When pcm_runtime is adding platform components it will scan all
registered components. In case of DPCM FE/BE some DAI links will
configure dummy platform. However both dummy codec and dummy platform
are using "snd-soc-dummy" as component->name. Dummy codec should be
skipped when adding platforms otherwise there'll be overflow and UBSAN
complains.
Reported-by: Zhipeng Wang <zhipeng.wang_1@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240305065606.3778642-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name
for dlc (snd_soc_dai_link_component). In this function call
dlc->dai_name is parsed via snd_soc_dai_name_get() (B).
(A) int snd_soc_get_dlc(...)
{
...
(B) dlc->dai_name = snd_soc_dai_name_get(dai);
...
}
(B) has a priority to return dai->name as dlc->dai_name. In most cases
card can probe successfully. However it has an issue that ASoC tries to
rebind card. Here is a simplified flow for example:
| a) Card probes successfully at first
| b) One of the component bound to this card is removed for some
| reason the component->dev is released
| c) That component is re-registered
v d) ASoC calls snd_soc_try_rebind_card()
a) points dlc->dai_name to dai->name. b) releases all resource of the
old DAI. c) creates new DAI structure. In result d) can not use
dlc->dai_name to add new created DAI.
So it's reasonable that prefer to return dai->driver->name in
snd_soc_dai_name_get() because dai->driver is a pre-defined global
variable. Also update snd_soc_is_matching_dai() for alignment.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240304072128.2845432-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 861b3415e4 upstream.
This reverts commit ed00a6945d,
which added a quirk entry to enable the Yellow Carp (YC)
driver for the Lenovo 21J2 laptop.
Although the microphone functioned with the YC driver, it
resulted in incorrect driver usage. The Lenovo 21J2 is not a
Yellow Carp platform, but a Pink Sardine platform, which
already has an upstreamed driver.
The microphone on the Lenovo 21J2 operates correctly with the
CONFIG_SND_SOC_AMD_PS flag enabled and does not require the
quirk entry. So this patch removes the quirk entry.
Thanks to Mukunda Vijendar [1] for pointing this out.
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Link: https://msgid.link/r/20240313015853.3573242-2-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: Luca Stefani <luca.stefani.ge1@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Since the i.MX drivers no longer use the imx8_*_clocks API
this can be removed.
Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Currently, the driver has to keep track of all the clocks
it uses via an array of "struct clk_bulk_data", which doesn't
scale well and is unnecessary. As such, replace the usage of
the imx8_*_clocks with "devm_clk_bulk_get_all()" and friends.
Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
According to eARC spec when CMDC status updated, which means
the HDMI_HPD of TX is changed, then eARC RX need to re-read
the EDID info.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Sandor Yu <sandor.yu@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
Enable interrupt of cmdc status update
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Sandor Yu <sandor.yu@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
Add constraint for rpmsg micfil sound card, only 16kHz is
supported.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
[ Upstream commit 9e2ab4b18e ]
The sample rates set by the rockchip_i2s_tdm driver in master mode are
inaccurate up to 5% in several cases, due to the driver logic to configure
clocks and a nasty interaction with the Common Clock Framework.
To understand what happens, here is the relevant section of the clock tree
(slightly simplified), along with the names used in the driver:
vpll0 _OR_ vpll1 "mclk_root"
clk_i2s2_8ch_tx_src "mclk_parent"
clk_i2s2_8ch_tx_mux
clk_i2s2_8ch_tx "mclk" or "mclk_tx"
This is what happens when playing back e.g. at 192 kHz using
audio-graph-card (when recording the same applies, only s/tx/rx/):
0. at probe, rockchip_i2s_tdm_set_sysclk() stores the passed frequency in
i2s_tdm->mclk_tx_freq (*) which is 50176000, and that is never modified
afterwards
1. when playback is started, rockchip_i2s_tdm_hw_params() is called and
does the following two calls
2. rockchip_i2s_tdm_calibrate_mclk():
2a. selects mclk_root0 (vpll0) as a parent for mclk_parent
(mclk_tx_src), which is OK because the vpll0 rate is a good for
192000 (and sumbultiple) rates
2b. sets the mclk_root frequency based on ppm calibration computations
2c. sets mclk_tx_src to 49152000 (= 256 * 192000), which is also OK as
it is a multiple of the required bit clock
3. rockchip_i2s_tdm_set_mclk()
3a. calls clk_set_rate() to set the rate of mclk_tx (clk_i2s2_8ch_tx)
to the value of i2s_tdm->mclk_tx_freq (*), i.e. 50176000 which is
not a multiple of the sampling frequency -- this is not OK
3a1. clk_set_rate() reacts by reparenting clk_i2s2_8ch_tx_src to
vpll1 -- this is not OK because the default vpll1 rate can be
divided to get 44.1 kHz and related rates, not 192 kHz
The result is that the driver does a lot of ad-hoc decisions about clocks
and ends up in using the wrong parent at an unoptimal rate.
Step 0 is one part of the problem: unless the card driver calls set_sysclk
at each stream start, whatever rate is set in mclk_tx_freq during boot will
be taken and used until reboot. Moreover the driver does not care if its
value is not a multiple of any audio frequency.
Another part of the problem is that the whole reparenting and clock rate
setting logic is conflicting with the CCF algorithms to achieve largely the
same goal: selecting the best parent and setting the closest clock
rate. And it turns out that only calling once clk_set_rate() on
clk_i2s2_8ch_tx picks the correct vpll and sets the correct rate.
The fix is based on removing the custom logic in the driver to select the
parent and set the various clocks, and just let the Clock Framework do it
all. As a side effect, the set_sysclk() op becomes useless because we now
let the CCF compute the appropriate value for the sampling rate. It also
implies that the whole calibration logic is now dead code and so it is
removed along with the "PCM Clock Compensation in PPM" kcontrol, which has
always been broken anyway. The handling of the 4 optional clocks also
becomes dead code and is removed.
The actual rates have been tested playing 30 seconds of audio at various
sampling rates before and after this change using sox:
time play -r <sample_rate> -n synth 30 sine 950 gain -3
The time reported in the table below is the 'real' value reported by the
'time' command in the above command line.
rate before after
--------- ------ ------
8000 Hz 30.60s 30.63s
11025 Hz 30.45s 30.51s
16000 Hz 30.47s 30.50s
22050 Hz 30.78s 30.41s
32000 Hz 31.02s 30.43s
44100 Hz 30.78s 30.41s
48000 Hz 29.81s 30.45s
88200 Hz 30.78s 30.41s
96000 Hz 29.79s 30.42s
176400 Hz 27.40s 30.41s
192000 Hz 29.79s 30.42s
While the tests are running the clock tree confirms that:
* without the patch, vpll1 is always used and clk_i2s2_8ch_tx always
produces 50176000 Hz, which cannot be divided for most audio rates
except the slowest ones, generating inaccurate rates
* with the patch:
- for 192000 Hz vpll0 is used
- for 176400 Hz vpll1 is used
- clk_i2s2_8ch_tx always produces (256 * <rate>) Hz
Tested on the RK3308 using the internal audio codec.
Fixes: 081068fd64 ("ASoC: rockchip: add support for i2s-tdm controller")
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-1-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f31e0d0c2c ]
Using __exit for the remove function results in the remove callback
being discarded with SND_SOC_TLV320ADC3XXX=y. When such a device gets
unbound (e.g. using sysfs or hotplug), the driver is just removed
without the cleanup being performed. This results in resource leaks. Fix
it by compiling in the remove callback unconditionally.
This also fixes a W=1 modpost warning:
WARNING: modpost: sound/soc/codecs/snd-soc-tlv320adc3xxx: section mismatch in reference: adc3xxx_i2c_driver+0x10 (section: .data) -> adc3xxx_i2c_remove (section: .exit.text)
(which only happens with SND_SOC_TLV320ADC3XXX=m).
Fixes: e9a3b57efd ("ASoC: codec: tlv320adc3xxx: New codec driver")
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://msgid.link/r/20240310143852.397212-2-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 76f5f55c45 ]
Make calibration functions configurable to support different calibration
data storage modes.
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/5859c77ffef752b8a9784713b412d815d7e2688c.1703891777.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 5f51de7e30 ("ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 59c6a3a43b ]
According to Amlogic datasheets for the SoCs supported by this driver, the
maximum bit clock rate is 100MHz.
The tdm interface allows the rates listed by the DAI driver, regardless of
the number slots or their width. However, these will impact the bit clock
rate.
Hitting the 100MHz limit is very unlikely for most use cases but it is
possible.
For example with 32 slots / 32 bits wide, the maximum rate is no longer
384kHz but ~96kHz.
Add the constraint accordingly if the component is not already active.
If it is active, the rate is already constrained by the first stream rate.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e3741a8d28 ]
By default, when mclk-fs is not provided, the tdm-interface driver
requests an MCLK that is 4x the bit clock, SCLK.
However there is no justification for this:
* If the codec needs MCLK for its operation, mclk-fs is expected to be set
according to the codec requirements.
* If the codec does not need MCLK the minimum is 2 * SCLK, because this is
minimum the divider between SCLK and MCLK can do.
Multiplying by 4 may cause problems because the PLL limit may be reached
sooner than it should, so use 2x instead.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 98f681b0f8 ]
Smatch complains about "head->full_size - head->header_size" can
underflow. To some extent, we're always going to have to trust the
firmware a bit. However, it's easy enough to add a check for negatives,
and let's add a upper bounds check as well.
Fixes: d2458baa79 ("ASoC: SOF: ipc3-loader: Implement firmware parsing and loading")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://msgid.link/r/5593d147-058c-4de3-a6f5-540ecb96f6f8@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5ad992c71b ]
clang-16 warns about casting functions to incompatible types, as is done
here to call clk_disable_unprepare:
sound/soc/meson/t9015.c:274:4: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict]
274 | (void(*)(void *))clk_disable_unprepare,
The pattern of getting, enabling and setting a disable callback for a
clock can be replaced with devm_clk_get_enabled(), which also fixes
this warning.
Fixes: 33901f5b9b ("ASoC: meson: add t9015 internal DAC driver")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Justin Stitt <justinstitt@google.com>
Link: https://msgid.link/r/20240213215807.3326688-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 98ac85a00f ]
clang-16 warns about casting functions to incompatible types, as is done
here to call clk_disable_unprepare:
sound/soc/meson/aiu.c:243:12: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict]
243 | (void(*)(void *))clk_disable_unprepare,
The pattern of getting, enabling and setting a disable callback for a
clock can be replaced with devm_clk_get_enabled(), which also fixes
this warning.
Fixes: 6ae9ca9ce9 ("ASoC: meson: aiu: add i2s and spdif support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Justin Stitt <justinstitt@google.com>
Link: https://msgid.link/r/20240213215807.3326688-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9a6d7c4fb2 ]
The devm_request_irq() call is done for "dma_rt" interrupt but the error
message printed "dma_tx" interrupt on failure, fix this by updating
dma_tx -> dma_rt in dev_err_probe() message. While at it aligned the code.
Signed-off-by: Lad Prabhakar <prabhakar.mahadev-lad.rj@bp.renesas.com>
Fixes: 38c042b59a ("ASoC: sh: rz-ssi: Update interrupt handling for half duplex channels")
Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://msgid.link/r/20240130150822.327434-1-prabhakar.mahadev-lad.rj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 222be59e5e ]
Driver uses kasprintf() to initialize fw_{code,data}_bin members of
struct acp_dev_data, but kfree() is never called to deallocate the
memory, which results in a memory leak.
Fix the issue by switching to devm_kasprintf(). Additionally, ensure the
allocation was successful by checking the pointer validity.
Fixes: f7da88003c ("ASoC: SOF: amd: Enable signed firmware image loading for Vangogh platform")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Emil Velikov <emil.velikov@collabora.com>
Link: https://msgid.link/r/20231219030728.2431640-6-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d0ada20279 ]
Handle potential acp_sofdsp_dai_links_create() errors in ACP SOF machine
driver's probe function. Note there is no need for an undo.
While at it, switch to dev_err_probe().
Fixes: 9f84940f50 ("ASoC: amd: acp: Add SOF audio support on Chrome board")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Emil Velikov <emil.velikov@collabora.com>
Link: https://msgid.link/r/20231219030728.2431640-4-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 96e202f8c5 ]
Use source instead of ret, which seems to be unrelated and will always
be zero.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-5-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b3a5113760 ]
The HP Pavilion Aero Laptop 13-be2xxx(8BD6) requires a quirk entry for its internal microphone to function.
Signed-off-by: Al Raj Hassain <alrajhassain@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://msgid.link/r/20240304103924.13673-1-alrajhassain@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f8b0127aca ]
The bios version can differ depending if it is a dual-boot variant of the tablet.
Therefore another DMI match is required.
Signed-off-by: Alban Boyé <alban.boye@protonmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ed00a6945d ]
Like many other models, the Lenovo 21J2 (ThinkBook 16 G5+ APO)
needs a quirk entry for the internal microphone to function.
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240228073914.232204-2-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c40aad7c81 ]
When the system is suspended while audio is active, the
sof_ipc4_pcm_hw_free() is invoked to reset the pipelines since during
suspend the DSP is turned off, streams will be re-started after resume.
If the firmware crashes during while audio is running (or when we reset
the stream before suspend) then the sof_ipc4_set_multi_pipeline_state()
will fail with IPC error and the state change is interrupted.
This will cause misalignment between the kernel and firmware state on next
DSP boot resulting errors returned by firmware for IPC messages, eventually
failing the audio resume.
On stream close the errors are ignored so the kernel state will be
corrected on the next DSP boot, so the second boot after the DSP panic.
If sof_ipc4_trigger_pipelines() is called from sof_ipc4_pcm_hw_free() then
state parameter is SOF_IPC4_PIPE_RESET and only in this case.
Treat a forced pipeline reset similarly to how we treat a pcm_free by
ignoring error on state sending to allow the kernel's state to be
consistent with the state the firmware will have after the next boot.
Link: https://github.com/thesofproject/sof/issues/8721
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240213115233.15716-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f7fe85b229 ]
Like many other models, the Lenovo 82UU (Yoga Slim 7 Pro 14ARH7)
needs a quirk entry for the internal microphone to function.
Signed-off-by: Attila Tőkés <attitokes@gmail.com>
Link: https://msgid.link/r/20240210193638.144028-1-attitokes@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d172205747 ]
As devm_pm_runtime_enable can fail due to memory allocations, it is
best to handle the error.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240206113850.719888-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4703b014f2 upstream.
It looks like the "!" character was added accidentally. The
regmap_update_bits_check() function is normally going to succeed. This
means the rest of the function is unreachable and we don't handle the
situation where "changed" is true correctly.
Fixes: 07f7d6e7a1 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/0c254c07-d1c0-4a5c-a22b-7e135cab032c@moroto.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit eba2eb2495 ]
snd_soc_card_get_kcontrol() must be holding a read lock on
card->controls_rwsem while walking the controls list.
Compare with snd_ctl_find_numid().
The existing function is renamed snd_soc_card_get_kcontrol_locked()
so that it can be called from contexts that are already holding
card->controls_rwsem (for example, control get/put functions).
There are few direct or indirect callers of
snd_soc_card_get_kcontrol(), and most are safe. Three require
changes, which have been included in this patch:
codecs/cs35l45.c:
cs35l45_activate_ctl() is called from a control put() function so
is changed to call snd_soc_card_get_kcontrol_locked().
codecs/cs35l56.c:
cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from
control get()/put() functions so is changed to call
snd_soc_card_get_kcontrol_locked().
fsl/fsl_xcvr.c:
fsl_xcvr_activate_ctl() is called from three places, one of which
already holds card->controls_rwsem:
1. fsl_xcvr_mode_put(), a control put function, which will
already be holding card->controls_rwsem.
2. fsl_xcvr_startup(), a DAI startup function.
3. fsl_xcvr_shutdown(), a DAI shutdown function.
To fix this, fsl_xcvr_activate_ctl() has been changed to call
snd_soc_card_get_kcontrol_locked() so that it is safe to call
directly from fsl_xcvr_mode_put().
The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been
changed to take a read lock on card->controls_rsem() around calls
to fsl_xcvr_activate_ctl(). While this is not very elegant, it
keeps the change small, to avoid this patch creating a large
collateral churn in fsl/fsl_xcvr.c.
Analysis of other callers of snd_soc_card_get_kcontrol() is that
they do not need any changes, they are not holding card->controls_rwsem
when they call snd_soc_card_get_kcontrol().
Direct callers of snd_soc_card_get_kcontrol():
fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function
fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler
Indirect callers via soc_component_notify_control():
codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler
codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler
codecs/ak4118.c: ak4118_irq_handler() - IRQ handler
codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used
Indirect callers via snd_soc_limit_volume():
qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function
ti/rx51.c: rx51_aic34_init() - DAI init function
I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c
and ak4118.c changes.
Backport note:
The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added
since the Fixes commit so won't all be present on older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 209c6cdfd2 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card")
Link: https://lore.kernel.org/r/20240221123710.690224-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c14f09f010 ]
Rewrite the handling of ASP1 TX mixer mux initialization to prevent a
deadlock during component_remove().
The firmware can overwrite the ASP1 TX mixer registers with
system-specific settings. This is mainly for hardware that uses the
ASP as a chip-to-chip link controlled by the firmware. Because of this
the driver cannot know the starting state of the ASP1 mixer muxes until
the firmware has been downloaded and rebooted.
The original workaround for this was to queue a work function from the
dsp_work() job. This work then read the register values (populating the
regmap cache the first time around) and then called
snd_soc_dapm_mux_update_power(). The problem with this is that it was
ultimately triggered by cs35l56_component_probe() queueing dsp_work,
which meant that it would be running in parallel with the rest of the
ASoC component and card initialization. To prevent accessing DAPM before
it was fully initialized the work function took the card mutex. But this
would deadlock if cs35l56_component_remove() was called before the work job
had completed, because ASoC calls component_remove() with the card mutex
held.
This new version removes the work function. Instead the regmap cache and
DAPM mux widgets are initialized the first time any of the associated ALSA
controls is read or written.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 07f7d6e7a1 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Link: https://lore.kernel.org/r/20240208123742.1278104-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f6c967941c ]
Put the silicon revision and secured flag in the wm_adsp fwf_name
string instead of including them in the part string.
This changes the format of the firmware name string from
cs35l56[s]-rev-misc[-system_name]
to
cs35l56-rev[-s]-misc[-system_name]
No firmware files have been published, so this doesn't cause a
compatibility break.
Silicon revision and secured flag are included in the firmware
filename to pick a firmware compatible with the part. These strings
were being added to the part string, but that is a misuse of the
string. The correct place for these is the fwf_name string, which
is specifically intended to select between multiple firmware files
for the same part.
Backport note:
This won't apply to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 608f1b0dbd ("ASoC: cs35l56: Move DSP part string generation so that it is done only once")
Link: https://msgid.link/r/20240129162737.497-12-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 07f7d6e7a1 ]
Defer initializing the state of the ASP1 mixer registers until
the firmware has been downloaded and rebooted.
On a SoundWire system the ASP is free for use as a chip-to-chip
interconnect. This can be either for the firmware on multiple
CS35L56 to share reference audio; or as a bridge to another
device. If it is a firmware interconnect it is owned by the
firmware and the Linux driver should avoid writing the registers.
However, if it is a bridge then Linux may take over and handle
it as a normal codec-to-codec link. Even if the ASP is used
as a firmware-firmware interconnect it is useful to have
ALSA controls for the ASP mixer. They are at least useful for
debugging.
CS35L56 is designed for SDCA and a generic SDCA driver would
know nothing about these chip-specific registers. So if the
ASP is being used on a SoundWire system the firmware sets up the
ASP mixer registers. This means that we can't assume the default
state of these registers. But we don't know the initial state
that the firmware set them to until after the firmware has been
downloaded and booted, which can take several seconds when
downloading multiple amps.
DAPM normally reads the initial state of mux registers during
probe() but this would mean blocking probe() for several seconds
until the firmware has initialized them. To avoid this, the
mixer muxes are set SND_SOC_NOPM to prevent DAPM trying to read
the register state. Custom get/set callbacks are implemented for
ALSA control access, and these can safely block waiting for the
firmware download.
After the firmware download has completed, the state of the
mux registers is known so a work job is queued to call
snd_soc_dapm_mux_update_power() on each of the mux widgets.
Backport note:
This won't apply cleanly to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-11-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 07687cd053 ]
Move the call to cs35l56_set_patch() earlier in cs35l56_init() so
that it only adds the register patch on first-time initialization.
The call was after the post_soft_reset label, so every time this
function was run to re-initialize the hardware after a reset it would
call regmap_register_patch() and add the same reg_sequence again.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 898673b905 ("ASoC: cs35l56: Move shared data into a common data structure")
Link: https://msgid.link/r/20240129162737.497-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ae861c466e ]
The cs35l56->component pointer is used by the suspend-resume handling to
know whether the driver is fully instantiated. This is to prevent it
queuing dsp_work which would result in calling wm_adsp when the driver
is not an instantiated ASoC component. So this pointer must be cleared
by cs35l56_component_remove().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e496112529 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1382d8b551 ]
In the case where __lpass_get_dmactl_handle is called and the driver
id dai_id is invalid the pointer dmactl is not being assigned a value,
and dmactl contains a garbage value since it has not been initialized
and so the null check may not work. Fix this to initialize dmactl to
NULL. One could argue that modern compilers will set this to zero, but
it is useful to keep this initialized as per the same way in functions
__lpass_platform_codec_intf_init and lpass_cdc_dma_daiops_hw_params.
Cleans up clang scan build warning:
sound/soc/qcom/lpass-cdc-dma.c:275:7: warning: Branch condition
evaluates to a garbage value [core.uninitialized.Branch]
Fixes: b81af585ea ("ASoC: qcom: Add lpass CPU driver for codec dma control")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://msgid.link/r/20240221134804.3475989-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1d5a2b5dd0 ]
ASoC is using 2 type of prefix (asoc_xxx() vs snd_soc_xxx()), but there
is no particular reason about that [1].
To reduce confusing, standarding these to snd_soc_xxx() is sensible.
This patch adds asoc_xxx() macro to keep compatible for a while.
It will be removed if all drivers were switched to new style.
Link: https://lore.kernel.org/r/87h6td3hus.wl-kuninori.morimoto.gx@renesas.com [1]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fs3ks26i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 1382d8b551 ("ASoC: qcom: Fix uninitialized pointer dmactl")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e33625c84b ]
The driver must write 0 to HALO_STATE before sending the SYSTEM_RESET
command to the firmware.
HALO_STATE is in DSP memory, which is preserved across a soft reset.
The SYSTEM_RESET command does not change the value of HALO_STATE.
There is period of time while the CS35L56 is resetting, before the
firmware has started to boot, where a read of HALO_STATE will return
the value it had before the SYSTEM_RESET. If the driver does not
clear HALO_STATE, this would return BOOT_DONE status even though the
firmware has not booted.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 8a731fd37f ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240216140535.1434933-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a6122b0b42 ]
This driver includes the legacy GPIO APIs <linux/gpio.h> and
<linux/of_gpio.h> but does not use any symbols from any of
them.
Drop the includes.
Further the driver is requesting "reset-gpios" rather than
just "reset" from the GPIO framework. This is wrong because
the gpiolib core will add "-gpios" before processing the
request from e.g. device tree. Drop the suffix.
The last problem means that the optional RESET GPIO has
never been properly retrieved and used even if it existed,
but nobody noticed.
Fixes: c1124c09e1 ("ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver.")
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-3-ee9f9d4655eb@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit daf3f0f99c ]
There's no need to overwrite fwf_name with a kstrdup() of the cs_dsp part
name. It is trivial to select either fwf_name or cs_dsp.part as the string
to use when building the filename in wm_adsp_request_firmware_file().
This leaves fwf_name entirely owned by the codec driver.
It also avoids problems with freeing the pointer. With the original code
fwf_name was either a pointer owned by the codec driver, or a kstrdup()
created by wm_adsp. This meant wm_adsp must free it if it set it, but not
if the codec driver set it. The code was handling this by using
devm_kstrdup().
But there is no absolute requirement that wm_adsp_common_init() must be
called from probe(), so this was a pseudo-memory leak - each new call to
wm_adsp_common_init() would allocate another block of memory but these
would only be freed if the owning codec driver was removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240129162737.497-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0adf963b84 ]
The SPDIF hardware block found in the H616 SoC has the same layout as
the one found in the H6 SoC, except that it is missing the receiver
side.
Since the driver currently only supports the transmit function, support
for the H616 is identical to what is currently done for the H6.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://msgid.link/r/20240127163247.384439-4-wens@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6cc2aa9a75 ]
Add condition check for cpu dai link initialization for amplifier
codec path, as same pcm id uses for both headset and speaker path
for RENOIR platforms.
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://msgid.link/r/20240118143023.1903984-3-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 34a1066981 upstream.
The tascodec_init() of the snd-soc-tas2781-comlib module is called from
snd-soc-tas2781-i2c and snd-hda-scodec-tas2781-i2c modules. It calls
request_firmware_nowait() with parameter THIS_MODULE and a cont/callback
from the latter modules.
The latter modules can be removed while their callbacks are running,
resulting in a general protection failure.
Add module parameter to tascodec_init() so request_firmware_nowait() can
be called with the module of the callback.
Fixes: ef3bcde75d ("ASoC: tas2781: Add tas2781 driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/118dad922cef50525e5aab09badef2fa0eb796e5.1707076603.git.soyer@irl.hu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fcbe487308 upstream.
commit 74ad8ed651 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops")
introduced a new allocation before the upper bounds check in
do_rx_work. As a result A DSP can cause bad allocations if spewing
garbage.
Fixes: 74ad8ed651 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops")
Reported-by: Tim Van Patten <timvp@google.com>
Cc: stable@vger.kernel.org
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213123834.4827-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 086df711d9 upstream.
WCD938x sound codec driver ignores return status of getting regulators
and returns EINVAL instead of EPROBE_DEFER. If regulator provider
probes after the codec, system is left without probed audio:
wcd938x_codec audio-codec: wcd938x_probe: Fail to obtain platform data
wcd938x_codec: probe of audio-codec failed with error -22
Fixes: 16572522ae ("ASoC: codecs: wcd938x-sdw: add SoundWire driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 6ef5d5b92f ]
There is a path in rt5645_jack_detect_work(), where rt5645->jd_mutex
is left locked forever. That may lead to deadlock
when rt5645_jack_detect_work() is called for the second time.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: cdba4301ad ("ASoC: rt5650: add mutex to avoid the jack detection failure")
Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Link: https://lore.kernel.org/r/1707645514-21196-1-git-send-email-khoroshilov@ispras.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d7332c4a4f ]
With the change in the widget free logic to power down the cores only
when the scheduler widgets are freed, we need to ensure that the
scheduler widget is freed only after all the widgets associated with the
scheduler are freed. This is to ensure that the secondary core that the
scheduler is scheduled to run on is kept powered on until all widgets
that need them are in use. While this works well for dynamic pipelines,
in the case of static pipelines the current logic does not take this into
account and frees all widgets in the order they occur in the
widget_list. So, modify this to ensure that the scheduler widgets are freed
only after all other types of widgets in the widget_list are freed.
Link: https://github.com/thesofproject/linux/issues/4807
Fixes: 31ed8da1c8 ("ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240208133432.1688-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit b53cc6144a upstream.
The PA gain can be set in steps of 1.5 dB from -3 dB to 18 dB, that is,
in 15 levels.
Fix the dB values for the PA volume control as experiments using wsa8835
show that the first 16 levels all map to the same lowest gain while the
last three map to the highest gain.
These values specifically need to be correct for the sound server to
provide proper volume control.
Note that level 0 (-3 dB) does not mute the PA so the mute flag should
also not be set.
Fixes: cdb09e6231 ("ASoC: codecs: wsa883x: add control, dapm widgets and map")
Cc: stable@vger.kernel.org # 6.0
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-2-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 46188db080 upstream.
The LPASS WSA macro codec driver is updating the digital gain settings
behind the back of user space on DAPM events if companding has been
enabled.
As compander control is exported to user space, this can result in the
digital gain setting being incremented (or decremented) every time the
sound server is started and the codec suspended depending on what the
UCM configuration looks like.
Soon enough playback will become distorted (or too quiet).
This is specifically a problem on the Lenovo ThinkPad X13s as this
bypasses the limit for the digital gain setting that has been set by the
machine driver.
Fix this by simply dropping the compander gain offset hack. If someone
cares about modelling the impact of the compander setting this can
possibly be done by exporting it as a volume control later.
Note that the volume registers still need to be written after enabling
clocks in order for any prior updates to take effect.
Fixes: 2c4066e5d4 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Cc: stable@vger.kernel.org # 5.11
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-4-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4d0e8bdfa4 upstream.
The lowest headphones volume setting does not mute so the leave the TLV
mute flag unset.
This is specifically needed to let the sound server use the lowest gain
setting.
Fixes: c03226ba15 ("ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR")
Cc: <stable@vger.kernel.org> # 6.5
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240122091130.27463-1-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c481016bb4 upstream.
The UCM configuration for the Lenovo ThinkPad X13s has up until now
been setting the speaker PA volume to the minimum -3 dB when enabling
the speakers, but this does not prevent the user from increasing the
volume further.
Limit the digital gain and PA volumes to a combined -3 dB in the machine
driver to reduce the risk of speaker damage until we have active speaker
protection in place (or higher safe levels have been established).
Note that the PA volume limit cannot be set lower than 0 dB or
PulseAudio gets confused when the first 16 levels all map to -3 dB.
Also note that this will probably need to be generalised using
machine-specific limits, but a common limit should do for now.
Cc: <stable@vger.kernel.org> # 6.5
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240122181819.4038-3-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 8a8a9ac8a4 ]
If same devices with same device IDs are present on different soundwire
buses, the probe fails due to conflicting device names and sysfs
entries:
sysfs: cannot create duplicate filename '/bus/soundwire/devices/sdw:0:0217:0204:00:0'
The link ID is 0 for both devices, so they should be differentiated by
the controller ID. Add the controller ID so, the device names and sysfs entries look
like:
sdw:1:0:0217:0204:00:0 -> ../../../devices/platform/soc@0/6ab0000.soundwire-controller/sdw-master-1-0/sdw:1:0:0217:0204:00:0
sdw:3:0:0217:0204:00:0 -> ../../../devices/platform/soc@0/6b10000.soundwire-controller/sdw-master-3-0/sdw:3:0:0217:0204:00:0
[PLB changes: use bus->controller_id instead of bus->id]
Fixes: 7c3cd189b8 ("soundwire: Add Master registration")
Cc: stable@vger.kernel.org
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Co-developed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20231017160933.12624-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ab09fb9c62 ]
Warnings related to missing data in firmware manifest have
proven to be too verbose. This relates to description of
DSP module cost expressed in cycles per chunk (CPC). If
a matching value is not found in the manifest, kernel will
pass a zero value and DSP firmware will use a conservative
value in its place.
Downgrade the warnings to dev_dbg().
Fixes: d8a2c98793 ("ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240115092209.7184-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e3b3ec967a ]
It's not granted that all entries of struct sof_conn_stream declare
a `normal_link` (a non-SOF, direct link) string, and this is the case
for SoCs that support only SOF paths (hence do not support both direct
and SOF usecases).
For example, in the case of MT8188 there is no normal_link string in
any of the sof_conn_stream entries and there will be more drivers
doing that in the future.
To avoid possible NULL pointer KPs, add a NULL check for `normal_link`.
Fixes: 0caf1120c5 ("ASoC: mediatek: mt8195: extract SOF common code")
Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://msgid.link/r/20240111105226.117603-1-angelogioacchino.delregno@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 51add1687f ]
dmi_platform_data[] first contains a DMI entry matching:
DMI_MATCH(DMI_PRODUCT_NAME, "EF20"),
and then contains an identical entry except for the match being:
DMI_MATCH(DMI_PRODUCT_NAME, "EF20EA"),
Since these are partial (non exact) DMI matches the first match
will also match any board with "EF20EA" in their DMI product-name,
drop the second, redundant, entry.
Fixes: a4dae468cf ("ASoC: rt5645: Add ACPI-defined GPIO for ECS EF20 series")
Cc: Chris Chiu <chiu@endlessos.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://msgid.link/r/20231126214024.300505-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ddd1ee12a8 ]
The Vangogh machine driver variant based on the MAX98388 amplifier, as
found on Valve's Steam Deck OLED, relies on probing via an ACPI match
table. This worked fine until commit 197b1f7f0d ("ASoC: amd: Add new
dmi entries to config entry") enabled SOF support for the target machine
(i.e. Galileo product), causing the sound card to enter the deferred
probe state indefinitely:
$ cat /sys/kernel/debug/devices_deferred
AMDI8821:00 acp5x_mach: Register card (acp5x-max98388) failed
The issue is related to commit e89f45edb7 ("ASoC: amd: vangogh: Add
check for acp config flags in vangogh platform"), which tries to
mitigate potential conflicts between SOF and generic ACP Vangogh
drivers, due to sharing the PCI device IDs.
However, the solution is effective only if the machine driver is
directly probed by pci-acp5x through platform_device_register_full().
Hence, remove the conflicting ACPI based probing and rely exclusively on
DMI quirks for sound card setup.
Fixes: dba22efd0d ("ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Emil Velikov <emil.velikov@collabora.com>
Link: https://msgid.link/r/20231209203229.878730-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2f03970198 ]
We use partial match for connecting DAI link and DAI widget. We need to
use partial match for disconnecting, too.
Fixes: fe88788779 ("ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget")
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/20231204214713.208951-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e38e252dbc ]
sof_sdw_rt_sdca_jack_exit() are used by different codecs, and some of
them use the same dai name.
For example, rt712 and rt713 both use "rt712-sdca-aif1" and
sof_sdw_rt_sdca_jack_exit().
As a result, sof_sdw_rt_sdca_jack_exit() will be called twice by
mc_dailink_exit_loop(). Set ctx->headset_codec_dev = NULL; after
put_device(ctx->headset_codec_dev); to avoid ctx->headset_codec_dev
being put twice.
Fixes: 5360c67046 ("ASoC: Intel: sof_sdw: add rt712 support")
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@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/20231204214200.203100-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 486ede0df8 ]
The drv_name in enumeration table for ALC5682I-VS codec does not match
the board id string in machine driver. Modify the entry of "10EC5682"
to enumerate "RTL5682" as well and remove invalid entry.
Fixes: 88b4d77d60 ("ASoC: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec")
Reported-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20231204214200.203100-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 50678d339d ]
This driver includes the legacy GPIO APIs <linux/gpio.h> and
<linux/of_gpio.h> but does not use any symbols from any of
them.
Drop the includes.
Further the driver is requesting "reset-gpios" rather than
just "reset" from the GPIO framework. This is wrong because
the gpiolib core will add "-gpios" before processing the
request from e.g. device tree. Drop the suffix.
The last problem means that the optional RESET GPIO has
never been properly retrieved and used even if it existed,
but nobody noticed.
Fixes: 3333cb7187 ("ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.")
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-2-ee9f9d4655eb@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit c344ef36db upstream.
The current code flow is:
1. snd_hdac_device_register()
2. set parameters needed by the hdac driver
3. request_codec_module()
the hdac driver is probed at this point
During boot the codec drivers are not loaded when the hdac device is
registered, it is going to be probed later when loading the codec module,
which point the parameters are set.
On module remove/insert
rmmod snd_sof_pci_intel_tgl
modprobe snd_sof_pci_intel_tgl
The codec module remains loaded and the driver will be probed when the
hdac device is created right away, before the parameters for the driver
has been configured:
1. snd_hdac_device_register()
the hdac driver is probed at this point
2. set parameters needed by the hdac driver
3. request_codec_module()
will be a NOP as the module is already loaded
Move the snd_hdac_device_register() later, to be done right before
requesting the codec module to make sure that the parameters are all set
before the device is created:
1. set parameters needed by the hdac driver
2. snd_hdac_device_register()
3. request_codec_module()
This way at the hdac driver probe all parameters will be set in all cases.
Link: https://github.com/thesofproject/linux/issues/4731
Fixes: a0575b4add ("ASoC: hdac_hda: Conditionally register dais for HDMI and Analog")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20231207095425.19597-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/ZYvUIxtrqBQZbNlC@shine.dominikbrodowski.net
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218304
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit b1b6131bca ]
Some BYTCR x86 tablets with a rt5640 codec have the left and right channels
of their speakers swapped.
Add a new BYT_RT5640_SWAPPED_SPEAKERS quirk for this which sets
cfg-spk:swapped in the components string to let userspace know
about the swapping so that the UCM profile can configure the mixer
to correct this.
Enable this new quirk on the Medion Lifetab S10346 which has its
speakers swapped.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20231217213221.49424-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 99c7bb44f5 ]
Add a quirk for the Medion Lifetab S10346, this BYTCR tablet has no CHAN
package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which
is the default for BYTCR devices.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20231217213221.49424-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a0ffa8115e ]
Masks the "DSP Virtual Mailbox 2 write" interrupt when before
issuing the hibernate command to the DSP. The interrupt is
unmasked when exiting runtime suspend as it is required for
DSP operation.
Without this change the DSP fires an interrupt when hibernating
causing the system spin between runtime suspend and runtime
resume.
Signed-off-by: Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231206160318.1255034-4-rriveram@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c3c8b08894 ]
Use the SYSTEM_SLEEP_PM_OPS handlers to prevent handling an IRQ
when the system is in the middle of suspending or resuming.
Signed-off-by: Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231206160318.1255034-3-rriveram@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 12e102b1bd ]
Make use of the recently introduced EXPORT_GPL_DEV_PM_OPS() macro, to
conditionally export the runtime/system PM functions.
Replace the old SET_{RUNTIME,SYSTEM_SLEEP,NOIRQ_SYSTEM_SLEEP}_PM_OPS()
helpers with their modern alternatives and get rid of the now
unnecessary '__maybe_unused' annotations on all PM functions.
Additionally, use the pm_ptr() macro to fix the following errors when
building with CONFIG_PM disabled:
Signed-off-by: Ricardo Rivera-Matos <rriveram@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231206160318.1255034-2-rriveram@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f44de6973 ]
Change the default MIC detection impedance threshold to 200ohm
to support low mic DC impedance headset.
Signed-off-by: David Rau <David.Rau.opensource@dm.renesas.com>
Link: https://lore.kernel.org/r/20231201042933.26392-1-David.Rau.opensource@dm.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e238b68e6d ]
Move the base_cfg to struct sof_ipc4_gain_data. This struct
describes the message payload passed to the firmware via the mailbox.
It is not wise to be 'clever' and try to use the first part of a struct
as IPC message without marking the message section as packed and aligned.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20231129131411.27516-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c447636970 ]
Separate the IPC message part as struct sof_ipc4_src_data. This struct
describes the message payload passed to the firmware via the mailbox.
It is not wise to be 'clever' and try to use the first part of a struct
as IPC message without marking the message section as packed and aligned.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20231129131411.27516-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a0575b4add ]
The current driver is registering the same dais for each hdev found in the
system which results duplicated widgets to be registered and the kernel
log contains similar prints:
snd_hda_codec_realtek ehdaudio0D0: ASoC: sink widget AIF1TX overwritten
snd_hda_codec_realtek ehdaudio0D0: ASoC: source widget AIF1RX overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Codec Output Pin1 overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Codec Input Pin1 overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Analog Codec Playback overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Digital Codec Playback overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Alt Analog Codec Playback overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Analog Codec Capture overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Digital Codec Capture overwritten
skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Alt Analog Codec Capture overwritten
To avoid such issue, split the dai array into HDMI and non HDMI array and
register them conditionally:
for HDMI hdev only register the dais needed for HDMI
for non HDMI hdev do not register the HDMI dais.
Depends-on: 3d1dc8b103 ("ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available")
Link: https://github.com/thesofproject/linux/issues/4509
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20231128123914.3986-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 31ed8da1c8 ]
In the current code, we enable a widget core when it is set up and
disable it when it is freed. This is problematic with IPC4 because
widget free is essentially a NOP and all widgets are freed in the
firmware when the pipeline is deleted. This results in a crash during
pipeline deletion when one of it's widgets is scheduled to run on a
secondary core and is powered off when widget is freed. So, change the
logic to enable all cores needed by all the modules in a pipeline when
the pipeline widget is set up and disable them after the pipeline
widget is freed.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20231124135743.24674-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0376b995bb ]
With IPC4, a pipeline may contain multiple modules in the data
processing domain and they can be scheduled to run on different cores.
Add a new field in struct snd_sof_pipeline to keep track of all the
cores that are associated with the modules in the pipeline. Set the
pipeline core mask for IPC3 when initializing the pipeline widget IPC
structure. For IPC4, set the core mark when initializing the pipeline
widget and initializing processing modules in the data processing domain.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20231124135743.24674-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3d1dc8b103 ]
When the HDMI is not present due to disabled display support
we will use dummy codec and the HDMI routes will refer to non existent
DAPM widgets.
Trim the route list from the HDMI routes to be able to probe the card even
if the HDMI dais are not registered.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20231124124015.15878-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 347ecf29a6 ]
As the input phy clock frequency will divided by 2 by default
on i.MX8MP with the implementation of clk-imx8mp-audiomix driver,
So the requested frequency need to be updated.
The relation of phy clock is:
sai_pll_ref_sel
sai_pll
sai_pll_bypass
sai_pll_out
sai_pll_out_div2
earc_phy_cg
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Link: https://lore.kernel.org/r/1700702093-8008-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cdba4301ad ]
This patch adds the jd_mutex to protect the jack detection control flow.
And only the headset type could check the button status.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20231122100123.2831753-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c33fd11042 ]
The bit 10 in TX_DPTH_CTRL register controls the TX clock rate.
If this bit is set, TX datapath clock should be = 2* TX bit rate.
If this bit is not set, TX datapath clock should be 10* TX bit rate.
As the spdif only case, we always use 2 * TX bit clock, so
this bit need to be set.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1700617373-6472-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f8ba14b780 ]
skl_platform_register() uses krealloc. When krealloc is fail,
then previous memory is not freed. The leak is also when soc
component registration failed.
Signed-off-by: Kamil Duljas <kamil.duljas@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20231116224112.2209-2-kamil.duljas@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 31e721fbd1 ]
The function has multiple return points at which it is not released
previously allocated memory.
Signed-off-by: Kamil Duljas <kamil.duljas@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20231116213926.2034-2-kamil.duljas@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1501f2597 ]
This issue is reproduced when W=1 build in compiler gcc-12.
The following are sparse warnings:
sound/soc/codecs/nau8822.c:199:25: sparse: sparse: incorrect type in assignment
sound/soc/codecs/nau8822.c:199:25: sparse: expected unsigned short
sound/soc/codecs/nau8822.c:199:25: sparse: got restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202311122320.T1opZVkP-lkp@intel.com/
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Link: https://lore.kernel.org/r/20231117043011.1747594-1-CTLIN0@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d5c65be34d ]
The resources should be freed when function return error.
Signed-off-by: Kamil Duljas <kamil.duljas@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20231116125150.1436-1-kamil.duljas@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 37e6fd0ceb ]
Bit 6 of INPPGA (INPPGAMUTE) does not control the Aux path, it controls
the input PGA path, as can been seen from Figure 8 Input Boost Stage in
the datasheet. Update the naming of things in the driver to match this
and update the routing to also reflect this.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231113155916.1741027-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0c6498a59f ]
HP 255 G10's internal microphone array can be made
to work by adding it to the quirk table.
Signed-off-by: Matus Malych <matus@malych.org>
Link: https://lore.kernel.org/r/20231112165403.3221-1-matus@malych.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b036d8ef31 ]
When a control changes value the return value from _put() should be 1 so
we get events generated to userspace notifying applications of the change.
While the I2S mux gets this right the S/PDIF mux does not, fix the return
value.
Fixes: c8609f3870 ("ASoC: meson: add g12a tohdmitx control")
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20240103-meson-enum-val-v1-4-424af7a8fb91@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 172c88244b ]
When a control changes value the return value from _put() should be 1 so
we get events generated to userspace notifying applications of the change.
We are checking if there has been a change and exiting early if not but we
are not providing the correct return value in the latter case, fix this.
Fixes: af2618a2ee ("ASoC: meson: g12a: add internal DAC glue driver")
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20240103-meson-enum-val-v1-3-424af7a8fb91@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1e00120680 ]
When writing to an enum we need to verify that the value written is valid
for the enumeration, the helper function snd_soc_item_enum_to_val() doesn't
do it since it needs to return an unsigned (and in any case we'd need to
check the return value).
Fixes: c8609f3870 ("ASoC: meson: add g12a tohdmitx control")
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20240103-meson-enum-val-v1-2-424af7a8fb91@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3150b70e94 ]
When writing to an enum we need to verify that the value written is valid
for the enumeration, the helper function snd_soc_item_enum_to_val() doesn't
do it since it needs to return an unsigned (and in any case we'd need to
check the return value).
Fixes: af2618a2ee ("ASoC: meson: g12a: add internal DAC glue driver")
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20240103-meson-enum-val-v1-1-424af7a8fb91@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 38744c3fa0 ]
AUD_PAD_TOP widget's correct register is AFE_AUD_PAD_TOP , and not zero.
Having a zero as register, it would mean that the `snd_soc_dapm_new_widgets`
would try to read the register at offset zero when trying to get the power
status of this widget, which is incorrect.
Fixes: b65c466220 ("ASoC: mediatek: mt8186: support adda in platform driver")
Signed-off-by: Eugen Hristev <eugen.hristev@collabora.com>
Link: https://lore.kernel.org/r/20231229114342.195867-1-eugen.hristev@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f9d378fc68 ]
There is error message when defer probe happens:
fsl_rpmsg rpmsg_audio: Unbalanced pm_runtime_enable!
Fix the error handler with pm_runtime_enable.
Fixes: b73d9e6225 ("ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg")
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://lore.kernel.org/r/20231225080608.967953-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 6dad45f4d2 upstream.
There are two problems with using regcache in this module.
The amplifier has 3 addressing levels (BOOK, PAGE, REG). The firmware
contains blocks that must be written to BOOK 0x8C. The regcache doesn't
know anything about BOOK, so regcache_sync writes invalid values to the
actual BOOK.
The module handles 2 or more separate amplifiers. The amplifiers have
different register values, and the module uses only one regmap/regcache
for all the amplifiers. The regcache_sync only writes the last amplifier
used.
The module successfully restores all the written register values (RC
profile, program, configuration, calibration) without regcache.
Remove regcache functions and set regmap cache_type to REGCACHE_NONE.
Link: https://lore.kernel.org/r/21a183b5a08cb23b193af78d4b1114cc59419272.1701906455.git.soyer@irl.hu/
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Acked-by: Mark Brown <broonie@kernel.org>
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/491aeed0e2eecc3b704ec856f815db21bad3ba0e.1703202126.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 8f0f016475 ]
When flag mclk_with_tere and mclk_direction_output enabled,
The SAI transmitter or receiver will be enabled in very early
stage, that if FSL_SAI_xMR is set by previous case,
for example previous case is one channel, current case is
two channels, then current case started with wrong xMR in
the beginning, then channel swap happen.
The patch is to clear xMR in hw_free() to avoid such
channel swap issue.
Fixes: 3e4a826129 ("ASoC: fsl_sai: MCLK bind with TX/RX enable bit")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://msgid.link/r/1702953057-4499-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 025222a9d6 ]
This fixes a problem introduced while fixing ELD reporting with no jack
set.
Most driver using the hdmi-codec will call the 'plugged_cb' callback
directly when registered to report the initial state of the HDMI connector.
With the commit mentionned, this occurs before jack is ready and the
initial report is lost for platforms actually providing a jack for HDMI.
Fix this by storing the hdmi connector status regardless of jack being set
or not and report the last status when jack gets set.
With this, the initial state is reported correctly even if it is
disconnected. This was not done initially and is also a fix.
Fixes: 15be353d55 ("ASoC: hdmi-codec: register hpd callback on component probe")
Reported-by: Zhengqiao Xia <xiazhengqiao@huaqin.corp-partner.google.com>
Closes: https://lore.kernel.org/alsa-devel/CADYyEwTNyY+fR9SgfDa-g6iiDwkU3MUdPVCYexs2_3wbcM8_vg@mail.gmail.com/
Cc: Hsin-Yi Wang <hsinyi@google.com>
Tested-by: Zhengqiao Xia <xiazhengqiao@huaqin.corp-partner.google.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20231218145655.134929-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Some components like platforms don't have DAIs. If the active count of
these components is ignored pinctrl may be wrongly selected between
default and sleep state. So need to increment or decrement the active
count for components without DAIs to avoid it.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20231204111532.3165-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It may cause confict between constraint variable read and write if
there is more than one stream running. Move rate constraint function to
probe() can avoid it.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
CS42888 codec provides 4 multi-bit ADC and 8 multi-bit DAC. Add support
for this codec in imx-card ASoC machine driver.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 716d4e5373 upstream.
Limit the speaker digital gains to 0dB so that the users will not damage them.
Currently there is a limit in UCM, but this does not stop the user form
changing the digital gains from command line. So limit this in driver
which makes the speakers more safer without active speaker protection in
place.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Johan Hovold <johan+linaro@kernel.org>
Tested-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://lore.kernel.org/r/20231204124736.132185-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
[ johan: backport to 6.6; rename snd_soc_rtd_to_cpu() ]
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fb9ad24485 upstream.
Volume can have ranges that start with negative values, ex: -84dB to
+40dB. Apply correct range check in snd_soc_limit_volume before setting
the platform_max. Without this patch, for example setting a 0dB limit on
a volume range of -84dB to +40dB would fail.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Johan Hovold <johan+linaro@kernel.org>
Reviewed-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://lore.kernel.org/r/20231204124736.132185-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b24e3590c9 upstream.
This patch adds ASUSTeK COMPUTER INC "E1504FA" to the quirks file acp6x-mach.c
to enable microphone array on ASUS Vivobook GO 15.
I have this laptop and can confirm that the patch succeeds in enabling the
microphone array.
Signed-off-by: Malcolm Hart <malcolm@5harts.com>
Cc: stable@vger.kernel.org
Rule: add
Link: https://lore.kernel.org/stable/875y1nt1bx.fsf%405harts.com
Link: https://lore.kernel.org/r/871qcbszh0.fsf@5harts.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 29046a78a3 ]
When wm_adsp_buffer_read() fails, we should free buf->regions.
Otherwise, the callers of wm_adsp_buffer_populate() will
directly free buf on failure, which makes buf->regions a leaked
memory.
Fixes: a792af69b0 ("ASoC: wm_adsp: Refactor compress stream initialisation")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231204074158.12026-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a2f35ed1d2 ]
The -1 value for active_decimator[dai_id] is considered as "not set",
but at probe the table is initialized a 0, this prevents enabling the
DEC0 Mixer since it will be considered as already set.
Initialize the table entries as -1 to fix tx_macro_tx_mixer_put().
Fixes: 1c6a7f5250 ("ASoC: codecs: tx-macro: fix active_decimator array")
Fixes: c1057a08af ("ASoC: codecs: tx-macro: fix kcontrol put")
Signed-off-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://lore.kernel.org/r/20231116-topic-sm8x50-upstream-tx-macro-fix-active-decimator-set-v1-1-6edf402f4b6f@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 14e8442e07 ]
On i.MX8MP, when the TERE and FSD_MSTR enabled before configuring
the word width, there will be no frame sync clock issue, because
old word width impact the generation of frame sync.
TERE enabled earlier only for i.MX8MP case for the hardware limitation,
So need to disable FSD_MSTR before configuring word width, then enable
FSD_MSTR bit for this specific case.
Fixes: 3e4a826129 ("ASoC: fsl_sai: MCLK bind with TX/RX enable bit")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1700474735-3863-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
enlarge the sleep time after reset, to avoid the i2c access
failure after reset sometimes.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
When flag mclk_with_tere and mclk_direction_output enabled,
The SAI transmitter or receiver will be enabled in very early
stage, that if FSL_SAI_xMR is set by previous case,
for example previous case is one channel, current case is
two channels, then current case started with wrong xMR in
the beginning, then channel swap happen.
The patch is to clear xMR in hw_free() to avoid such
channel swap issue.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
Set flipped DAI hardware clock providers/consumers format for CPU DAI
side.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Use macro definition to directly tell CPU DAI drivers if they are clock
provider or consumer.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Period size is constrained to multiple of maxburst on i.MX95.
Applications on i.MX platform like AFE prefer to set the period size to
a value which is usually multiple of 8. To unify with the previous
platforms, maxburst is also specified to 8 on i.MX95.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
This is the 6.6.3 stable release
* tag 'v6.6.3': (526 commits)
Linux 6.6.3
drm/amd/display: Change the DMCUB mailbox memory location from FB to inbox
drm/amd/display: Clear dpcd_sink_ext_caps if not set
...
Signed-off-by: Jason Liu <jason.hui.liu@nxp.com>
Conflicts:
arch/arm64/boot/dts/freescale/fsl-ls208xa.dtsi
drivers/usb/dwc3/core.c
This is the 6.6.2 stable release
* tag 'v6.6.2': (634 commits)
Linux 6.6.2
btrfs: make found_logical_ret parameter mandatory for function queue_scrub_stripe()
btrfs: use u64 for buffer sizes in the tree search ioctls
...
Signed-off-by: Jason Liu <jason.hui.liu@nxp.com>
Conflicts:
drivers/clk/imx/clk-imx8mq.c
drivers/clk/imx/clk-imx8qxp.c
drivers/media/i2c/ov5640.c
drivers/misc/pci_endpoint_test.c
commit 72151ad0cb upstream.
Driver compares widget name in wsa_macro_spk_boost_event() widget event
callback, however it does not handle component's name prefix. This
leads to using uninitialized stack variables as registers and register
values. Handle gracefully such case.
Fixes: 2c4066e5d4 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Cc: stable@vger.kernel.org
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://lore.kernel.org/r/20231003155422.801160-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 805ce81826 upstream.
In the current setup the PA is left unmuted even when the
Soundwire ports are not started streaming. This can lead to click
and pop sounds during start.
There is a same issue in the reverse order where in the PA is
left unmute even after the data stream is stopped, the time
between data stream stopping and port closing is long enough
to accumulate DC on the line resulting in Click/Pop noise
during end of stream.
making use of new mute_unmute_on_trigger flag is helping a
lot with this Click/Pop issues reported on this Codec
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://lore.kernel.org/r/20231027105747.32450-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f0220575e6 upstream.
In some setups like Speaker amps which are very sensitive, ex: keeping them
unmute without actual data stream for very short duration results in a
static charge and results in pop and clicks. To minimize this, provide a way
to mute and unmute such codecs during trigger callbacks.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://lore.kernel.org/r/20231027105747.32450-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
[ johan: backport to v6.6.2 ]
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit fbb74e5637 ]
We need to check for an active device as otherwise we get warnings
for some mcbsp instances for "Runtime PM usage count underflow!".
Reported-by: Andreas Kemnade <andreas@kemnade.info>
Signed-off-by: Tony Lindgren <tony@atomide.com>
Link: https://lore.kernel.org/r/20231030052340.13415-1-tony@atomide.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2cb5478839 ]
The Lenovo Yoga Tab 3 Pro YT3-X90 x86 tablet, which ships with Android with
a custom kernel as factory OS, does not list the used WM5102 codec inside
its DSDT.
Workaround this with a new snd_soc_acpi_intel_baytrail_machines[] entry
which matches on the SST id instead of the codec id like nocodec does,
combined with using a machine_quirk callback which returns NULL on
other machines to skip the new entry on other machines.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20231021211534.114991-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1c48fd6bb ]
Driver will receive exception IPC message and process it by
snd_sof_dsp_panic.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230919092416.4137-10-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1a1c3d794e ]
If the driver properties do not define a cirrus,firmware-uid try to get the
PCI SSID as the UID.
On PCI-based systems the PCI SSID is used to uniquely identify the specific
sound hardware. This is the standard mechanism for x86 systems and is the
way to get a unique system identifier for systems that use the CS35L56 on
SoundWire.
For non-SoundWire systems there is no Windows equivalent of the ASoC driver
in I2C/SPI mode. These would be:
1. HDA systems, which are handled by the HDA subsystem.
2. Linux-specific systems.
3. Composite devices where the cs35l56 is not present in ACPI and is
configured using software nodes.
Case 2 can use the firmware-uid property, though the PCI SSID is supported
as an alternative, as it is the standard PCI mechanism.
Case 3 is a SoundWire system where some other codec is the SoundWire bridge
device and CS35L56 is not listed in ACPI. As these are SoundWire systems
they will normally use the PCI SSID.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d8b387544f ]
If the PCI SSID has been set in the struct snd_soc_acpi_mach_params,
copy this to struct snd_soc_card so that it can be used by other
ASoC components.
This is important for components that must apply system-specific
configuration.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ba2de401d3 ]
Pass the PCI SSID of the audio interface through to the machine driver.
This allows the machine driver to use the SSID to uniquely identify the
specific hardware configuration and apply any platform-specific
configuration.
struct snd_sof_pdata is passed around inside the SOF code, but it then
passes configuration information to the machine driver through
struct snd_soc_acpi_mach and struct snd_soc_acpi_mach_params. So SSID
information has been added to both snd_sof_pdata and
snd_soc_acpi_mach_params.
PCI does not define 0x0000 as an invalid value so we can't use zero to
indicate that the struct member was not written. Instead a flag is
included to indicate that a value has been written to the
subsystem_vendor and subsystem_device members.
sof_pci_probe() creates the struct snd_sof_pdata. It is passed a struct
pci_dev so it can fill in the SSID value.
sof_machine_check() finds the appropriate struct snd_soc_acpi_mach. It
copies the SSID information across to the struct snd_soc_acpi_mach_params.
This done before calling any custom set_mach_params() so that it could be
used by the set_mach_params() callback to apply variant params.
The machine driver receives the struct snd_soc_acpi_mach as its
platform_data.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230912163207.3498161-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d601bb78f0 ]
To avoid power leakage, it is recommended to replace the default pinctrl
state with dynamic pinctrl since certain audio pinmux functions can
remain in a HIGH state even when audio is disabled. Linking pinctrl with
DAPM using SND_SOC_DAPM_PINCTRL will ensure that audio pins remain in
GPIO mode by default and only switch to an audio function when necessary.
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230825024935.10878-2-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The bit 10 in TX_DPTH_CTRL register controls the TX clock rate.
If this bit is set, TX datapath clock should be = 2* TX bit rate.
If this bit is not set, TX datapath clock should be 10* TX bit rate.
As the spdif only case, we always use 2 * TX bit clock, so
this bit need to be set.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1700617373-6472-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
* origin/audio/rpmsg: (9 commits)
ASoC: imx-rpmsg: Add sysclk_id for wm8960 codec
LF-8762-1: ASoC: imx-rpmsg: Force codec power on in low power audio mode
LF-4972: ASoC: fsl_rpmsg: allocate a smaller buffer size for capture
MLK-25741-5: ASoC: fsl_rpmsg: Set CPU DAI for rpmsg-micfil
MLK-25741-3: ASoC: imx-audio-rpmsg: Distinguish ak4497 with others on i.MX8MM
...
* origin/audio/codec: (11 commits)
LF-10364-4: ASoC: fsl_mqs: Add support for i.MX95 platform
ASoC: soc-pcm.c: Make sure DAI parameters cleared if the DAI becomes inactive
LF-7658-1: sound: soc: codecs: pcm186x: add bclk ratio dai ops
LF-7975-2: ASoC: ak5558: check codec exist or not
LF-7975-1: ASoC: ak4458: check codec exist or not
...
* origin/audio/card:
LF-7658-2: sound: soc: fsl: imx pcm512x add pcm186x adc support
MLK-25946: ASoC: imx-card: Set mclk for codec
MLK-24930-1: sound: soc: fsl: imx pcm512x: iqaudio dac
LF-2511-3: ASoC: fsl-asoc-card: Merge some features to this common driver
[ Upstream commit 4bdcbc31ad ]
The name currently used to get the clock includes the dapm prefix.
It should use the name as provided to the widget, without the prefix.
Fixes: 3caac75968 ("ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20231106103712.703962-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 15be353d55 ]
The HDMI hotplug callback to the hdmi-codec is currently registered when
jack is set.
The hotplug not only serves to report the ASoC jack state but also to get
the ELD. It should be registered when the component probes instead, so it
does not depend on the card driver registering a jack for the HDMI to
properly report the ELD.
Fixes: 25ce4f2b35 ("ASoC: hdmi-codec: Get ELD in before reporting plugged event")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20231106104013.704356-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 004fc58ede ]
Property 'playback-codecs' is referenced as 'speaker-codec' in the error
message, and this can lead to confusion.
Correct the error message such that the correct property name is
referenced.
Fixes: 0da16e370d ("ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s")
Signed-off-by: Eugen Hristev <eugen.hristev@collabora.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20231031103139.77395-1-eugen.hristev@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1a3b7eab85 ]
Sometimes the codec probe would be called earlier than the hardware initialization.
Therefore, the speaker route should be added before the the first_hw_init check.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Fixes: f3da2ed110 ("ASoC: rt1712-sdca: enable pm_runtime in probe, keep status as 'suspended'")?
Link: https://lore.kernel.org/r/20231030103644.1787948-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9e630efb5a ]
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context. We fix it by calling
pm_runtime_disable when error returns.
Fixes: 955ac62405 ("ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers")
Signed-off-by: Zhang Shurong <zhang_shurong@foxmail.com>
Link: https://lore.kernel.org/r/tencent_C0D62E6D89818179A02A04A0C248F0DDC40A@qq.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 28809aaeab ]
When adding CODECs to a DAI link, the code should stop processing more
CODECs when the expected number of CODECs are discovered. This fixes a
small corner case issue introduced when support for different devices
on the same SoundWire link was added. In the case of aggregated
devices everything is fine, as all devices intended for the DAI link
will be marked with the same group and any not intended for that DAI
are skipped by the group check. However for non-aggregated devices the
group check is bypassed and the current code does not stop after it
has found the first device. Meaning if additional non-aggregated devices
are present on the same SoundWire link they will be erroneously added
into the DAI link.
Fix this issue, and provide a small optimisation by ceasing to process
devices once we have reached the required number of devices for the
current DAI link.
Fixes: 317dcdecaf ("ASoC: intel: sof_sdw: Allow different devices on the same link")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-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/20231019173411.166759-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f549a82aff ]
In an effort to not call sof_ops_free twice, we stopped running it when
probe was aborted.
Check the result of cancel_work_sync to see if this was the case.
Fixes: 31bb7bd9ff ("ASoC: SOF: core: Only call sof_ops_free() on remove if the probe was successful")
Cc: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Maarten Lankhorst <maarten.lankhorst@linux.intel.com>
Link: https://lore.kernel.org/r/20231009115437.99976-2-maarten.lankhorst@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fbfe616ad4 ]
Otherwise a warning will be detected as below:
warning: Function parameter or member 'mclk' not described in
'codec_priv'
Fixes: 1075df4bde ("ASoC: fsl-asoc-card: add nau8822 support")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20231007040117.22446-1-hui.wang@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a221b2e33 ]
This patch fixes the warnings of "Function parameter or member 'xxx'
not described".
>> sound/soc/fsl/mpc5200_dma.c:116: warning: Function parameter or member 'component' not described in 'psc_dma_trigger'
sound/soc/fsl/mpc5200_dma.c:116: warning: Function parameter or member 'substream' not described in 'psc_dma_trigger'
sound/soc/fsl/mpc5200_dma.c:116: warning: Function parameter or member 'cmd' not described in 'psc_dma_trigger'
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202310061914.jJuekdHs-lkp@intel.com/
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Fixes: 6d1048bc11 ("ASoC: fsl: mpc5200_dma: remove snd_pcm_ops")
Link: https://lore.kernel.org/r/87il7fcqm8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3746284c23 ]
If, for any reason, the open-coded arithmetic causes a wraparound,
the protection that `struct_size()` adds against potential integer
overflows is defeated. Fix this by hardening call to `struct_size()`
with `size_add()`.
Fixes: f9efae9549 ("ASoC: SOF: ipc4-topology: Add support for base config extension")
Signed-off-by: "Gustavo A. R. Silva" <gustavoars@kernel.org>
Reviewed-by: Kees Cook <keescook@chromium.org>
Link: https://lore.kernel.org/r/ZQSr15AYJpDpipg6@work
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3efcb471f8 ]
The commit 1da681e528 ("ASoC: soc-pcm.c: Clear DAIs parameters after
stream_active is updated") tries to make sure DAI parameters can be
cleared properly through moving the cleanup to the place where stream
active status is updated. However, it will cause the cleanup only
happening in soc_pcm_close().
Suppose a case: aplay -Dhw:0 44100.wav 48000.wav. The case calls
soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free()->
soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. The
parameters would be remained in the system even if the playback of
44100.wav is finished.
The case requires us clearing parameters in phase of soc_pcm_hw_free().
However, moving the DAI parameters cleanup back to soc_pcm_hw_free()
has the risk that DAIs parameters never be cleared if there're more
than one stream, see commit 1da681e528 ("ASoC: soc-pcm.c: Clear DAIs
parameters after stream_active is updated") for more details.
To meet all these requirements, in addition to do DAI parameters
cleanup in soc_pcm_hw_free(), also check it in soc_pcm_close() to make
sure DAI parameters cleared if the DAI becomes inactive.
Fixes: 1da681e528 ("ASoC: soc-pcm.c: Clear DAIs parameters after stream_active is updated")
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://lore.kernel.org/r/20230920153621.711373-2-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2d5661e600 ]
According to the documentation, drivers are responsible for undoing at
removal time all runtime PM changes done during probing.
Hence, add the missing calls to pm_runtime_dont_use_autosuspend(), which
are necessary for undoing pm_runtime_use_autosuspend().
Note this would have been handled implicitly by
devm_pm_runtime_enable(), but there is a need to continue using
pm_runtime_enable()/pm_runtime_disable() in order to ensure the runtime
PM is disabled as soon as the remove() callback is entered.
Fixes: f517ba4924 ("ASoC: cs35l41: Add support for hibernate memory retention mode")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-7-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9f8948db98 ]
The interrupt handler invokes pm_runtime_get_sync() without checking the
returned error code.
Add a proper verification and switch to pm_runtime_resume_and_get(), to
avoid the need to call pm_runtime_put_noidle() for decrementing the PM
usage counter before returning from the error condition.
Fixes: f517ba4924 ("ASoC: cs35l41: Add support for hibernate memory retention mode")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-6-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 77bf613f0b ]
Enabling the active/passive shared boosts requires setting SYNC_EN, but
*not* before receiving the PLL Lock signal.
Due to improper error handling, it was not obvious that waiting for the
completion operation times out and, consequently, the shared boost is
never activated.
Further investigations revealed the signal is triggered while
snd_pcm_start() is executed, right after receiving the
SNDRV_PCM_TRIGGER_START command, which happens long after the
SND_SOC_DAPM_PRE_PMU event handler is invoked as part of
snd_pcm_prepare(). That is where cs35l41_global_enable() is called
from.
Increasing the wait duration doesn't help, as it only causes an
unnecessary delay in the invocation of snd_pcm_start(). Moving the wait
and the subsequent regmap operations to the SNDRV_PCM_TRIGGER_START
callback is not a solution either, since they would be executed in an
IRQ-off atomic context.
Solve the issue by setting the SYNC_EN bit in PWR_CTRL3 register right
after receiving the PLL Lock interrupt.
Additionally, drop the unnecessary writes to PWR_CTRL1 register, part of
the original mdsync_up_seq, which would have toggled GLOBAL_EN with
unwanted consequences on PLL locking behavior.
Fixes: f503056493 ("ALSA: cs35l41: Add shared boost feature")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: David Rhodes <david.rhodes@cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-5-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5ad668a9ce ]
Technically, an interrupt handler can be called before probe() finishes
its execution, hence ensure the pll_lock completion object is always
initialized before being accessed in cs35l41_irq().
Fixes: f503056493 ("ALSA: cs35l41: Add shared boost feature")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-4-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4bb5870ab6 ]
The return code of regmap_multi_reg_write() call related to "MDSYNC up"
sequence is shadowed by the subsequent regmap_read_poll_timeout()
invocation, which will hit a timeout in case the write operation above
fails.
Make sure cs35l41_global_enable() returns the correct error code instead
of -ETIMEDOUT.
Additionally, to be able to distinguish between the timeouts of
wait_for_completion_timeout() and regmap_read_poll_timeout(), print an
error message for the former and return immediately. This also avoids
having to wait unnecessarily for the second time.
Fixes: f8264c7592 ("ALSA: cs35l41: Poll for Power Up/Down rather than waiting a fixed delay")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-3-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a9a3f54a23 ]
The return code of regmap_multi_reg_write() call related to "MDSYNC
down" sequence is shadowed by the subsequent
wait_for_completion_timeout() invocation, which is expected to time
timeout in case the write operation failed.
Let cs35l41_global_enable() return the correct error code instead of
-ETIMEDOUT.
Fixes: f503056493 ("ALSA: cs35l41: Add shared boost feature")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Add compatible string and specific soc data to support MQS on i.MX95
platform. Only MQS2 is supported currently.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
Add compatible string and specific soc data to support SAI on i.MX95
platform.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
commit 7dd692217b upstream.
Some Chromebooks do not populate the product family DMI value resulting
in firmware load failures.
Add another quirk detection entry that looks for "Google" in the BIOS
version. Theoretically, PRODUCT_FAMILY could be replaced with
BIOS_VERSION, but it is left as a quirk to be conservative.
Cc: stable@vger.kernel.org
Signed-off-by: Mark Hasemeyer <markhas@chromium.org>
Acked-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20231020145953.v1.1.Iaf5702dc3f8af0fd2f81a22ba2da1a5e15b3604c@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The commit 1da681e528 ("ASoC: soc-pcm.c: Clear DAIs parameters after
stream_active is updated") tries to make sure DAI parameters can be
cleared properly through moving the cleanup to the place where stream
active status is updated. However, it will cause the cleanup only
happening in soc_pcm_close().
Suppose a case: aplay -Dhw:0 44100.wav 48000.wav. The case calls
soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free()->
soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. The
parameters would be remained in the system even if the playback of
44100.wav is finished.
The case requires us clearing parameters in phase of soc_pcm_hw_free().
However, moving the DAI parameters cleanup back to soc_pcm_hw_free()
has the risk that DAIs parameters never be cleared if there're more
than one stream, see commit 1da681e528 ("ASoC: soc-pcm.c: Clear DAIs
parameters after stream_active is updated") for more details.
To meet all these requirements, in addition to do DAI parameters
cleanup in soc_pcm_hw_free(), also check it in soc_pcm_close() to make
sure DAI parameters cleared if the DAI becomes inactive.
Fixes: 1da681e528 ("ASoC: soc-pcm.c: Clear DAIs parameters after stream_active is updated")
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://lore.kernel.org/r/20230920153621.711373-2-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add bclk ratio dai ops to override default lrck div
clock settings; Allow sysclk to be overrided.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
The audio board is moveable, sometime it may not be
attached with EVK board, so add codec exist check
in probe(), make sure the codec is exist then sound
card can be registered.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
[Squash LF-8202]
The audio board is moveable, sometime it may not be
attached with EVK board, so add codec exist check
in probe(), make sure the codec is exist then sound
card can be registered.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Squash: LF-7986
On i.MX8MP, when the TERE and FSD_MSTR enabled before configuring
the word width, there will be no frame sync clock issue, because
old word width impact the generation of frame sync.
TERE enabled earlier only for i.MX8MP case for the hardware limitation,
So need to disable FSD_MSTR before configuring word width, then enable
FSD_MSTR bit for this specific case.
Fixes: 2458507da7 ("LF-8527-1: ASoC: fsl_sai: MCLK bind with TX/RX enable bit")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Chancel Liu <chancel.liu@nxp.com>
Add pcm186x adc support found on hifiberry dacplusadcpro
audio hats
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
In some cases, ASoC machine driver may modify the mclk frequency
according to sample rate but the value in codec is still initial
frequency which should be replaced. For example, we should update
mclk before setup for cs42xx8 mclk relating registers.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
ASoC machine sound driver for IQAudio PiDAC plus/pro
Rev3 for iMX SoC, high resolution codec supporting
upto 384khz sample rate on SAI; Include support for
Hifiberry audio hats that uses external oscillators for
dac system clock.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Squash: [MA-20999-4]
Switch to fsl-asoc-card as default machine driver for
wm8962/wm8960/cs42888/mqs/wm8524 and drop original
machine driver for these codecs. but some features which
have not upstreamed and can't be upstreamed need to be
merge to this common driver.
Merge the constraint of rate for each codec
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
The SPDIF clock may not flexible for all sample rates,
so add constraint according to the dts property.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
The sof imx pcm device is a device which should support
double buffering.
Found this issue with pipewire. When there is no
SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will
set headroom to be zero, and because sof pcm device
don't support residue report, when the latency setting
is small, the "delay" always larger than "target" in
alsa-pcm.c, that reading next period data is not
scheduled on time.
With SNDRV_PCM_INFO_BATCH flag in driver, the pipewire
will select a smaller period size for device, then
the task of reading next period data will be scheduled
on time.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
Without this the micfil_params were incorrectly read.
Fixes 1665ca32b6 ("LF-9376-5: ASoC: SOF: Add DAI configuration for PDM interface")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We need this to signal that DAI link supports only 1 direction that
can only be either playback or capture.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
cstream private data stores sof_compr_stream struct. The current
code is a result of a bad rebase.
In order for this to work we also need to add posn_offset and update
it in sof_compr_set_params.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
[Leo: most of the patch has been upstreamed. The patch probably need to
be dropped or description updated ]
Signed-off-by: Li Yang <leoyang.li@nxp.com>
Commit 9ba23717b2 ("ASoC: SOF: imx8m: Implement DSP start") used
fsl,dsp-ctrl for audiomix compatible named instead of "fsl,imx8mp-audio-blk-ctrl".
Actually, the intention of the change was good but was incomplete.
Because of 61f3c905e66e1 ("LF-4209-6: arm64: dts: imx8mp: update DSP")
we already have a reference to audiomix register via fsl,dsp-ctrl so
we no longer need to get the audiomix regmap via compatible string but
get it via phandle.
So, in this patch we will retrieve theaudiomix regmap via phandle. In a
subsequent patch we will get rid of audiomix-dsp-regmap property since
we already have fsl,dsp-ctrl.
Fixes: 9ba23717b2 ("ASoC: SOF: imx8m: Implement DSP start")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
The 'ignore_machine' field is currently used to ignore all FE dailinks
statically added by the machine drivers, as well as override the
fixups for the BE dailinks. The motivation for this field was
primarily to reuse the same machine driver on Intel devices, both with
legacy and SOF-based platform drivers.
SOF is now used on Mediatek platforms, where the same card uses
SOF-based dailinks to deal with DSP-managed streams, as well as
'regular' dailinks. The 'ignore_machine' field set by the core SOF
platform driver is too strong, with dailinks not managed by SOF being
modified.
This patch adds a stricter filtering so that only dailinks managed by
a topology-based SOF driver are modified.
Reported-by: YC Hung <yc.hung@mediatek.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
These functions are used by the userspace to determine what the firmware
supports and tools like cplay should use in terms of sample rate, bit
rate, buffer size and channel count.
The current implementation uses i.MX8 tested scenarios!
Signed-off-by: Paul Olaru <paul.olaru@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Upstream only accepted our version of DSP clocks, remaining
that we need to find a better way to define DAI/DMA clocks.
Lets add back the DAI/DMA clocks until we find a better solution.
Note that these are optional clocks and we make use of clk bulk API.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
On OF platforms information about machine driver is stored
in sof_pdata->machine_drv_name.
Without this SOF probe fails with:
error: no matching ASoC machine driver found - aborting probe
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
ACPI creates tables with information about the machine driver.
With DT there is no need for such tables because we can directly
get all the information needed from DT file.
This patch introduces machine driver property inside dsp node.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Introduce two DT properties in dsp node:
* fw-filename, optional property giving the firmware filename
(if this is missing fw filename is read from board description)
* tplg-filename, mandatory giving the topology filename.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
On i.MX93, there is only one audio PLL, that some sample rate can't
be supported, so add constraint according to the clock source rate.
Actually, the pll clk added to dts node represents sai is working in
master mode. That is to say, audio pll clk shouldn't be added if sai
works on slave mode.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Squash：9e3214eea1c7 ("MLK-25956: ASoC: fsl_sai: Fix fsl_sai build warning: incompatible-pointer-types")
Squash: f61d8d4e31 ("LF-6648: ASoC: fsl_sai: Add backup constraint rate list")
