IF YOU WOULD LIKE TO GET AN ACCOUNT, please write an
email to Administrator. User accounts are meant only to access repo
and report issues and/or generate pull requests.
This is a purpose-specific Git hosting for
BaseALT
projects. Thank you for your understanding!
Только зарегистрированные пользователи имеют доступ к сервису!
Для получения аккаунта, обратитесь к администратору.
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: e89f45edb747e ("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/20230605085839.2157268-1-arnd@kernel.org
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: e89f45edb747e ("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>
Add compatible string "mediatek,mt8188-nau8825" to support new board
with nau8825 codec.
Introduce two properties "dai-format" and "mediatek,clk-provider" under
dai-link subnode to configure dai-link parameters via dts.
"codec" property is removed from required property of dai-link subnode.
For co-clock case, it's possible two dai-links should be configured to
the same dai format, but only one cpu dai is bound with codec.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://lore.kernel.org/r/20230526093150.22923-8-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds multiple i2s codecs support including NAU88L25,
MAX98390, and the dumb amp like NAU8318 usage. In addition, dmic-codec
is also added to skip the beginning pop noise.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://lore.kernel.org/r/20230526093150.22923-7-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When a widget is added to dapm via snd_soc_dapm_new_widgets,
dapm_debugfs_add_widget is also called to create a corresponding debugfs
file. However, when a widget is freed by snd_soc_dapm_free_widget, the
corresponding debugfs is not cleared. As a result, the freed widget is
still seen in the dapm directory.
This patch adds dapm_debugfs_free_widget to free the debugfs of a
specified widget, and it's called at snd_soc_dapm_free_widget to clean
up the debugfs for freed widget.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Link: https://lore.kernel.org/r/20230526093150.22923-6-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two changes included in the patch.
First, add set_dailink_daifmt() function, so dai_fmt can be updated by
the configuration in dai-link sub node.
Second, remove codec phandle from required property in dai-link sub node.
For example, user possibly needs to update dai-format for all etdm
co-clock dai-links, but codec doesn't need to be specified in capture
dai-link for a speaker amp.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Link: https://lore.kernel.org/r/20230526093150.22923-5-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some userspace applications need jack control events, so register hdmi
and dp jack pins to activate jack control events.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Link: https://lore.kernel.org/r/20230526093150.22923-4-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ADDA_BE is used to connect to mt6359. For machine mt8188-mt6359, codec
for ADDA_BE must be mt6359 which are configured on the machine driver.
Besides, ADDA_BE is divided into two dais, UL_SRC_BE and DL_SRC_BE.
As a result, remove ADDA_BE from items of link-name.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Link: https://lore.kernel.org/r/20230526093150.22923-3-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
MT8188 will support SOF. In SOF, be_hw_params_fixup callback are used to
configure BE hardware parameters. However, playback and capture stream
share the same callback function in which it can't know the stream type.
It's possible to require different parameters for playback and capture
stream, so separate them into two dais for SOF usage.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Alexandre Mergnat <amergnat@baylibre.com>
Link: https://lore.kernel.org/r/20230526093150.22923-2-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There's an issue on SAI synchronous mode that TX/RX side can't get BCLK
from RX/TX it sync with if BYP bit is asserted. It's a workaround to
fix it that enable SION of IOMUX pad control and assert BCI.
For example if TX sync with RX which means both TX and RX are using clk
form RX and BYP=1. TX can get BCLK only if the following two conditions
are valid:
1. SION of RX BCLK IOMUX pad is set to 1
2. BCI of TX is set to 1
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://lore.kernel.org/r/20230530103012.3448838-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The code in asoc_simple_startup was treating any non-zero return from
snd_pcm_hw_constraint_minmax as an error, when this can return 1 in some
normal cases and only negative values indicate an error.
When this happened, it caused asoc_simple_startup to disable the clocks
it just enabled and return 1, which was not treated as an error by the
calling code which only checks for negative return values. Then when the
PCM is eventually shut down, it causes the clock framework to complain
about disabling clocks that were not enabled.
Fix the check for snd_pcm_hw_constraint_minmax return value to only
treat negative values as an error.
Fixes: 5ca2ab459817 ("ASoC: simple-card-utils: Add new system-clock-fixed flag")
Signed-off-by: Robert Hancock <robert.hancock@calian.com>
Link: https://lore.kernel.org/r/20230602011936.231931-1-robert.hancock@calian.com
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/
Merge series from Trevor Wu <trevor.wu@mediatek.com>:
These patches concern modifications made in mt8186[1]. The clock
unregistration mechanism used in mt8188 and mt8195 is similar with
mt8186, resulting in the same problem existing within the driver.
Therefore, the solution has also been applied to these two platforms.
[1] https://lore.kernel.org/all/20230511092437.1.I31cceffc8c45bb1af16eb613e197b3df92cdc19e@changeid/
The dma pointer must be set to the passed stream pointer, even
if that pointer is NULL.
Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230601124907.3128170-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
MAX98363 does not support 32bit depth audio.
Removed 32bit from the supported format list.
Instead, added 16bit and 24bit to the list.
Signed-off-by: Ryan Lee <ryans.lee@analog.com>
Link: https://lore.kernel.org/r/20230601130600.25344-1-ryan.lee.analog@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Change the message at the start of bin file loading from
cs_dsp_dbg() to cs_dsp_info() so that there is confirmation
in the kernel log that a bin file was loaded, and the name
of the file.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230531170158.2744700-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add bindings for TDM driver which supports multi-channel audio playback
and capture on JH7110 platform.
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Walker Chen <walker.chen@starfivetech.com>
Link: https://lore.kernel.org/r/20230526145402.450-2-walker.chen@starfivetech.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During mt8195_afe_init_clock(), mt8195_audsys_clk_register() was called
followed by several other devm functions. At mt8195_afe_deinit_clock()
located at mt8195_afe_pcm_dev_remove(), mt8195_audsys_clk_unregister()
was called.
However, there was an issue with the order in which these functions were
called. Specifically, the remove callback of platform_driver was called
before devres released the resource, resulting in a use-after-free issue
during remove time.
At probe time, the order of calls was:
1. mt8195_audsys_clk_register
2. afe_priv->clk = devm_kcalloc
3. afe_priv->clk[i] = devm_clk_get
At remove time, the order of calls was:
1. mt8195_audsys_clk_unregister
3. free afe_priv->clk[i]
2. free afe_priv->clk
To resolve the problem, we can utilize devm_add_action_or_reset() in
mt8195_audsys_clk_register() so that the remove order can be changed to
3->2->1.
Fixes: 6746cc858259 ("ASoC: mediatek: mt8195: add platform driver")
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Douglas Anderson <dianders@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230601033318.10408-3-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During mt8188_afe_init_clock(), mt8188_audsys_clk_register() was called
followed by several other devm functions. The caller of
mt8188_afe_init_clock() utilized devm_add_action_or_reset() to call
mt8188_afe_deinit_clock(). However, the order was incorrect, causing a
use-after-free issue during remove time.
At probe time, the order of calls was:
1. mt8188_audsys_clk_register
2. afe_priv->clk = devm_kcalloc
3. afe_priv->clk[i] = devm_clk_get
At remove time, the order of calls was:
1. mt8188_audsys_clk_unregister
3. free afe_priv->clk[i]
2. free afe_priv->clk
To resolve the problem, it's necessary to move devm_add_action_or_reset()
to the appropriate position so that the remove order can be 3->2->1.
Fixes: f6b026479b13 ("ASoC: mediatek: mt8188: support audio clock control")
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: Douglas Anderson <dianders@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230601033318.10408-2-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>
Thinkpad Neo14 Ryzen Edition uses Ryzen 6800H processor, and adding to
quirks list for acp6x will enable internal mic.
Signed-off-by: Sicong Jiang <kevin.jiangsc@gmail.com>
Link: https://lore.kernel.org/r/20230531090635.89565-1-kevin.jiangsc@gmail.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: b8b88b70875a ("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: b8b88b70875a ("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>
