| Age | Commit message (Collapse) | Author | Files | Lines |
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Optimize the memory usage in struct snd_pcm_runtime - use boolean
value for the standard sync ID scheme.
Introduce snd_pcm_set_sync_per_card function to build synchronization
IDs.
Signed-off-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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Until the commit e11f0f90a626 ("ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO
internal command"), there was a possibility to pass information
about the synchronized streams to the user space. The mentioned
commit removed blindly the appropriate code with an irrelevant comment.
The revert may be appropriate, but since this API was lost for several
years without any complains, it's time to improve it. The hardware
parameters may change the used stream clock source (e.g. USB hardware)
so move this synchronization ID to hw_params as read-only field.
It seems that pipewire can benefit from this API (disable adaptive
resampling for perfectly synchronized PCM streams) now.
Note that the contents of ID is not supposed to be used for direct
comparison with a specific byte sequence. The "empty" case is when
all bytes are zero (driver does not offer this information)
and all other cases must be only used for equal comparison among
PCM streams (including different sound cards) if they are using
identical hardware clock.
Cc: Takashi Sakamoto <[email protected]>
Signed-off-by: Jaroslav Kysela <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
Link: https://patch.msgid.link/[email protected]
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Apply quirk to fix combo jack detection on a new Clevo model: V5[46]0TU
Signed-off-by: Michał Kopeć <[email protected]>
Cc: <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Takashi Iwai <[email protected]>
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The 'wsa->dev' is assigned closer to the end of the probe() function, so
the dev_err() must not use it - it is still NULL at this point. Instead
there is already a local 'dev' variable.
Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 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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The 'rx->dev' is assigned closer to the end of the probe() function, so
the dev_err() must not use it - it is still NULL at this point. Instead
there is already a local 'dev' variable.
Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions")
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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bitfield.h is not explicitly included but it is required for FIELD_PREP
to be expanded by the preprocessor. If it is not implicitly included,
there will be a compiler error (as seen with ARCH=hexagon allmodconfig):
sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP);
| ^
sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8'
42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH'
65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH'
70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */
| ^
Include bitfield.h explicitly, so that FIELD_PREP is always expanded,
clearing up the compiler error.
Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <[email protected]>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-include-for-field_prep-v1-1-0c5d7f71921b@kernel.org
Signed-off-by: Mark Brown <[email protected]>
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clang points out that ret may be used uninitialized in
lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with
CONFIG_WERROR):
sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized]
326 | "failed to init register map: %d\n", ret);
| ^~~
sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning
310 | int ret;
| ^
| = 0
1 error generated.
One solution would be a small refactoring of the second parameter in
dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in
the if statement. However, a nicer solution for debugging purposes,
which is the point of this statement, would be to use the '%pe'
specifier to symbolically print the error pointer value. Do so, which
eliminates the uninitialized use of ret, clearing up the warning.
Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <[email protected]>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-uninitialized-ret-v1-1-985d86189739@kernel.org
Signed-off-by: Mark Brown <[email protected]>
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Convert the RT5677 audio CODEC bindings to DT schema.
Signed-off-by: Animesh Agarwal <[email protected]>
Cc: Daniel Baluta <[email protected]>
Reviewed-by: Rob Herring (Arm) <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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calibrated data files
Add name_prefix as the prefix name of DSP firmwares
and calibrated data files which stored speaker
calibrated impedance.
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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Add support to parse static master port map information from device tree.
This is required for correct port mapping between soundwire device and
master ports.
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Tested-by: Krzysztof Kozlowski <[email protected]>
Reviewed-by: Neil Armstrong <[email protected]>
Tested-by: Neil Armstrong <[email protected]> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <[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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Document port mapping property for wsa884x. Port mapping is required
to be able map correct master ports.
All the device ports are not mapped in same order as master ports, so
there is a need for having static port mapping for WSA codecs.
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Tested-by: Neil Armstrong <[email protected]> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add support to parse static master port map information from device tree.
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Tested-by: Krzysztof Kozlowski <[email protected]>
Tested-by: Neil Armstrong <[email protected]> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <[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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Document port mapping property for wsa883x. Port mapping is required
to be able map correct master ports for VI feedback.
All the device ports are not mapped in same order as master ports, so
there is a need for having static port mapping for WSA codecs.
Signed-off-by: Manikantan R <[email protected]>
Reviewed-by: Krzysztof Kozlowski <[email protected]>
Tested-by: Neil Armstrong <[email protected]> # on SM8650-HDK
Signed-off-by: Srinivas Kandagatla <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would
work, but might be not user friendly in some case, for example in case
of system hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state
names.
Signed-off-by: Kuninori Morimoto <[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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simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would
work, but might be not user friendly in some case, for example in case
of system hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state
names.
Original
> amixer set "MUX" "Input 1"
Use mux-names
sound_mux: mux {
compatible = "simple-audio-mux";
mux-gpios = <...>;
state-labels = "Label_A", "Label_B";
};
> amixer set "MUX" "Label_A"
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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Check that all fields of a V2 algorithm header fit into the available
firmware data buffer.
The wmfw V2 format introduced variable-length strings in the algorithm
block header. This means the overall header length is variable, and the
position of most fields varies depending on the length of the string
fields. Each field must be checked to ensure that it does not overflow
the firmware data buffer.
As this ia bugfix patch, the fixes avoid making any significant change to
the existing code. This makes it easier to review and less likely to
introduce new bugs.
Signed-off-by: Richard Fitzgerald <[email protected]>
Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs")
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Move the payload length check in cs_dsp_load() and cs_dsp_coeff_load()
to be done before the block is processed.
The check that the length of a block payload does not exceed the number
of remaining bytes in the firwmware file buffer was being done near the
end of the loop iteration. However, some code before that check used the
length field without validating it.
Signed-off-by: Richard Fitzgerald <[email protected]>
Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs")
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Return an error from cs_dsp_power_up() if a block header is longer
than the amount of data left in the file.
The previous code in cs_dsp_load() and cs_dsp_load_coeff() would loop
while there was enough data left in the file for a valid region. This
protected against overrunning the end of the file data, but it didn't
abort the file processing with an error.
Signed-off-by: Richard Fitzgerald <[email protected]>
Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs")
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Fix the checking that firmware file buffer is large enough for the
wmfw header, to prevent overrunning the buffer.
The original code tested that the firmware data buffer contained
enough bytes for the sums of the size of the structs
wmfw_header + wmfw_adsp1_sizes + wmfw_footer
But wmfw_adsp1_sizes is only used on ADSP1 firmware. For ADSP2 and
Halo Core the equivalent struct is wmfw_adsp2_sizes, which is
4 bytes longer. So the length check didn't guarantee that there
are enough bytes in the firmware buffer for a header with
wmfw_adsp2_sizes.
This patch splits the length check into three separate parts. Each
of the wmfw_header, wmfw_adsp?_sizes and wmfw_footer are checked
separately before they are used.
Signed-off-by: Richard Fitzgerald <[email protected]>
Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs")
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Convert the RT5659/RT5658 audio CODEC bindings to DT schema.
Signed-off-by: Animesh Agarwal <[email protected]>
Cc: Daniel Baluta <[email protected]>
Reviewed-by: Rob Herring (Arm) <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add compatible string and specific soc data to support rpmsg sound card
on i.MX95 platform.
Signed-off-by: Chancel Liu <[email protected]>
Acked-by: Shengjiu Wang <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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Add compatible string for i.MX95 platform which supports audio
function through rpmsg channel between Cortex-A and Cortex-M core.
Signed-off-by: Chancel Liu <[email protected]>
Acked-by: Rob Herring (Arm) <[email protected]>
Link: https://patch.msgid.link/[email protected]
Signed-off-by: Mark Brown <[email protected]>
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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]>
|
|
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]>
|
|
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]>
|