[ Upstream commit b9a98cdd3ac7b80d8ea0f6acd81c88ad3d8bcb4a ]
The Clarett 8Pre USB works the same as the Clarett+ 8Pre, only the USB
ID is different.
Tested-by: Philippe Perrot <philippe@perrot-net.fr>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/e59f47b29e2037f031b56bde10474c6e96e31ba5.1694705811.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d98cc489029dba4d99714c2e8ec4f5ba249f6851 ]
By moving the USB IDs from the device_info struct into
scarlett2_devices[], that will allow for devices with different
USB IDs to share the same device_info.
Tested-by: Philippe Perrot <philippe@perrot-net.fr>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/8263368e8d49e6fcebc709817bd82ab79b404468.1694705811.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: b9a98cdd3ac7 ("ALSA: scarlett2: Add support for Clarett 8Pre USB")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 23fb6bc2696119391ec3a92ccaffe50e567c515e ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit db185362fca554b201e2c62beb15a02bb39a064b ]
The ASUS M7600RE (Vivobook Pro 16X OLED) needs a quirks-table entry for the
internal microphone to function properly.
Signed-off-by: Mitch Cooley <m.cooley.198@gmail.com>
Link: https://msgid.link/r/CALijGznExWW4fujNWwMzmn_K=wo96sGzV_2VkT7NjvEUdkg7Gw@mail.gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9b714a59b719b1ba9382c092f0f7aa4bbe94eba1 ]
The speakers on the Lenovo Yoga 9 14IMH9 are similar to previous generations
such as the 14IAP7, and the bass speakers can be fixed using similar methods
with one caveat: 14IMH9 uses CS35L41 amplifiers which need to be activated
separately.
Signed-off-by: Jichi Zhang <i@jichi.ca>
Message-ID: <20240315081954.45470-3-i@jichi.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5b417fe0cded0b5917683398e6519aae8045cd40 ]
Update board selection with tables specifying supported I2S
configurations. DMIC/HDAudio board selection require no update as
dmic/hdaudio machine boards are generic and not tied to any specific
codec.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ]
Miglia Harmony Audio (OXFW970) has a quirk to put the number of
accumulated quadlets in CIP payload into the dbc field of CIP header.
This commit handles the quirk in the packet processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a13f0c3c0e8fb3e61fbfd99c6b350cf9be0c4660 ]
Valve's Steam Deck OLED is uniquely identified by vendor and product
name (Galileo) DMI fields.
Simplify the quirk by removing the unnecessary match on product family.
Additionally, fix the related comment as it points to the old product
variant.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Emil Velikov <emil.velikov@collabora.com>
Link: https://msgid.link/r/20231219030728.2431640-7-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 1576f263ee2147dc395531476881058609ad3d38 upstream.
This patch addresses an issue with the Panasonic CF-SZ6's existing quirk,
specifically its headset microphone functionality. Previously, the quirk
used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design
of a single 3.5mm jack for both mic and audio output effectively. The
device uses pin 0x19 for the headset mic without jack detection.
Following verification on the CF-SZ6 and discussions with the original
patch author, i determined that the update to
ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change
is custom-designed for the CF-SZ6's unique hardware setup, which includes
a single 3.5mm jack for both mic and audio output, connecting the headset
microphone to pin 0x19 without the use of jack detection.
Fixes: 0fca97a29b ("ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk")
Signed-off-by: I Gede Agastya Darma Laksana <gedeagas22@gmail.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240401174602.14133-1-gedeagas22@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit daf6c4681a74034d5723e2fb761e0d7f3a1ca18f upstream.
This patch adds the existing fixup to certain TF platforms implementing
the ALC274 codec with a headset jack. It fixes/activates the inactive
microphone of the headset.
Signed-off-by: Christoffer Sandberg <cs@tuxedo.de>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240328102757.50310-1-wse@tuxedocomputers.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit b9846a386734e73a1414950ebfd50f04919f5e24 ]
Before ACP firmware loading, DSP interrupts are not expected.
Sometimes after reboot, it's observed that before ACP firmware is loaded
false DSP interrupt is reported.
Registering the interrupt handler before acp initialization causing false
interrupts sometimes on reboot as ACP reset is not applied.
Correct the sequence by invoking acp initialization sequence prior to
registering interrupt handler.
Fixes: 738a2b5e2c ("ASoC: SOF: amd: Add IPC support for ACP IP block")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240404041717.430545-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 03f56ed4ead162551ac596c9e3076ff01f1c5836 ]
As already anticipated in the original commit, playback was broken for
very short samples. I just didn't expect it to be an actual problem,
because we're talking about less than 1.5 milliseconds here. But clearly
such wavetable samples do actually exist.
The problem was that for such short samples we'd set the current
position beyond the end of the loop, so we'd run off the end of the
sample and play garbage.
This is a bigger (more audible) problem than the original one, which was
that we'd start playback with garbage (whatever was still in the cache),
which would be mostly masked by the note's attack phase.
So revert to the old behavior for now. We'll subsequently fix it
properly with a bigger patch series.
Note that this isn't a full revert - the dead code is not re-introduced,
because that would be silly.
Fixes: df335e9a8b ("ALSA: emu10k1: fix synthesizer sample playback position and caching")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240401145805.528794-1-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fc563aa900659a850e2ada4af26b9d7a3de6c591 ]
In snd_soc_info_volsw(), mask is generated by figuring out the index of
the most significant bit set in max and converting the index to a
bitmask through bit shift 1. Unintended wraparound occurs when max is an
integer value with msb bit set. Since the bit shift value 1 is treated
as an integer type, the left shift operation will wraparound and set
mask to 0 instead of all 1's. In order to fix this, we type cast 1 as
`1ULL` to prevent the wraparound.
Fixes: 7077148fb5 ("ASoC: core: Split ops out of soc-core.c")
Signed-off-by: Stephen Lee <slee08177@gmail.com>
Link: https://msgid.link/r/20240326010131.6211-1-slee08177@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit adb354bbc231b23d3a05163ce35c1d598512ff64 ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: a0b7c59ac1 ("ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c8b2e5c1b959d100990e4f0cbad38e7d047bb97c ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 7a8735c155 ("ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit aae86cfd8790bcc7693a5a0894df58de5cb5128c ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: b69de265bd ("ASoC: rt711: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ee287771644394d071e6a331951ee8079b64f9a7 ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 23adeb7056 ("ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 310a5caa4e861616a27a83c3e8bda17d65026fa8 ]
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 02fb23d727 ("ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 ]
Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules
to be loaded on boot so that Serial Multi Instantiate can add the
devices to the bus and begin the driver init sequence.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Message-ID: <20240328121355.18972-1-simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f193957b0fbbba397c8bddedf158b3bf7e4850fc ]
wm_adsp_write_ctl() must hold the pwr_lock mutex when calling
cs_dsp_get_ctl().
This was previously partially fixed by commit 781118bc2f
("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()")
but this only put locking around the call to cs_dsp_coeff_write_ctrl(),
missing the call to cs_dsp_get_ctl().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 781118bc2f ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()")
Link: https://msgid.link/r/20240307110227.41421-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cafe9c6a72cf1ffe96d2561d988a141cb5c093db ]
Initialization is completed before adding the component as that can
start the process of the device binding and trigger actions that check
init_done.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Message-ID: <20240325145510.328378-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 051e0840ffa8ab25554d6b14b62c9ab9e4901457 upstream.
The dreamcastcard->timer could schedule the spu_dma_work and the
spu_dma_work could also arm the dreamcastcard->timer.
When the snd_pcm_substream is closing, the aica_channel will be
deallocated. But it could still be dereferenced in the worker
thread. The reason is that del_timer() will return directly
regardless of whether the timer handler is running or not and
the worker could be rescheduled in the timer handler. As a result,
the UAF bug will happen. The racy situation is shown below:
(Thread 1) | (Thread 2)
snd_aicapcm_pcm_close() |
... | run_spu_dma() //worker
| mod_timer()
flush_work() |
del_timer() | aica_period_elapsed() //timer
kfree(dreamcastcard->channel) | schedule_work()
| run_spu_dma() //worker
... | dreamcastcard->channel-> //USE
In order to mitigate this bug and other possible corner cases,
call mod_timer() conditionally in run_spu_dma(), then implement
PCM sync_stop op to cancel both the timer and worker. The sync_stop
op will be called from PCM core appropriately when needed.
Fixes: 198de43d75 ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Duoming Zhou <duoming@zju.edu.cn>
Message-ID: <20240326094238.95442-1-duoming@zju.edu.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ae065d0ce9e36ca4efdfb9b96ce3395bd1c19372 upstream.
The "Speaker Digital Gain" kcontrol controls the TAS2781_DVC_LVL (0x1A)
register. Unfortunately the tas2563 does not have DVC_LVL, but has
INT_MASK0 in 0x1A, which has been misused so far.
Since commit c1947ce61ff4 ("ALSA: hda/realtek: tas2781: enable subwoofer
volume control") the volume of the tas2781 amplifiers can be controlled
by the master volume, so this digital gain kcontrol is not needed.
Remove it.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <741fc21db994efd58f83e7aef38931204961e5b2.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 861b3415e4dee06cc00cd1754808a7827b9105bf upstream.
This reverts commit ed00a6945dc32462c2d3744a3518d2316da66fcc,
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>
commit a17bd44c0146b00fcaa692915789c16bd1fb2a81 upstream.
The HP EliteBook using ALC236 codec which using 0x02 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240304134033.773348-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 34ab5bbc6e82214d7f7393eba26d164b303ebb4e upstream.
It will be enable headset Mic for Acer NB platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/fe0eb6661ca240f3b7762b5b3257710d@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d397b6e56151099cf3b1f7bfccb204a6a8591720 upstream.
Headset Mic will no show at resume back.
This patch will fix this issue.
Fixes: d7f32791a9 ("ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/4713d48a372e47f98bba0c6120fd8254@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9e2ab4b18ebd46813fc3459207335af4d368e323 ]
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 f31e0d0c2cad23e0cc48731634f85bb2d8707790 ]
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 a39d51ff1f52cd0b6fe7d379ac93bd8b4237d1b7 ]
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 9fc91a6fe37c78ef301aed4251f7e50b8524e72d ]
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 68f7f3ff6c2a0998be9dc07622bd0d16fd1fda20 ]
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: 9fc91a6fe37c ("ALSA: hda/tas2781: restore power state after system_resume")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f51de7e30c7282162a631af8a425b54a4576346 ]
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 76f5f55c45b906710c9565a7e68c8d782c46b394 ]
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: 5f51de7e30c7 ("ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bec7760a6c5fa59593dac264fa0c628e46815986 ]
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 c850c9121cc8de867ce3ac36b9ae9d05f62bef14 ]
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 c062166995c9e57d5cd508b332898f79da319802 ]
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 59c6a3a43b221cc2a211181b1298e43b2c2df782 ]
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 e3741a8d28a1137f8b19ae6f3d6e3be69a454a0a ]
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 98f681b0f84cfc3a1d83287b77697679e0398306 ]
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 5ad992c71b6a8e8a547954addc7af9fbde6ca10a ]
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 98ac85a00f31d2e9d5452b825a9ed0153d934043 ]
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 d7bf73809849463f76de42aad62c850305dd6c5d ]
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 9a6d7c4fb2801b675a9c31a7ceb78c84b8c439bc ]
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 222be59e5eed1554119294edc743ee548c2371d0 ]
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 d0ada20279db2649a7549a2b8a4a3379c59f238d ]
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 96e202f8c52ac49452f83317cf3b34cd1ad81e18 ]
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 b3a51137607cee7c814cd3a75d96f78b9ee1dc1f ]
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 f8b0127aca8c60826e7354e504a12d4a46b1c3bb ]
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 ed00a6945dc32462c2d3744a3518d2316da66fcc ]
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 b34bf65838f7c6e785f62681605a538b73c2808c ]
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 c40aad7c81e5fba34b70123ed7ce3397fa62a4d2 ]
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 f7fe85b229bc30cb5dc95b4e9015a601c9e3a8cd ]
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 551539a8606e28cb2a130f8ef3e9834235b456c4 ]
The DMI strings used for the LattePanda board DMI quirks are very generic.
Using the dmidecode database from https://linux-hardware.org/ shows
that the chosen DMI strings also match the following 2 laptops
which also have a rt5645 codec:
Insignia NS-P11W7100 https://linux-hardware.org/?computer=E092FFF8BA04
Insignia NS-P10W8100 https://linux-hardware.org/?computer=AFB6C0BF7934
All 4 hw revisions of the LattePanda board have "S70CR" in their BIOS
version DMI strings:
DF-BI-7-S70CR100-*
DF-BI-7-S70CR110-*
DF-BI-7-S70CR200-*
LP-BS-7-S70CR700-*
See e.g. https://linux-hardware.org/?computer=D98250A817C0
Add a partial (non exact) DMI match on this string to make the LattePanda
board DMI match more precise to avoid false-positive matches.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://msgid.link/r/20240211212736.179605-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d1722057477a3786b8c0d60c28fc281f6ecf1cc3 ]
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 4703b014f28bf7a2e56d1da238ee95ef6c5ce76b 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: 07f7d6e7a124 ("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 67c3d7717efbd46092f217b1f811df1b205cce06 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 1fdf4e8be7059e7784fec11d30cd32784f0bdc83 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 77ce96543b03f437c6b45f286d8110db2b6622a3 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 eba2eb2495f47690400331c722868902784e59de ]
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 c14f09f010cc569ae7e2f6ef02374f6bfef9917e ]
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: 07f7d6e7a124 ("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 f6c967941c5d6fa526fdd64733a8d86bf2bfab31 ]
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 07f7d6e7a124d3e4de36771e2a4926d0e31c2258 ]
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 07687cd0539f8185b6ba0c0afba8473517116d6a ]
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 ae861c466ee57e15a29d97629e1c564e3f714a4f ]
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 1382d8b55129875b2e07c4d2a7ebc790183769ee ]
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 1d5a2b5dd0a8d2b2b535b5266699429dbd48e62f ]
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: 1382d8b55129 ("ASoC: qcom: Fix uninitialized pointer dmactl")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4df49712eb54141be00a9312547436d55677f092 ]
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>
[ Upstream commit e33625c84b75e4f078d7f9bf58f01fe71ab99642 ]
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 a6122b0b4211d132934ef99e7b737910e6d54d2f ]
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 eaa1b01fe709d6a236a9cec74813e0400601fd23 ]
For devices with multiple clock sources connected to a selector, we need
to check what a clock selector control request has returned. This is
needed to ensure that a requested clock source is indeed selected and for
autoclock feature to work.
For devices with single clock source connected, if we get an error there
is nothing else we can do about it. We can't skip clock selector setup as
it is required by some devices. So lets just ignore error in this case.
This should fix various buggy Mackie devices:
[ 649.109785] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.111946] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.113822] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
There is also interesting info from the Windows documentation [1] (this
is probably why manufacturers dont't even test this feature):
"The USB Audio 2.0 driver doesn't support clock selection. The driver
uses the Clock Source Entity, which is selected by default and never
issues a Clock Selector Control SET CUR request."
Link: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/usb-2-0-audio-drivers [1]
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217314
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218175
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218342
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240201115308.17838-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit daf3f0f99cde93a066240462b7a87cdfeedc04c0 ]
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 0adf963b8463faa44653e22e56ce55f747e68868 ]
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 346f59d1e8ed0eed41c80e1acb657e484c308e6a ]
Many devices with a single alternate setting do not have a Valid
Alternate Setting Control and validation performed by
validate_sample_rate_table_v2v3() doesn't work on them and is not
really needed. So check the presense of control before sending
altsetting validation requests.
MOTU Microbook IIc is suffering the most without this check. It
takes up to 40 seconds to bootup due to how slow it switches
sampling rates:
[ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60
[ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0
[ 2659.164829] usb 3-2: Product: MicroBook IIc
[ 2659.164830] usb 3-2: Manufacturer: MOTU
[ 2659.166204] usb 3-2: Found last interface = 3
[ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug
[ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3
[ 2679.322321] usb 3-2: Creating new data endpoint #3
[ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1
[ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110
[ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug
[ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84
[ 2694.444721] usb 3-2: Creating new data endpoint #84
[ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240129121254.3454481-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 56beedc88405fd8022edfd1c2e63d1bc6c95efcb ]
Apollo Lake seems to also suffer from IRQ timing issues. After being up for ~4
minutes, a Pentium N4200 system ends up falling back to workqueue-based IRQ
handling:
[ 208.019906] snd_hda_intel 0000:00:0e.0: IRQ timing workaround is activated
for card #0. Suggest a bigger bdl_pos_adj.
Unfortunately, the Baytrail and Braswell workaround value of 32 samples isn't
enough to fix the issue here. Default to 64 samples.
Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240122114512.55808-3-rsalvaterra@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6cc2aa9a75f2397d42b78d4c159bc06722183c78 ]
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 34a1066981a967eab619938e7b35a9be6b4c34e1 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 fcbe4873089c84da641df75cda9cac2e9addbb4b 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 4639c5021029d49fd2f97fa8d74731f167f98919 upstream.
The SWS JS201D need a different pinconfig from windows driver.
Add a quirk to use a specific pinconfig to SWS JS201D.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240205013802.51907-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 32f03f4002c5df837fb920eb23fcd2f4af9b0b23 upstream.
The HP mt645 G7 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/20240215154922.778394-1-alexandru.gagniuc@hp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1513664f340289cf10402753110f3cff12a738aa upstream.
The HP ZBook Power using ALC236 codec which using 0x02 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240122074826.1020964-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a2ed0a44d637ef9deca595054c206da7d6cbdcbc upstream.
Customer has reported an issue with specific desktop platform
where two CS42L42 codecs are connected to CS8409 HDA bridge.
If "Master Volume Control" is created then on Ubuntu OS UCM
left/right balance slider in UI audio settings has no effect.
This patch will fix this issue for a target paltform.
Fixes: 20e5077241 ("ALSA: hda/cs8409: Add support for dolphin")
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240122184710.5802-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 086df711d9b886194481b4fbe525eb43e9ae7403 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>
commit c7de2d9bb68a5fc71c25ff96705a80a76c8436eb upstream.
Vaio VJFE-ADL is equipped with ALC269VC, and it needs
ALC298_FIXUP_SPK_VOLUME quirk to make its headset mic work.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240201122114.30080-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2468e8922d2f6da81a6192b73023eff67e3fefdd upstream.
There currently exists two thinkpad headset jack fixups:
ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK
ALC285_FIXUP_THINKPAD_HEADSET_JACK
The latter is applied to alc285 and alc287 thinkpads which contain
bass speakers.
However, the former was only being applied to alc285 thinkpads,
leaving non-bass alc287 thinkpads with no headset button controls.
This patch fixes that by adding ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK
to the alc287 chains, allowing the detection of headset buttons.
Signed-off-by: José Relvas <josemonsantorelvas@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240131113407.34698-3-josemonsantorelvas@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit efb56d84dd9c3de3c99fc396abb57c6d330038b5 upstream.
If you connect an external headset/microphone to the 3.5mm jack on the
Acer Swift 1 SF114-32 it does not recognize the microphone. This fixes
that and gives the user the ability to choose between internal and
headset mic.
Signed-off-by: David Senoner <seda18@rolmail.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240126155626.2304465-1-seda18@rolmail.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c6dce23ec993f7da7790a9eadb36864ceb60e942 upstream.
The laptop requires a quirk ID to enable its internal microphone. Add
it to the DMI quirk table.
Reported-by: Techno Mooney <techno.mooney@gmail.com>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218402
Cc: stable@vger.kernel.org
Signed-off-by: Techno Mooney <techno.mooney@gmail.com>
Signed-off-by: Bagas Sanjaya <bagasdotme@gmail.com>
Link: https://msgid.link/r/20240129081148.1044891-1-bagasdotme@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 6ef5d5b92f7117b324efaac72b3db27ae8bb3082 ]
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: cdba4301adda ("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 e5aa6d51a2ef8c7ef7e3fe76bebe530fb68e7f08 ]
Commit 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic
CS35L56 amplifier") adds configs SND_HDA_SCODEC_CS35L56_{I2C,SPI},
which selects the non-existing config CS_DSP. Note the renaming in
commit d7cfdf17cb ("firmware: cs_dsp: Rename KConfig symbol CS_DSP ->
FW_CS_DSP"), though.
Select the intended config FW_CS_DSP.
This broken select command probably was not noticed as the configs also
select SND_HDA_CS_DSP_CONTROLS and this then selects FW_CS_DSP. So, the
select FW_CS_DSP could actually be dropped, but we will keep this
redundancy in place as the author originally also intended to have this
redundancy of selects in place.
Fixes: 73cfbfa9ca ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com>
Reviewed-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240209082044.3981-1-lukas.bulwahn@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d7332c4a4f1a7d16f054c6357fb65c597b6a86a7 ]
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: 31ed8da1c8e5 ("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 668abe6dc7b61941fa5c724c06797efb0b87f070 upstream.
The quirk table entries should be put in the USB ID order, but some
entries have been put in random places. Re-sort them.
Fixes: bf990c1023 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue")
Fixes: fd28941cff ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless")
Fixes: dfd5fe19db ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless")
Fixes: 4a63e68a29 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.")
Fixes: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Fixes: 4fb7c24f69 ("ALSA: usb-audio: Add quirk for Fiero SC-01")
Fixes: 2307a0e1ca ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)")
Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7822baa844a87cbb93308c1032c3d47d4079bb8a upstream.
The RODE NT-USB+ is marketed as a professional usb microphone, however the
usb audio interface is a mess:
[ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd
[ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4
[ 1.503912] usb 1-5: config 1 has no interface number 4
[ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09
[ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ 1.519697] usb 1-5: Product: RØDE NT-USB+
[ 1.519699] usb 1-5: Manufacturer: RØDE
[ 1.519700] usb 1-5: SerialNumber: 1D773A1A
[ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82
[ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82
[ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2
Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get.
I have asked Rode support to fix it, but they show no interest.
Signed-off-by: Sean Young <sean@mess.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a969210066054ea109d8b7aff29a9b1c98776841 upstream.
The device fails to initialize otherwise, giving the following error:
[ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1
Signed-off-by: Julian Sikorski <belegdol+github@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240123084935.2745-1-belegdol+github@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d915a6850e27efb383cd4400caadfe47792623df upstream.
Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601
Cc: <stable@vger.kernel.org>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240124130239.358298-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b53cc6144a3f6c8b56afcdec89d81195c9b0dc69 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 46188db080bd1df7d2d28031b89e56f2fdbabd67 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 4d0e8bdfa4a57099dc7230952a460903f2e2f8de 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 c481016bb4f8a9c059c39ac06e7b65e233a61f6a 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 7aeb259086487417f0fecf66e325bee133e8813a ]
When OMTP headset plugin the headset jack of CX8070 and SN6160 sound cards,
the headset type detection circuit will recognize the headset type as CTIA.
At this point, plugout and plugin the headset will get the correct headset
type as OMTP.
The reason for the failure of headset type recognition is that the sound
card creation will enable the VREF voltage of the headset mic, which
interferes with the headset type automatic detection circuit. Plugout and
plugin the headset will restart the headset detection and get the correct
headset type.
The patch is disable the VREF voltage when the headset is not present, and
will enable the VREF voltage when the headset is present.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Link: https://lore.kernel.org/r/20240108110235.3867-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 26257869672fd4a06a60c2da841e15fb2cb47bbe ]
In a couple of loops over the all streams, we check the bitmap against
the loop counter. A more correct reference would be, however, the
index of each stream, instead.
This patch corrects the check of bitmaps to the stream index.
Note that this change doesn't fix anything for now; all existing
drivers set up the stream indices properly, hence the loop count is
always equal with the stream index. That said, this change is only
for consistency.
Link: https://lore.kernel.org/r/20231121154125.4888-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8a8a9ac8a4972ee69d3dd3d1ae43963ae39cee18 ]
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 ab09fb9c629ed3aaea6a82467f08595dbc549726 ]
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 454abb80e26ab85323a30e52aa7b0ee9aae1d38a ]
Since commit 4005d1ba0a ("ASoC: soc-dai: don't call PCM audio ops if
the stream is not supported") HDMI playback is broken with avs driver.
This happens because for HDMI stream (unlike generic HDA one)
channels_min for stream is not set when creating PCMs. Fix this by
setting the value based on first available converter.
Fixes: 4005d1ba0a ("ASoC: soc-dai: don't call PCM audio ops if the stream is not supported")
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240112113349.2905328-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e3b3ec967a7d93b9010a5af9a2394c8b5c8f31ed ]
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>
commit fb3c007fde80d9d3b4207943e74c150c9116cead upstream.
Lenovo M70 Gen5 is equipped with ALC623, and it needs
ALC283_FIXUP_HEADSET_MIC quirk to make its headset mic work.
Signed-off-by: Bin Li <bin.li@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240117154123.21578-1-bin.li@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b018cee7369896c7a15bfdbe88f168f3dbd8ba27 upstream.
On some HP ZBooks, the audio LEDs can be enabled by
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF. So use it accordingly.
Signed-off-by: Yo-Jung Lin <leo.lin@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240116020722.27236-1-leo.lin@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bc7863d18677df66b2c7a0e172c91296ff380f11 upstream.
This HP Laptop uses ALC236 codec with COEF 0x07 idx 1 controlling
the mute LED. This patch enables the already existing quirk for
this device.
Signed-off-by: Çağhan Demir <caghandemir@marun.edu.tr>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240115172303.4718-1-caghandemir@marun.edu.tr
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a03cfad512ac24a35184d7d87ec0d5489e1cb763 upstream.
There was a typo in oxygen mixer code that didn't update the right
channel value properly for the capture volume. Let's fix it.
This trivial fix was originally reported on Bugzilla.
Fixes: a360156049 ("[ALSA] oxygen: add front panel controls")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=156561
Link: https://lore.kernel.org/r/20240112111023.6208-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 04f8f053252b86c7583895c962d66747ecdc61b7 ]
Ensure the value passed to scarlett2_mixer_ctl_put() is between 0 and
SCARLETT2_MIXER_MAX_VALUE so we don't attempt to access outside
scarlett2_mixer_values[].
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Link: https://lore.kernel.org/r/3b19fb3da641b587749b85fe1daa1b4e696c0c1b.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 50603a67daef161c78c814580d57f7f0be57167e ]
The *_ctl_get() functions which call scarlett2_update_*() were not
checking the return value. Fix to check the return value and pass to
the caller.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Link: https://lore.kernel.org/r/32a5fdc83b05fa74e0fcdd672fbf71d75c5f0a6d.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2190b9aea4eb92ccf3176e35c17c959e40f1a81b ]
Line outputs 3 & 4 on the Gen 3 18i8 are internally the analogue 7 and
8 outputs, and this renumbering is hidden from the user by
line_out_remap(). By allowing higher values (representing non-analogue
outputs) to be passed to line_out_remap(), repeated code from
scarlett2_mux_src_enum_ctl_get() and scarlett2_mux_src_enum_ctl_put()
can be removed.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/3b70267931f5994628ab27306c73cddd17b93c8f.1698342632.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 50603a67daef ("ALSA: scarlett2: Add missing error checks to *_ctl_get()")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ca459dfa7d4ed9098fcf13e410963be6ae9b6bf3 ]
scarlett2_usb_set_config() calls scarlett2_usb_get() but was not
checking the result. Return the error if it fails rather than
continuing with an invalid value.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e15fae6c5 ("ALSA: usb-audio: scarlett2: Allow bit-level access to config")
Link: https://lore.kernel.org/r/def110c5c31dbdf0a7414d258838a0a31c0fab67.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f6ff6931a1c0065a55448108940371e1ac8075f ]
scarlett2_config_save() was ignoring the return value from
scarlett2_usb(). As this function is not called from user-space we
can't return the error, so call usb_audio_err() instead.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Link: https://lore.kernel.org/r/bf0a15332d852d7825fa6da87d2a0d9c0b702053.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 51add1687f39292af626ac3c2046f49241713273 ]
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 ddd1ee12a8fb6e4d6f86eddeba64c135eee56623 ]
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 2f03970198d6438d95b96f69041254bd39aafed0 ]
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 e38e252dbceeef7d2f848017132efd68e9ae1416 ]
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 486ede0df82dd74472c6f5651e38ff48f7f766c1 ]
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 50678d339d670a92658e5538ebee30447c88ccb3 ]
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 c344ef36dbc2fe920ec7291b68b11fe867a2c8f6 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: a0575b4add21 ("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>
commit 6b3d14b7f9b1acaf7303d8499836bf78ee9c470c upstream.
This enables the mute and mic-mute LEDs on the HP Envy X360 13-ay0xxx
convertibles.
The quirk 'ALC245_FIXUP_HP_X360_MUTE_LEDS' already exists and is now
enabled for this device.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216197
Signed-off-by: Tom Jason Schwanke <tom@catboys.cloud>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/651b26e9-e86b-45dd-aa90-3e43d6b99823@catboys.cloud
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 51d976079976c800ef19ed1b542602fcf63f0edb ]
These models use 2xCS35L41amps with HDA using SPI and I2C.
Models use internal and external boost.
All models require DSD support to be added inside
cs35l41_hda_property.c
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20231218151221.388745-6-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b1b6131bca35a55a69fadc39d51577968fa2ee97 ]
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 99c7bb44f5749373bc01b73af02b50b69bcbf43d ]
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 a0ffa8115e1ea9786b03edc3f431d2f4ef3e7a2e ]
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 c3c8b088949b9ccb88da2f84d3c3cc06580a6a43 ]
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 12e102b1bd22ee00361559d57a5876445bcb2407 ]
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 eb99b1b72a424a79f56c972e0fd7ad01fe93a008 ]
Stop timer in the 'trigger' and 'sync_stop' callbacks since we want
the timer to be stopped before the DMA buffer is released. Otherwise,
it could trigger a kernel panic in some circumstances, for instance
when the DMA buffer is already released but the timer callback is
still running.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Link: https://lore.kernel.org/r/20231206223211.12761-1-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f44de697383fcc9a9a1a78f99e09d1838704b90 ]
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 e238b68e6dc89ddab52bd98216fe5623e94792b1 ]
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 c447636970e3409ac39f0bb8c2dcff6b726f36b0 ]
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 a0575b4add21a243cc3257e75ad913cd5377d5f2 ]
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: 3d1dc8b1030d ("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 31ed8da1c8e5e504710bb36863700e3389f8fc81 ]
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 0376b995bb7a65fb0c056f3adc5e9695ad0c1805 ]
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 3d1dc8b1030df8ca0fdfd4905c88ee10db943bf8 ]
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 347ecf29a68cc8958fbcbd26ef410d07fe9d82f4 ]
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 cdba4301adda7c60a2064bf808e48fccd352aaa9 ]
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 c33fd110424dfcb544cf55a1b312f43fe1918235 ]
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 f8ba14b780273fd290ddf7ee0d7d7decb44cc365 ]
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 31e721fbd194d5723722eaa21df1d14cee7e12b5 ]
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 c1501f2597dd08601acd42256a4b0a0fc36bf302 ]
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 d5c65be34df73fa01ed05611aafb73b440d89e29 ]
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 37e6fd0cebf0b9f71afb38fd95b10408799d1f0b ]
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 0c6498a59fbbcbf3d0a58c282dd6f0bca0eed92a ]
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 7b4c93a50a2ebbbaf656cc4fa6aca74a6166d85b ]
When looking up DMIC blob from the NHLT table and the format is 32 bits,
ignore the vbps matching for 32 bps for DMIC since some NHLT table have
the vbps as 24, some have it as 32.
The DMIC hardware supports only one type of 32 bit sample size, which is
24 bit sampling on the MSB side and bits[1:0] is used for indicating the
channel number.
Signed-off-by: Peter 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: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20231127111658.17275-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7c9caa299335df94ad1c58f70a22f16a540eab60 ]
This patch corrected the speaker and headset mic pin config to the more
appropriate values.
Signed-off-by: Vasiliy Kovalev <kovalev@altlinux.org>
Link: https://lore.kernel.org/r/20231117170923.106822-1-kovalev@altlinux.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b036d8ef3120b996751495ce25994eea58032a98 ]
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 172c88244b5f2d3375403ebb504d407be0fded59 ]
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 1e001206804be3f3d21f4a1cf16e5d059d75643f ]
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 3150b70e944ead909260285dfb5707d0bedcf87b ]
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 38744c3fa00109c51076121c2deb4f02e2f09194 ]
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 f9d378fc68c43fd41b35133edec9cd902ec334ec ]
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 18a434f32fa61b3fda8ddcd9a63d5274569c6a41 upstream.
There is a HP ZBook which using ALC236 codec and need the
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make mute LED
and micmute LED work.
[ confirmed that the new entries are for new models that have no
proper name, so the strings are left as "HP" which will be updated
eventually later -- tiwai ]
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240102024916.19093-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 13a5b21197587a3d9cac9e1a00de9b91526a55e4 upstream.
The HP Pavilion 14 ec1xxx series uses the HP mainboard 8A0F with the
ALC287 codec.
The mute led can be enabled using the already existing
ALC287_FIXUP_HP_GPIO_LED quirk.
Tested on an HP Pavilion ec1003AU
Signed-off-by: Aabish Malik <aabishmalik3337@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231229170352.742261-3-aabishmalik3337@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4e7914eb1dae377b8e6de59c96b0653aacb47646 upstream.
Remove sound controls in hda_unbind to make
module loadable after module unload.
Add a driver specific struct (tas2781_hda) to store
the controls.
This patch depends on patch:
ALSA: hda/tas2781: do not use regcache
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/362aa3e2f81b9259a3e5222f576bec5debfc5e88.1703204848.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6dad45f4d28977bd1948973107cf325d431e5b7e 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>
commit ec1de5c214eb5a892fdb7c450748249d5e2840f5 upstream.
Currently, cur_prog/cur_conf remains at the default value (-1), while
program 0 has been loaded into the amplifiers.
In the playback hook, tasdevice_tuning_switch tries to restore the
cur_prog/cur_conf. In the runtime_resume/system_resume,
tasdevice_prmg_load tries to load the cur_prog as well.
Set cur_prog and cur_conf to 0 if available in the firmware.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/038add0bdca1f979cc7abcce8f24cbcd3544084b.1702596646.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 48d6b91798a6694fdd6edb62799754b9d3fe0792 upstream.
Increase the quirk delay from 2 seconds to 4 seconds. This reflects a
change in the Windows driver in which the delay was increased to about
3.7 seconds. The larger delay fixes an issue where the device fails to
work unless it was powered up early during boot.
Also clarify in the quirk comment that the quirk is only applied to
older devices (USB ID 07fd:0008).
Signed-off-by: Jeremie Knuesel <knuesel@gmail.com>
Suggested-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=211975
Link: https://lore.kernel.org/r/20231217112243.33409-1-knuesel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 8f0f01647550daf9cd8752c1656dcb0136d79ce1 ]
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 025222a9d6d25eee2ad9a1bb5a8b29b34b5ba576 ]
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: 15be353d55f9 ("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>
commit 315deab289924c83ab1ded50022e8db95d6e428b upstream.
Calling component_add starts loading the firmware, the callback function
writes the program to the amplifiers. If the module resets the
amplifiers after component_add, it happens that one of the amplifiers
does not work because the reset and program writing are interleaving.
Call tas2781_reset before component_add to ensure reliable
initialization.
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/4d23bf58558e23ee8097de01f70f1eb8d9de2d15.1702511246.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6c6fa2641402e8e753262fb61ed9a15a7cb225ad upstream.
If the module can load the RCA but not the firmware binary, it will call
the cleanup functions. Then unloading the module causes general
protection fault due to double free.
Do not call the cleanup functions in tasdev_fw_ready.
general protection fault, probably for non-canonical address
0x6f2b8a2bff4c8fec: 0000 [#1] PREEMPT SMP NOPTI
Call Trace:
<TASK>
? die_addr+0x36/0x90
? exc_general_protection+0x1c5/0x430
? asm_exc_general_protection+0x26/0x30
? tasdevice_config_info_remove+0x6d/0xd0 [snd_soc_tas2781_fmwlib]
tas2781_hda_unbind+0xaa/0x100 [snd_hda_scodec_tas2781_i2c]
component_unbind+0x2e/0x50
component_unbind_all+0x92/0xa0
component_del+0xa8/0x140
tas2781_hda_remove.isra.0+0x32/0x60 [snd_hda_scodec_tas2781_i2c]
i2c_device_remove+0x26/0xb0
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/1a0885c424bb21172702d254655882b59ef6477a.1702510018.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 33071422714a4c9587753b0ccc130ca59323bf42 upstream.
The code does not properly check whether the calibration variable is
available in the EFI. If it is not available, it causes a NULL pointer
dereference.
Check the return value of the first get_variable call also.
BUG: kernel NULL pointer dereference, address: 0000000000000000
Call Trace:
<TASK>
? __die+0x23/0x70
? page_fault_oops+0x171/0x4e0
? srso_alias_return_thunk+0x5/0x7f
? schedule+0x5e/0xd0
? exc_page_fault+0x7f/0x180
? asm_exc_page_fault+0x26/0x30
? crc32_body+0x2c/0x120
? tas2781_save_calibration+0xe4/0x220 [snd_hda_scodec_tas2781_i2c]
tasdev_fw_ready+0x1af/0x280 [snd_hda_scodec_tas2781_i2c]
request_firmware_work_func+0x59/0xa0
Fixes: 5be27f1e3e ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/f1f6583bda918f78556f67d522ca7b3b91cebbd5.1702251102.git.soyer@irl.hu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9b726bf6ae11add6a7a52883a21f90ff9cbca916 upstream.
The HP laptop 15-db0403ng uses the ALC236 codec and controls the mute
LED using COEF 0x07 index 1.
Sound card subsystem: Hewlett-Packard Company Device [103c:84ae]
Use the existing quirk for this model.
Signed-off-by: Hartmut Knaack <knaack.h@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/e61815d0-f1c7-b164-e49d-6ca84771476a@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 924f5ca2975b2993ee81a7ecc3c809943a70f334 upstream.
On ASUSTeK Z170M PLUS and Z170 PRO GAMING systems, the display codec
pins are not registered properly without the force-connect quirk. The
codec will report only one pin as having external connectivity, but i915
finds all three connectors on the system, so the two drivers are not
in sync.
Issue found with DRM igt-gpu-tools test kms_hdmi_inject@inject-audio.
Link: https://gitlab.freedesktop.org/drm/intel/-/issues/9801
Cc: Ville Syrjälä <ville.syrjala@linux.intel.com>
Cc: Jani Saarinen <jani.saarinen@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231208132127.2438067-3-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3b1ff57e24a7bcd2e2a8426dd2013a80d1fa96eb upstream.
Add one more older NUC model that requires quirk to force all pins to be
connected. The display codec pins are not registered properly without
the force-connect quirk. The codec will report only one pin as having
external connectivity, but i915 finds all three connectors on the
system, so the two drivers are not in sync.
Issue found with DRM igt-gpu-tools test kms_hdmi_inject@inject-audio.
Link: https://gitlab.freedesktop.org/drm/igt-gpu-tools/-/issues/3
Cc: Ville Syrjälä <ville.syrjala@linux.intel.com>
Cc: Jani Saarinen <jani.saarinen@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231208132127.2438067-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 716d4e5373e9d1ae993485ab2e3b893bf7104fb1 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 fb9ad24485087e0f00d84bee7a5914640b2b9024 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 8804fa04a492f4176ea407390052292912227820 upstream.
The Framework 16" laptop has the same controller as other Framework
models. Apply the presence detection quirk.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231206193927.2996-1-mario.limonciello@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 33038efb64f7576bac635164021f5c984d4c755f upstream.
The Framework Laptop 13 (AMD Ryzen 7040Series) has an ALC295 with
a disconnected or faulty headset mic presence detect similar to the
previous models. It works with the same quirk chain as
309d7363ca. This model has a VID:PID
of f111:0006.
Signed-off-by: Tim Bosse <flinn@timbos.se>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231206142629.388615-1-flinn@timbos.se
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6f7e4664e597440dfbdb8b2931c561b717030d07 upstream.
Lenovo M90 Gen5 is equipped with ALC897, and it needs
ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work.
Signed-off-by: Bin Li <bin.li@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231204100450.642783-1-bin.li@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit cd14dedf15be432066e63783c63d650f2800cd48 upstream.
XPS 9530 has 2 tweeters and 2 subwoofers powered by CS35L41 amplifier, SPI
connected. For subwoofers to work, it requires both to enable amplifier
support, and to enable output to subwoofers via 0x17 quirk (similalry to
XPS 9510/9520).
Signed-off-by: Aleksandrs Vinarskis <alex.vinarskis@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231203233006.100558-1-alex.vinarskis@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bbb8e71965c3737bdc691afd803a34bfd61cfbeb upstream.
These values mirror those of the Pioneer DJM-250MK2 as the channel layout
appears identical based on my observations. This duplication could be removed in
later contributions if desired.
Signed-off-by: Sarah Grant <s@srd.tw>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231201181654.5058-1-s@srd.tw
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b24e3590c94ab0aba6e455996b502a83baa5c31c 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 29046a78a3c0a1f8fa0427f164caa222f003cf5b ]
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 a2f35ed1d237c459100adb0c39bb811d7f170977 ]
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 14e8442e0789598514f3c9de014950de9feda7a4 ]
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>
commit baaacbff64d9f34b64f294431966d035aeadb81c upstream.
This platform need to set Mic VREF to 100%.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/0916af40f08a4348a3298a9a59e6967e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a337c355719c42a6c5b67e985ad753590ed844fb upstream.
It's been reported that the runtime PM on KONTRON SinglePC (PCI SSID
1734:1232) caused a stall of playback after a bunch of invocations.
(FWIW, this looks like an timing issue, and the stall happens rather
on the controller side.)
As a workaround, disable the default power-save on this platform.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231130151321.9813-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 61cbc08fdb04fd445458b0f4cba7e6929afdfaef ]
These ASUS Zenbook laptops use Realtek HDA codec combined with
2xCS35L41 Amplifiers using SPI or I2C with External Boost or
Internal Boost.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231115162116.494968-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6ae90e906aed727759b88eb2b000fcdc8fcd94a3 ]
Proposed patch fixes initialization of CSC3551 on the UM5302LA laptop.
Patching DSDT table is not required since ASUS did added _DSD entry.
Nothing new introduced but reused work started by Stefan B.
Currently there is no official firmware available for 10431A83 on
cirrus git unfortunately.
For testing used 104317f3 (which is also seems on i2c bus):
$ cd /lib/firmware/cirrus/ && \
for fw in $(find ./ -name '*104317f3*'); do newfw=$(echo $fw | sed 's/104317f3/10431a83/g'); echo echo "$fw -> $newfw"; ln -s $f $newfw; done
With the patch applied to 6.6.0 and obviously symlinks to 104317F3 FW,
speakers works and to my susrprise they sound quite good and loud
without distortion.
Probably confirmation from cirrus team is needed on firmware.
Signed-off-by: Vitalii Torshyn <vitaly.torshyn@gmail.com>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218119
Link: https://lore.kernel.org/r/CAHiQ-bCMPpCJ8eOYAaVVoqGkFixS1qTgSS4xfbZvL4oZV9LYew@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 61cbc08fdb04 ("ALSA: hda/realtek: Add quirks for ASUS 2024 Zenbooks")
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 5d639b60971f003d3a9b2b31f8ec73b0718b5d57 upstream.
These HP laptops use Realtek HDA codec combined with 2 or 4 CS35L41
Amplifiers using SPI with Internal Boost.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231115162116.494968-3-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b944aa9d86d5f782bfe5e51336434c960304839c upstream.
HP 255 G10 has a mute LED that can be made to work using quirk
ALC236_FIXUP_HP_MUTE_LED_COEFBIT2.
Enable already existing quirk - at correct line to keep order
Signed-off-by: Matus Malych <matus@malych.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231114133524.11340-1-matus@malych.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4b21a669ca21ed8f24ef4530b2918be5730114de upstream.
Add ALC295 to pin fall back table.
Remove 5 pin quirks for Dell ALC295.
ALC295 was only support MIC2 for external MIC function.
ALC295 assigned model "ALC269_FIXUP_DELL1_MIC_NO_PRESENCE" for pin
fall back table.
It was assigned wrong model. So, let's remove it.
Fixes: fbc571290d ("ALSA: hda/realtek - Fixed Headphone Mic can't record on Dell platform")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/7c1998e873834df98d59bd7e0d08c72e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c7a60651953359f98dbf24b43e1bf561e1573ed4 upstream.
As reported recently, ALSA core info helper may cause a deadlock at
the forced device disconnection during the procfs operation.
The proc_remove() (that is called from the snd_card_disconnect()
helper) has a synchronization of the pending procfs accesses via
wait_for_completion(). Meanwhile, ALSA procfs helper takes the global
mutex_lock(&info_mutex) at both the proc_open callback and
snd_card_info_disconnect() helper. Since the proc_open can't finish
due to the mutex lock, wait_for_completion() never returns, either,
hence it deadlocks.
TASK#1 TASK#2
proc_reg_open()
takes use_pde()
snd_info_text_entry_open()
snd_card_disconnect()
snd_info_card_disconnect()
takes mutex_lock(&info_mutex)
proc_remove()
wait_for_completion(unused_pde)
... waiting task#1 closes
mutex_lock(&info_mutex)
=> DEADLOCK
This patch is a workaround for avoiding the deadlock scenario above.
The basic strategy is to move proc_remove() call outside the mutex
lock. proc_remove() can work gracefully without extra locking, and it
can delete the tree recursively alone. So, we call proc_remove() at
snd_info_card_disconnection() at first, then delete the rest resources
recursively within the info_mutex lock.
After the change, the function snd_info_disconnect() doesn't do
disconnection by itself any longer, but it merely clears the procfs
pointer. So rename the function to snd_info_clear_entries() for
avoiding confusion.
The similar change is applied to snd_info_free_entry(), too. Since
the proc_remove() is called only conditionally with the non-NULL
entry->p, it's skipped after the snd_info_clear_entries() call.
Reported-by: Shinhyung Kang <s47.kang@samsung.com>
Closes: https://lore.kernel.org/r/664457955.21699345385931.JavaMail.epsvc@epcpadp4
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20231109141954.4283-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 72151ad0cba8a07df90130ff62c979520d71f23b 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 805ce81826c896dd3c351a32814b28557f9edf54 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 f0220575e65abe09c09cd17826a3cdea76e8d58f 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 fbb74e56378d8306f214658e3d525a8b3f000c5a ]
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 2cb54788393134d8174ee594002baae3ce52c61e ]
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 f93dc90c2e8ed664985e366aa6459ac83cdab236 ]
While AudioDSP drivers assign streams exclusively of HOST or LINK type,
nothing blocks a user to attempt to assign a COUPLED stream. As
supplied substream instance may be a stub, what is the case when
code-loading, such scenario ends with null-ptr-deref.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20231006102857.749143-2-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1c48fd6bbe788458e3685fea74bdb3cb148ff93 ]
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 1a1c3d794ef65ef2978c5e65e1aed3fe6f014e90 ]
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 d8b387544ff4d02eda1d1839a0c601de4b037c33 ]
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 ba2de401d32625fe538d3f2c00ca73740dd2d516 ]
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 d601bb78f06b9e3cbb52e6b87b88add9920a11b6 ]
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>
[ Upstream commit f0d9da19d7de9e845e7a93a901c4b9658df6b492 ]
Dell new platform support dual speaker. But BIOS verb table only show one speaker.
It will fill verb table for second speaker. Then bind with CS AMP model.
Fixes: de90f5165b ("ALSA: hda/realtek: Add support for DELL Oasis 13/14/16 laptops")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/4dd390a77bf742b8a518ac2deee00b0f@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4bdcbc31ad2112385ad525b28972c45015e6ad70 ]
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 15be353d55f9e12e34f9a819f51eb41fdef5eda8 ]
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 004fc58edea6f00db9ad07b40b882e8d976f7a54 ]
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 1a3b7eab8500a6b923f7b62cc8aa4d832c7dfb3e ]
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 9e630efb5a4af56fdb15aa10405f5cfd3f5f5b83 ]
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 28809aaeabdf2c01ffe597553146527d1fba3589 ]
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 f549a82aff57865c47b5abd17336b23cd9bb2d2c ]
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 fbfe616ad40c06d68b83b657a94cd2e709dda37b ]
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 4a221b2e3340f4a3c2b414c46c846a26c6caf820 ]
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 3746284c233d5cf5f456400e61cd4a46a69c6e8c ]
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 3efcb471f871cc095841d411f98c593228ecbac6 ]
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 85a1bf86fac0c195929768b4e92c78cad107523b ]
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().
Fixes: 1873ebd30c ("ALSA: hda: cs35l41: Support Hibernation during Suspend")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230907171010.1447274-11-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Geoffrey D. Bennett