[ 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 a39d51ff1f ]
If a usb audio device sets more bits than the amount of channels
it could write outside of the map array.
Signed-off-by: Johan Carlsson <johan.carlsson@teenage.engineering>
Fixes: 04324ccc75 ("ALSA: usb-audio: add channel map support")
Message-ID: <20240313081509.9801-1-johan.carlsson@teenage.engineering>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9fc91a6fe3 ]
After system_resume the amplifers will remain off, even if they were on
before system_suspend.
Use playback_started bool to save the playback state, and restore power
state based on it.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <1742b61901781826f6e6212ffe1d21af542d134a.1709918447.git.soyer@irl.hu>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 68f7f3ff6c ]
Make the amp available immediately after a module
load to avoid having to wait for a PCM hook action.
(eg. unloading & loading the module while listening
music)
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/7f2f65d9212aa16edd4db8725489ae59dbe74c66.1703895108.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 9fc91a6fe3 ("ALSA: hda/tas2781: restore power state after system_resume")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f51de7e30 ]
The runtime_resume function calls prmg_load and apply_calibration
functions, but system_resume also calls them, so calling
pm_runtime_force_resume before reset is unnecessary.
For consistency, do not call the pm_runtime_force_suspend in
system_suspend, as runtime_suspend does the same.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d0b4cc1248b9d375d59c009563da42d60d69eac3.1709918447.git.soyer@irl.hu>
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 bec7760a6c ]
The amplifier doesn't loose register state in software shutdown mode, so
there is no need to reset the cur_* values.
Without these resets, the amplifier can be turned on after
runtime_suspend without waiting for the program and
profile to be restored.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <aa27ae084150988bf6a0ead7e3403bc485d790f8.1709918447.git.soyer@irl.hu>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c850c9121c ]
The system_resume function uses dev_info for tracing, but the other pm
functions use dev_dbg.
Use dev_dbg as the other pm functions.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <140f3c689c9eb5874e6eb48a570fcd8207f06a41.1709918447.git.soyer@irl.hu>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c062166995 ]
Realtek codec on HP Envy laptop series are heavily modified by vendor.
Therefore, need intervention to make it work properly. The patch fixes:
- B&O soundbar speakers (between lid and keyboard) activation
- Enable LED on mute button
- Add missing process coefficient which affects the output amplifier
- Volume control synchronization between B&O soundbar and side speakers
- Unmute headset output on several HP Envy models
- Auto-enable headset mic when plugged
This patch was tested on HP Envy x360 13-AR0107AU with Realtek ALC285
The only unsolved problem is output amplifier of all built-in speakers
is too weak, which causes volume of built-in speakers cannot be loud
as vendor's proprietary driver due to missing _DSD parameter in the
firmware. The solution is currently on research. Expected to has another
patch in the future.
Potential fix to related issues, need test before close those issues:
- https://bugzilla.kernel.org/show_bug.cgi?id=189331
- https://bugzilla.kernel.org/show_bug.cgi?id=216632
- https://bugzilla.kernel.org/show_bug.cgi?id=216311
- https://bugzilla.kernel.org/show_bug.cgi?id=213507
Signed-off-by: Athaariq Ardhiansyah <foss@athaariq.my.id>
Message-ID: <20240310140249.3695-1-foss@athaariq.my.id>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 d7bf738098 ]
clang-16 points out a control flow integrity (kcfi) issue when event
callbacks get converted to incompatible types:
sound/core/seq/seq_midi.c:135:30: error: cast from 'int (*)(struct snd_rawmidi_substream *, const char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
135 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/core/seq/seq_virmidi.c:83:31: error: cast from 'int (*)(struct snd_rawmidi_substream *, const unsigned char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
83 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
For addressing those errors, introduce wrapper functions that are used
for callbacks and bridge to the actual function call with pointer
cast.
The code was originally added with the initial ALSA merge in linux-2.5.4.
[ the patch description shamelessly copied from Arnd's original patch
-- tiwai ]
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240213101020.459183-1-arnd@kernel.org
Link: https://lore.kernel.org/r/20240213135343.16411-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 b34bf65838 ]
It had pop noise from Headphone port when system reboot state.
If NID 58h Index 0x0 to fill default value, it will reduce pop noise.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/7493e207919a4fb3a0599324fd010e3e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
commit 67c3d7717e upstream.
The HP mt440 Thin Client uses an ALC236 codec and needs the
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make the mute and
micmute LEDs work.
There are two variants of the USB-C PD chip on this device. Each uses
a different BIOS and board ID, hence the two entries.
Signed-off-by: Eniac Zhang <eniac-xw.zhang@hp.com>
Signed-off-by: Alexandru Gagniuc <alexandru.gagniuc@hp.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240220175812.782687-1-alexandru.gagniuc@hp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1fdf4e8be7 upstream.
On my EliteBook 840 G8 Notebook PC (ProdId 5S7R6EC#ABD; built 2022 for
german market) the Mute LED is always on. The mute button itself works
as expected. alsa-info.sh shows a different subsystem-id 0x8ab9 for
Realtek ALC285 Codec, thus the existing quirks for HP 840 G8 don't work.
Therefore, add a new quirk for this type of EliteBook.
Signed-off-by: Hans Peter <flurry123@gmx.ch>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240219164518.4099-1-flurry123@gmx.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 77ce96543b upstream.
The local helper function to compare the given pair of cycle count
evaluates them. If the left value is less than the right value, the
function returns negative value.
If the safe cycle is less than the current cycle, it is the case of
cycle lost. However, it is not currently handled properly.
This commit fixes the bug.
Cc: <stable@vger.kernel.org>
Fixes: 705794c53b ("ALSA: firewire-lib: check cycle continuity")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218033026.72577-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 4df49712eb ]
We forgot to remove the line for snd-rtctimer from Makefile while
dropping the functionality. Get rid of the stale line.
Fixes: 34ce71a96d ("ALSA: timer: remove legacy rtctimer")
Link: https://lore.kernel.org/r/20240221092156.28695-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
