
Cc: Maxim Kochetkov <[email protected]>
Signed-off-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Sound Card need to consider/adjust HW control ordering based on the
combination of CPU/Codec. The controlling feature is already supported
on ASoC, but Simple Audio Card / Audio Graph Card still not support it.
Let's support it.
Cc: Maxim Kochetkov <[email protected]>
Signed-off-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Replace 'int' with proper 'enum lpass_codec_version' in every place
which handles the parsed codec version (not raw register values!) to be
explicit about contents of the variable. This makes code easier to read
and compilers could check missing switch cases.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Reviewed-by: Dmitry Baryshkov <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Qualcomm LPASS macro codec driver parses registers in order to
detect version of the codec. It recognizes codecs v2.0 - v2.8, however
we know that there are earlier versions and 'enum lpass_codec_version'
has also v1.0, v1.1 and v1.2. If by any chance we run on unrecognized
version, driver will use random value from the stack as the codec
version.
Fix it by mapping such cases to an enum of value 0:
LPASS_CODEC_VERSION_UNKNOWN.
Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version")
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Reviewed-by: Dmitry Baryshkov <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Modify register setting sequence of enabling inline command
to fix issue of random interrupt from push-button.
Signed-off-by: Jack Yu <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The conversion of SPP to MIDI2 UMP called a wrong function, and the
secondary argument wasn't taken. As a result, MSB of SPP was always
zero. Fix to call the right function.
Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events")
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Starting with v2.5 of Qualcomm LPASS Codec, few registers in the WSA
macro block change. Bring proper support for this v2.5 and newer
versions, to fix second speaker playback (speaker was silent).
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Dmitry Baryshkov <[email protected]>
Link: https://patch.msgid.link/20240625-qcom-audio-wsa-second-speaker-v1-3-f65ffdfc368c@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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The driver for Qualcomm LPASS WSA macro codec was developed and tested
on codec v2.1, however v2.5 has significant changes in the registers.
The driver correctly works for v2.1 codec, but has issues when running
on SoC with v2.5 codec (so starting with SM8450, even though playback
works properly on that SoC).
Prepare the driver for handling differences in register layouts of newer
version. This does not have functional impact on older codec versions,
but just:
1. Renames few soc_enums and widgets as v2.1,
2. For registers being different in v2.5, moves the defaults and regmap
configuration to new structures,
3. Adds new 'struct wsa_reg_layout' with offsets and masks for few
registers, so most of the code can stay unchaged on v2.5,
4. Chooses proper widgets, regmap config and register layout based on
version of the codec.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Dmitry Baryshkov <[email protected]>
Link: https://patch.msgid.link/20240625-qcom-audio-wsa-second-speaker-v1-2-f65ffdfc368c@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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WSA_MACRO_MUX_INP_MASK2 define is not used.
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Acked-by: Dmitry Baryshkov <[email protected]>
Link: https://patch.msgid.link/20240625-qcom-audio-wsa-second-speaker-v1-1-f65ffdfc368c@linaro.org
Signed-off-by: Mark Brown <[email protected]>
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The Vivobook S 16X IPS needs a quirks-table entry for the internal microphone to function properly.
Signed-off-by: Vyacheslav Frantsishko <[email protected]>
Reviewed-by: Mario Limonciello <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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HP EliteBook 645/665 G11 needs ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to
make mic-mute/audio-mute working.
Signed-off-by: Dirk Su <[email protected]>
Cc: <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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The recent fix for Lenovo IdeaPad 330-17IKB replaced the quirk entry,
and this eventually breaks the existing quirk for Lenovo Yoga Duet 7
13ITL6 equipped with the same PCI SSID 17aa:3820.
For applying a proper quirk for each model, check the codec SSID
additionally. Fortunately Yoga Duet has a different codec SSID,
0x17aa3802.
(Interestingly, 17aa:3802 has another conflict of SSID between another
Yoga model vs 14IRP8 which we had to work around similarly.)
Fixes: b1fd0d1285b1 ("ALSA: hda/realtek: Enable headset mic on IdeaPad 330-17IKB 81DM")
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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The SDCA spec defines a 'selected_mode' control which can override
the 'detected_mode' reported by hardware.
This is useful for platform integration as well as in cases
where the hardware(e.g. 3.5mm jack cable) is not able to accurately detect the jack type.
Signed-off-by: Shuming Fan <[email protected]>
Tested-by: [email protected]
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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We have also v2.1 version of the codec (see 'enum lpass_codec_version'),
so handle it as well in all switch cases.
Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions")
Signed-off-by: Krzysztof Kozlowski <[email protected]>
Reviewed-by: Srinivas Kandagatla <[email protected]>
Link: https://patch.msgid.link/[email protected]
Reviewed-by: Dmitry Baryshkov <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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i2c_device_id::driver_data to 0
These drivers don't use the driver_data member of struct i2c_device_id,
so don't explicitly initialize this member.
This prepares putting driver_data in an anonymous union which requires
either no initialization or named designators. But it's also a nice
cleanup on its own.
Signed-off-by: Uwe Kleine-König <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Commit ae8fc2948b48 ("ASoC: dt-bindings: add missing vender prefix on
filename") renames a few files in Documentation/devicetree/bindings/sound/,
but misses to adjust the file entries pointing to those files in
MAINTAINERS.
Hence, ./scripts/get_maintainer.pl --self-test=patterns complains about
broken references.
Adjust the file entries in NXP SGTL5000 DRIVER and TEXAS INSTRUMENTS AUDIO
(ASoC/HDA) DRIVERS.
Signed-off-by: Lukas Bulwahn <[email protected]>
Acked-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Merge series from Pierre-Louis Bossart <[email protected]>:
Minor additions and cleanups this time.
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Merge series from Zhang Yi <[email protected]>:
We propose four patches to solve headphone detection and suspend issues.
And there are several registers that should be read-only registers. So
we create es8326_writeable_register, and set these registers to false.
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Both snd_dmaengine_pcm_close() and
snd_dmaengine_pcm_close_release_chan() are almost identical except for
a call of dma_release_channel() in the latter. For code
simplification, unify them internally.
Just a code refactoring, and no functional changes.
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Pull 6.10-rc devel branch for further cleanups of dmaengine and others.
Signed-off-by: Takashi Iwai <[email protected]>
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When dmaengine supports pause function, in suspend state,
dmaengine_pause() is called instead of dmaengine_terminate_async(),
In end of playback stream, the runtime->state will go to
SNDRV_PCM_STATE_DRAINING, if system suspend & resume happen
at this time, application will not resume playback stream, the
stream will be closed directly, the dmaengine_terminate_async()
will not be called before the dmaengine_synchronize(), which
violates the call sequence for dmaengine_synchronize().
This behavior also happens for capture streams, but there is no
SNDRV_PCM_STATE_DRAINING state for capture. So use
dmaengine_tx_status() to check the DMA status if the status is
DMA_PAUSED, then call dmaengine_terminate_async() to terminate
dmaengine before dmaengine_synchronize().
Signed-off-by: Shengjiu Wang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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This HP Laptop uses ALC236 codec with COEF 0x07 controlling
the mute LED. Enable existing quirk for this device.
Signed-off-by: Aivaz Latypov <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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snd_pcm_resume() should bail out if the stream isn't in a suspended
state. Otherwise it'd allow doubly resume.
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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