Rt712 is a multi function codec which shpports headset, amp, and dmic
functions. Rt712 has two sdw interfaces and codec drivers, one for jack
and amp, the other for dmic. part id 0x712 is for jack and amp, and 0x1712
is for dmic.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Let rename rt711_sdca to rt_sdca_jack and let it be used for all
Realtek sdca jacks.
The commit uses component->name_prefix to construct card->components,
and determine which codec it is. So, we have to set name_prefix
properly.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Jack Detection source can be applied to all jacks, not only rt711.
No function changes.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
if (!SOF_RT711_JDSRC(sof_sdw_quirk)) is tested in rt711_sdca_add_codec_
device_props(), and we don't add software node to the device if jack
source is not set. We need to do the same test in
sof_sdw_rt711_sdca_exit(), and avoid removing software node if jack
source is not set.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A codec may support multiple dais for different purpose. For example,
the rt712 codec supports jack and amp on different dais and machine
driver needs to create different dailink for those dais.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We append codec type to dailink name to distinguish different dailink
on the same sdw link and direction. But we could create multi dailinks
for a codec and the dailink name will be duplicated if we append codec
type to the dailink name.
Appending dai type instead of codec type can solve the issue.
For example, if a codec supports JACK on dai 0 and AMP on dai 1, the
existing code will create dailinks
SDW0-Playback-SimpleJack or SDW0-Playback-SmartAmp for both dailinks,
and it will be SDW0-Playback-SimpleJack for dailink 0 and
SDW0-Playback-SmartAmp for dailink 1 after this change.
Then codec type is not used any more and can be removed.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
codec_info_list[codec_index] is used multiple times in the
create_sdw_dailink() function. Adding a codec_info pointer to shorten
the code. This is a preparation for the following up patches.
No function changed.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, we assign dailink ids in order, and shift with codec type.
The purpose is to have consistent dailink ids for topologies.
This can be simplified if we have a predefined dailink id in
sof_sdw_dai_info.
We reuse the existing ids as the predefine ids. So the dailink ids will
not be changed by this commit.
With this change, we no longer need to check the adr order described in a
snd_soc_acpi_link_adr array.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code create a dailink for a codec. However, we may need
multi dailinks for a codec. This commit adds a new struct in
sof_sdw_codec_info{} to store the dai info of a codec.
The initial assumption if that we will create at most 3 dailink types
for a codec, since this is the max known with upcoming SDCA devices. We
may need to increase this number as new SDCA 'functions' become available.
One strong assumption is that all dailinks exposed are independent, as per
SDCA directions.
This commit just moves some items into the new sof_sdw_dai_info struct.
There is no function changed. Multi dais supported will be added in the
follow up commits.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230602202225.249209-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The vangogh driver just gained a link time dependency that now causes
randconfig builds to fail:
x86_64-linux-ld: sound/soc/amd/vangogh/pci-acp5x.o: in function `snd_acp5x_probe':
pci-acp5x.c:(.text+0xbb): undefined reference to `snd_amd_acp_find_config'
Fixes: e89f45edb7 ("ASoC: amd: vangogh: Add check for acp config flags in vangogh platform")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20230602124447.863476-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Walker Chen <walker.chen@starfivetech.com>:
This patchset adds TDM audio driver for the StarFive JH7110 SoC. The
first patch adds device tree binding for TDM module. The second patch
adds tdm driver support for JH7110 SoC. The last patch adds device tree
node and pins configuration of tdm to JH7110 dts.
The series has been tested on the VisionFive 2 board by plugging an
audio expansion board.
For more information of audio expansion board, you can take a look
at the following webpage:
https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/
Add mt8188 and mt8186 .dbg_dump callback to print some information when
DSP panic occurs.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Yaochun Hung <yc.hung@mediatek.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230601034939.15802-2-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch is to manage HSD feature for power saving. The detail is to
disable HSD feature after the headset detection is done. When the jack
is inserted, the HSD feature will be enabled again.
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Link: https://lore.kernel.org/r/20230531075334.168637-1-CTLIN0@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_get_playback_capture() (A) returns number of substreams for
playback/capture, and then, we can use playback/capture_only flag (X)(Y).
(A) static int soc_get_playback_capture(...)
{
...
(X) if (dai_link->playback_only) {
(*) *playback = 1;
*capture = 0;
}
(Y) if (dai_link->capture_only) {
*playback = 0;
(*) *capture = 1;
}
...
}
But this flag should not have effect to opposite side stream (*).
This patch tidyup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87sfbezlq8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_get_playback_capture() (A) returns number of substreams for
playback/capture (B).
(A) static int soc_get_playback_capture(...,
(B) int *playback, int *capture)
{
...
for_each_xxx(...) {
if (xxx)
return -EINVAL;
=> *playback = 1;
...
=> *capture = 1;
...
}
...
}
But, it is directly updating playback/capture which is the result of this
function even though it might be error. It should be updated in case of
succeed only. This patch updates it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87ttvuzlqe.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_get_playback_capture() (A) checks dai_link status, and indicate error
if it was not matching (B).
(A) static int soc_get_playback_capture(...)
{
...
^ if (dai_link->dynamic && dai_link->num_cpus > 1) {
| dev_err(rtd->dev,
(B) "DPCM doesn't support Multi CPU for Front-Ends yet\n");
| return -EINVAL;
v }
...
}
We can use 100 char for 1 line today. This patch cleanup error code line.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87v8gazlqk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_get_playback_capture() (A) is using rtd->dai_link->xxx everywhere.
Because of that, 1 line is unnecessarily long and not readable.
(A) static int soc_get_playback_capture(...)
{
if (rtd->dai_link->dynamic ...) {
^^^^^^^^^^^^^
...
} else {
int cpu_capture = rtd->dai_link->c2c_params ?
^^^^^^^^^^^^^
...
}
if (rtd->dai_link->playback_only) {
^^^^^^^^^^^^^
...
}
...
}
This patch uses variable "dai_link" to be clear code.
Nothing changes the meanings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87wn0qzlqp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_get_playback_capture() (A) returns number of substreams for
playback/capture (B).
ASoC will probe the Sound Card and mapps CPU<->Codec pair.
(A) static int soc_get_playback_capture(...,
(B) int *playback, int *capture)
{
...
if (rtd->dai_link->playback_only) {
*playback = 1;
*capture = 0;
}
if (rtd->dai_link->capture_only) {
*playback = 0;
*capture = 1;
}
(C)
return 0;
}
But it might be no playback no capture if it returns playback=0, capture=0.
It is very difficult to notice about it. This patch indicates error at (C)
then.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87y1l6zlqx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>:
This patch series handles a few issues related to the ES8316 audio
codec, discovered while doing some testing on the Rock 5B board.
When using the codec through the generic audio graph card, there are at
least two calls of es8316_set_dai_sysclk(), with the effect of limiting
the allowed sample rates according to the MCLK/LRCK ratios supported by
the codec:
1. During audio card setup, to set the initial MCLK - see
asoc_simple_init_dai().
2. Before opening a stream, to update MCLK, according to the stream
sample rate and the multiplication factor - see
asoc_simple_hw_params().
In some cases the initial MCLK might be set to a frequency that doesn't
match any of the supported ratios, e.g. 12287999 instead of 12288000,
which is only 1 Hz below the supported clock, as that is what the
hardware reports. This creates an empty list of rate constraints, which
is further passed to snd_pcm_hw_constraint_list() via
es8316_pcm_startup(), and causes the following error on the very first
access of the sound card:
$ speaker-test -D hw:Analog,0 -F S16_LE -c 2 -t wav
Broken configuration for playback: no configurations available: Invalid argument
Setting of hwparams failed: Invalid argument
Note that all subsequent retries succeed thanks to the updated MCLK set
at point 2 above, which uses a computed frequency value instead of a
reading from the hardware registers. Normally this would have mitigated
the issue, but es8316_pcm_startup() executes before the 2nd call to
es8316_set_dai_sysclk(), hence it cannot make use of the updated
constraints.
Since es8316_pcm_hw_params() performs anyway a final validation of MCLK
against the stream sample rate and the supported MCLK/LRCK ratios, fix
the issue by ensuring that sysclk_constraints list is only set when at
least one supported sample rate is autodetected by the codec.
Fixes: b8b88b7087 ("ASoC: add es8316 codec driver")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20230530181140.483936-3-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The following error occurs when trying to restore a previously saved
ALSA mixer state (tested on a Rock 5B board):
$ alsactl --no-ucm -f /tmp/asound.state store hw:Analog
$ alsactl --no-ucm -I -f /tmp/asound.state restore hw:Analog
alsactl: set_control:1475: Cannot write control '2:0:0:ALC Capture Target Volume:0' : Invalid argument
According to ES8316 datasheet, the register at address 0x2B, which is
related to the above mixer control, contains by default the value 0xB0.
Considering the corresponding ALC target bits (ALCLVL) are 7:4, the
control is initialized with 11, which is one step above the maximum
value allowed by the driver:
ALCLVL | dB gain
-------+--------
0000 | -16.5
0001 | -15.0
0010 | -13.5
.... | .....
0111 | -6.0
1000 | -4.5
1001 | -3.0
1010 | -1.5
.... | .....
1111 | -1.5
The tests performed using the VU meter feature (--vumeter=TYPE) of
arecord/aplay confirm the specs are correct and there is no measured
gain if the 1011-1111 range would have been mapped to 0 dB:
dB gain | VU meter %
--------+-----------
-6.0 | 30-31
-4.5 | 35-36
-3.0 | 42-43
-1.5 | 50-51
0.0 | 50-51
Increment the max value allowed for ALC Capture Target Volume control,
so that it matches the hardware default. Additionally, update the
related TLV to prevent an artificial extension of the dB gain range.
Fixes: b8b88b7087 ("ASoC: add es8316 codec driver")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://lore.kernel.org/r/20230530181140.483936-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
The topology file and the machine driver rely on common definitions
for the dailink stream_name. To avoid any backwards-compatibility
problems, the machine driver stream names are set in stone and cannot
be modified.
This is problematic when we try to name some of the topology widgets
after the stream_name, since the widget name is limited to 44
characters
tools/include/uapi/sound/asound.h:#define SNDRV_CTL_ELEM_ID_NAME_MAXLEN 44
Existing examples include "Analog Playback and Capture" for HDaudio
dailinks, which leaves less than 20 chars to identify widgets/controls
with a meaningful name.
Since the 44-char limit is part of the UAPI definitions, we assumed
there is no way to increase it.
This patchset suggests instead a partial match which allows topology
files to use a shorter stream_name, which in turn allows for
self-explanatory widget names that comply with the 44-char limit.
This should not break any existing setup but with the introduction of
a partial match new dailinks should be named carefully to avoid
confusions between e.g. 'link1' and 'link10'. The last patch fixes
such an issue in the 'nocodec' test topology used by Intel.
With a common kernel config for nocodec and codec modes, the number of DAI
drivers will be set to 15 for nocodec as well. So adjust this when set
the machine params for the nocodec mode if the debug flag is set.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20230526204149.456068-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This allows setting shorter names for the widget stream names in
topology. For example, in the case of HDA Analog DAI link, the stream
name is "Analog Playback and Capture". But it is enough to match "Analog"
in the DAI link stream name with a widget's stream name. This is needed
to set more meaningful names for the DAI widgets using the stream name
in topology.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20230526204149.456068-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This allows for setting shorter link names in topology. For example,
for the HDA Analog DAI link, just "Analog" would suffice instead of
"Analog Playback and Capture"
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230526204149.456068-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use pcm_for_each_format for the PCM format iteration and fix the
following sparse warnings.
sound/soc/ti/davinci-mcasp.c:1336:26: sparse: warning: restricted snd_pcm_format_t degrades to integer
sound/soc/ti/davinci-mcasp.c:1358:26: sparse: warning: restricted snd_pcm_format_t degrades to integer
sound/soc/ti/davinci-mcasp.c:1438:26: sparse: warning: restricted snd_pcm_format_t degrades to integer
No functional changes.
Signed-off-by: Min-Hua Chen <minhuadotchen@gmail.com>
Link: https://lore.kernel.org/r/20230519211636.3699-1-minhuadotchen@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have SOF and generic ACP support enabled for Vangogh platform
on some machines. Since we have same PCI id used for probing,
add check for machine configuration flag to avoid conflict with
newer pci drivers. Such machine flag has been initialized via
dmi match on few Vangogh based machines. If no flag is
specified probe and register older platform device.
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://lore.kernel.org/r/20230530110802.674939-1-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add mt8188 dai driver and specify of_machine to support mt8188 audio.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Yaochun Hung <yc.hung@mediatek.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230523025933.30494-2-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add 4.8Mhz 9.6Mhz and 19.2MHz SCLK values
for MCLK 12MHz and 12.288MHz requested by Intel.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230524125236.57149-1-vitalyr@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from David Rau <David.Rau.opensource@dm.renesas.com>:
This patchset adds the support of DA7219 Jack insertion detection
polarity selection.
The first patch replaces the old txt binding with a new schema binding.
The second patch adds `dlg,jack-ins-det-pty` property for Jack
insertion detection polarity selection.
The last patch adds the driver support for this topic.
The series has been verified on the DA7219 development kit.
Merge series from Daniel Baluta <daniel.baluta@oss.nxp.com>:
This patch series provides better handling of cases where sending
a data blob to FW results in a validation error.
In this case we restore to the last good known value instead of keeping
the data that firwmare rejected.
Merge series from Trevor Wu <trevor.wu@mediatek.com>:
The patch series includes a kselftest fix and changes for extending
driver capability to support more use cases.
Merge series from Claudiu Beznea <claudiu.beznea@microchip.com>:
Series removes the pm_runtime.h inclusion in files where
APIs exported though pm_runtime.h are not used. In case
of files that make use of pm.h which comes form pm_runtime.h
added patch 2/2.
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The following series will enable multicore support on MTL platforms similarly
to other Intel platforms.
The TGL patch is included to simplify the core_put implementation.
Multicore support can be enabled by updated topologies, with current set of
tplg files this series is not introducing any runtime change.
Originally, lineout playback source can only be DAC_3RD. Some SoC
masters only support stereo MTKAIF outputs, so lineout path can't be
used in such case.
MTKAIF connections are as follows.
MOSI0 -> DAC_L
MOSI1 -> DAC_R
MOSI2 -> DAC_3rd
In the patch, lineout playback source can be chosen between DAC_L and
DAC_3rd, so sound can be outputted via lineout even though SoC only
supports stereo MTKAIF outputs.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://lore.kernel.org/r/20230508071532.21665-5-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
kselftest tries to read/write the default value. The default register
value of playback gain is 0x1F(mute), but max gain we specified is 0x12.
The range of the control is 0x0~0x12 and mute(0x1F) is only used in the
driver internally. To solve the problem, implement a new callback
mt6359_get_playback_volsw to report user configured volume instead of
the register value.
In addition, update max of "Headset Volume" to 0x12, so it can match the
maximum seen on latest data sheet.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://lore.kernel.org/r/20230508071532.21665-3-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There are three output data pins MISO0, MISO1 and MISO2 for mt6359.
UL_SRC should be enabled when MISO0 or MISO1 is used, and UL_SRC_34
should be enabled when MISO2 is used.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://lore.kernel.org/r/20230508071532.21665-2-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The function is improved in the way that if the firmware returns a
validation error on the newly sent bytes, then the kernel will
automatically restore to the old bytes value for a given kcontrol.
This way, if the firmware rejects a data blob then the kernel will also
reject it, instead of saving it for the next suspend/resume cycle. The
old behaviour is that the kernel would save it anyway and on next
firmware boot it would apply the previously-rejected configuration,
leading to errors during playback.
Additionally, the function also saves previously validated
configurations, so that if the firmware does end up rejecting a new
bytes value the kernel can send an old, previously-valid configuration.
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Paul Olaru <paul.olaru@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20230503081049.73847-3-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
