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case name_prefix is not NULL
fix to save the dsp bin file name into the correct array, coef_binaryname,
instead of rca_binaryname, in case name_prefix is not NULL.
Signed-off-by: Shenghao Ding <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Parameter xindex is not used in macro HDSPM_TCO_LTC_FRAMES and
HDSPM_TCO_VIDEO_INPUT_FORMAT,so just remove it to simplify the code.
Signed-off-by: He Lugang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Since there are a few corner cases where the S/G buffer allocation
isn't performed (e.g. depending on IOMMU implementations), it'd be
better to allow the default buffer preallocation size for x86, too.
The default for x86 is still kept to 0, as it should work in most
cases.
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Hopefully the last PR for 6.11, at least for this level of amount.
In addition to the usual HD-audio quirks, there are more changes in
ASoC, but all look small and device-specific fixes, and nothing stands
out. The only slightly big change is sunxi I2S fix, which looks quite
safe to apply, too"
* tag 'sound-6.11-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
ALSA: hda/realtek - Fix inactive headset mic jack for ASUS Vivobook 15 X1504VAP
ALSA: hda/realtek: Support mute LED on HP Laptop 14-dq2xxx
ALSA: hda/realtek: Enable Mute Led for HP Victus 15-fb1xxx
ALSA: hda/realtek: extend quirks for Clevo V5[46]0
ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
ALSA: hda: add HDMI codec ID for Intel PTL
ALSA: hda/realtek: add patch for internal mic in Lenovo V145
ASoC: sunxi: sun4i-i2s: fix LRCLK polarity in i2s mode
ASoC: amd: yc: Add a quirk for MSI Bravo 17 (D7VEK)
ASoC: mediatek: mt8188-mt6359: Modify key
ASoc: SOF: topology: Clear SOF link platform name upon unload
ALSA: hda/conexant: Add pincfg quirk to enable top speakers on Sirius devices
ASoC: SOF: ipc: replace "enum sof_comp_type" field with "uint32_t"
ASoC: fix module autoloading
ASoC: tda7419: fix module autoloading
ASoC: google: fix module autoloading
ASoC: intel: fix module autoloading
ASoC: tegra: Fix CBB error during probe()
ASoC: dapm: Fix UAF for snd_soc_pcm_runtime object
ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
...
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The devm_clk_get_enabled() helpers:
- call devm_clk_get()
- call clk_prepare_enable() and register what is needed in order to
call clk_disable_unprepare() when needed, as a managed resource.
This simplifies the code and avoids the calls to clk_disable_unprepare().
Signed-off-by: ying zuxin <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Return when error occurs instead of proceeding to for loop which will
use val uninitialized.
Fixes: f6f7d25b1103 ("ASoC: amd: acp: Add pte configuration for ACP7.0 platform")
Signed-off-by: Muhammad Usama Anjum <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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In the function loongson_asoc_card_probe, it is simpler
to return the value of function devm_snd_soc_register_card
directly.
Signed-off-by: Tang Bin <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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There is no links_num in struct snd_soc_acpi_mach {}, and we test
!link->num_adr as a condition to end the loop in hda_sdw_machine_select().
So an empty item in struct snd_soc_acpi_link_adr array is required.
Fixes: f77ae7fcdc4763 ("ASoC: Intel: soc-acpi-intel-mtl-match: add cs42l43 only support")
Signed-off-by: Bard Liao <[email protected]>
Reviewed-by: Ranjani Sridharan <[email protected]>
Reviewed-by: Péter Ujfalusi <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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There is no links_num in struct snd_soc_acpi_mach {}, and we test
!link->num_adr as a condition to end the loop in hda_sdw_machine_select().
So an empty item in struct snd_soc_acpi_link_adr array is required.
Fixes: dd3bd9dc4708 ("ASoC: Intel: soc-acpi-intel-lnl-match: add cs42l43 only support")
Signed-off-by: Bard Liao <[email protected]>
Reviewed-by: Ranjani Sridharan <[email protected]>
Reviewed-by: Péter Ujfalusi <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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IEC958-3 defines sampling rate up to 768 kHz.
Such rates maybe used with high bandwidth IEC958 links, such as eARC.
Signed-off-by: Jerome Brunet <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The cs35l34 adds a useless rate constraint on startup.
It does not set SNDRV_PCM_RATE_KNOT and the rates set are already a subset
of the ones provided in the constraint list, so it is a no-op.
>From the rest of the code, it is likely HW supports more than the 32, 44.1
and 48kHz listed in CS35L34_RATES but there is no way to know for sure
without proper documentation.
Keep the driver as it is for now and just drop the useless constraint.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint lists was necessary to support 12kHz and 24kHz.
These rates are now available through SNDRV_PCM_RATE_12000 and
SNDRV_PCM_RATE_24000.
Use them and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 12kHz and 24kHz.
These rates are now available through SNDRV_PCM_RATE_12000 and
SNDRV_PCM_RATE_24000.
Use them and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 12kHz, 24kHz and
128kHz. These rates are now available through SNDRV_PCM_RATE_12000,
SNDRV_PCM_RATE_24000 and SNDRV_PCM_RATE_128000.
Use them and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 12kHz and 24kHz.
These rates are now available through SNDRV_PCM_RATE_12000 and
SNDRV_PCM_RATE_24000.
Use them and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 12kHz and 24kHz.
These rates are now available through SNDRV_PCM_RATE_12000 and
SNDRV_PCM_RATE_24000.
Use them and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 12kHz and 24kHz.
These rates are now available through SNDRV_PCM_RATE_12000 and
SNDRV_PCM_RATE_24000.
Use them and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 128kHz.
This rate is now available through SNDRV_PCM_RATE_128000.
Use it and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 128kHz.
This rate is now available through SNDRV_PCM_RATE_128000.
Use it and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint lists were necessary to support 12kHz and 24kHz.
These rates are now available through SNDRV_PCM_RATE_12000 and
SNDRV_PCM_RATE_24000.
Use them and drop the custom rate constraint rules.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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The custom rate constraint list was necessary to support 128kHz.
This rate is now available through SNDRV_PCM_RATE_128000.
Use it and drop the custom rate constraint rule.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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This adds a sample rate definition for 12kHz, 24kHz and 128kHz.
Admittedly, just a few drivers are currently using these sample
rates but there is enough of a recurrence to justify adding a definition
for them and remove some custom rate constraint code while at it.
The new definitions are not added to the interval definitions, such as
SNDRV_PCM_RATE_8000_44100, because it would silently add new supported
rates to drivers that may or may not support them. For sure the drivers
have not been tested for these new rates so it is better to leave them out
of interval definitions.
That being said, the added rates are multiples of well know rates families,
it is very likely that a lot of devices out there actually supports them.
Signed-off-by: Jerome Brunet <[email protected]>
Reviewed-by: David Rhodes <[email protected]>
Acked-by: Mark Brown <[email protected]>
Reviewed-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.11
A larger set of fixes than I'd like at this point, but mainly due to
people working on fixing module autoloading by adding missing exports of
ID tables rather than anything particularly concerning. There are some
other runtime fixes and quirks, and a tweak to the ABI definition for
SOF which ensures that a struct layout doesn't vary depending on the
architecture of the host.
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Replace comma between expressions with semicolons.
Using a ',' in place of a ';' can have unintended side effects.
Although that is not the case here, it is seems best to use ';'
unless ',' is intended.
Found by inspection.
No functional change intended.
Compile tested only.
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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Replace comma between expressions with semicolons.
Using a ',' in place of a ';' can have unintended side effects.
Although that is not the case here, it is seems best to use ';'
unless ',' is intended.
Found by inspection.
No functional change intended.
Compile tested only.
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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Merge series from [email protected]:
This serie aim to add the following audio support for the Genio 350-evk:
- Playback
- 2ch Headset Jack (Earphone)
- 1ch Line-out Jack (Speaker)
- 8ch HDMI Tx
- Capture
- 1ch DMIC (On-board Digital Microphone)
- 1ch AMIC (On-board Analogic Microphone)
- 1ch Headset Jack (External Analogic Microphone)
Of course, HDMI playback need the MT8365 display patches [1] and a DTS
change documented in "mediatek,mt8365-mt6357.yaml".
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When the headset is connected, there is no automatic switching of the
capture source - you can only manually select the headset microphone
in pavucontrol.
This patch fixes/activates the inactive microphone of the headset.
Signed-off-by: Vasiliy Kovalev <[email protected]>
Cc: <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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The period size represents the size of the DMA descriptor. To ensure all
DMA descriptors start from a well-aligned address, the period size must
be divided by (sample size * maxburst), not just by maxburst.
This adjustment allows for computing a higher maxburst value, thereby
increasing the performance of the DMA transfer.
Previously, snd_pcm_lib_period_bytes() returned 0 because the runtime HW
parameters are computed after the hw_params() callbacks are used.
To address this, we now use params_*() functions to compute the period
size accurately. This change optimizes the DMA transfer performance by
ensuring proper alignment and efficient use of maxburst values.
[[email protected]: Reword commit message and commit title.
Add macros with values for maximum DMA chunk size allowed.
Add DMA_BURST_ALIGNED preprocessor function to check the alignment of the
DMA burst]
Signed-off-by: Codrin Ciubotariu <[email protected]>
Signed-off-by: Andrei Simion <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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- Add specific config to enable:
- MT8365 sound support
- MT6357 audio codec support
- Add the mt8365 directory and all drivers under it.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add the support of MT6357 PMIC audio codec.
Signed-off-by: Nicolas Belin <[email protected]>
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Merge series from Zhang Zekun <[email protected]>:
There are some helper functions which can be used to simplify the code.
So, let's use these functions to make code more simple.
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of_get_child_count() can help to get the num of child directly and we
don't need to manually count the child num. No functional change with
this conversion.
Signed-off-by: Zhang Zekun <[email protected]>
Reviewed-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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for_each_child_of_node_scoped() can put the device_node automatically.
So, using it to make the code logic more simple and remove the device_node
clean up code.
Signed-off-by: Zhang Zekun <[email protected]>
Reviewed-by: Kuninori Morimoto <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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In the former commits, the callback of isochronous context runs on usual
work process. In the case, ALSA PCM device has a flag, nonatomic, to
acquire mutex lock instead of spin lock for PCM substream group.
This commit uses the flag. It has an advantage in the case that ALSA PCM
application uses the large size of intermediate buffer, since it takes
too long time even in tasklet softIRQ to process many of isochronous
packets, then result in the delay of system event due to disabled IRQ so
long. It is avertible to switch to nonatomic operation.
Reviewed-by: Takashi Iwai <[email protected]>
Tested-by: Edmund Raile <[email protected]>
Link: https://lore.kernel.org/r/[email protected]
Signed-off-by: Takashi Sakamoto <[email protected]>
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mt2701_cs42448_machine_probe
Utilize the defined parameter 'dev' to make the code cleaner.
Signed-off-by: Liu Jing <[email protected]>
Reviewed-by: Matthias Brugger <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device.
Signed-off-by: Maximilien Perreault <[email protected]>
Cc: <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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The mute led is controlled by ALC245. This patch enables the already
existing quirk for this device.
Signed-off-by: Adam Queler <[email protected]>
Cc: <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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Merge series from Venkata Prasad Potturu <[email protected]>:
1. Refactor acp generic driver to support all platforms.
2. Add i2s/tdm and support for acp7.0 and acp7.1 platforms.
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Merge series from Alexandre Mergnat <[email protected]>:
This serie aim to add the following audio support for the Genio 350-evk:
- Playback
- 2ch Headset Jack (Earphone)
- 1ch Line-out Jack (Speaker)
- 8ch HDMI Tx
- Capture
- 1ch DMIC (On-board Digital Microphone)
- 1ch AMIC (On-board Analogic Microphone)
- 1ch Headset Jack (External Analogic Microphone)
Of course, HDMI playback need the MT8365 display patches [1] and a DTS
change documented in "mediatek,mt8365-mt6357.yaml".
Applied patch:
- mfd: mt6397-core: register mt6357 sound codec
Test passed:
- mixer-test log: [3]
- pcm-test log: [4]
[1]: https://lore.kernel.org/all/[email protected]/
[2]: https://lore.kernel.org/all/[email protected]/
[3]: https://pastebin.com/pc43AVrT
[4]: https://pastebin.com/cCtGhDpg
[5]: https://gitlab.baylibre.com/baylibre/mediatek/bsp/linux/-/commits/sound/for-next/add-i350-audio-support
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In function loongson_card_parse_of(), when get device_node
'codec' failed, the function of_node_put(codec) should not
be invoked, thus fix error release.
Fixes: d24028606e76 ("ASoC: loongson: Add Loongson ASoC Sound Card Support")
Signed-off-by: tangbin <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Support multiple endpoints on TLV320AIC32x4 codec port when
used in of_graph context.
This patch allows to share the codec port between two CPU DAIs.
Example:
Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec
is connected to two serial audio interfaces, which are configured
either as rx or tx.
>From AsoC point of view the topolgy is the following:
// 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4)
Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0
Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0
In the DT two endpoints have to be associated to the codec port:
tlv320aic32x4_port: port {
tlv320aic32x4_tx_endpoint: endpoint@0 {
remote-endpoint = <&sai2a_endpoint>;
};
tlv320aic32x4_rx_endpoint: endpoint@1 {
remote-endpoint = <&sai2b_endpoint>;
};
};
However, when the audio graph card parses the codec nodes, it expects
to find DAI interface indexes matching the endpoints indexes.
The current patch forces the use of DAI id 0 for both endpoints,
which allows to share the codec DAI between the two CPU DAIs
for playback and capture streams respectively.
Signed-off-by: Marek Vasut <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Return devm_of_clk_add_hw_provider() in order to transfer the error, if it
fails due to resource allocation failure or device tree clock provider
registration failure.
Cc: [email protected]
Fixes: ebbfabc16d23 ("ASoC: rt5682: Add CCF usage for providing I2S clks")
Signed-off-by: Ma Ke <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add a driver for the Audio Front End (AFE) PCM to provide Audio
Uplink (UL) and Downlink (DL) paths.
Use the ALSA SoC Dynamic Audio Power Management to add widget and
kcontrol supports.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add a specific soundcard for mt8365-evk. It supports audio jack
in/out, dmics, the amic and lineout.
Signed-off-by: Nicolas Belin <[email protected]>
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add Pulse Code Modulation Device Audio Interface support for MT8365 SoC.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add Digital Micro Device Audio Interface support for MT8365 SoC.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add ADDA Device Audio Interface support for MT8365 SoC.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add I2S Device Audio Interface support for MT8365 SoC.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add audio clock wrapper and audio tuner control.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add header files for register definition and structure.
Reviewed-by: AngeloGioacchino Del Regno <[email protected]>
Signed-off-by: Alexandre Mergnat <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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