[ Upstream commit fc976f5629afb4160ee77798b14a693eac903ffd ]
The struct nhlt_format's fmt_config is a flexible array, it must not be
used as normal array.
When moving to the next nhlt_fmt_cfg we need to take into account the data
behind the ->config.caps (indicated by ->config.size).
The logic of the code also changed: it is no longer saves the _last_
fmt_cfg for all found rates.
Fixes: bc2bd45b1f ("ASoC: Intel: Skylake: Parse nhlt and register clock device")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20220630065638.11183-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1c4f29ec878bbf1cc0a1eb54ae7da5ff98e19641 ]
The register default is 0x28 per the datasheet, and the amp gain field
is supposed to be shifted left by one. With the wrong default, the ALSA
controls lie about the power-up state. With the wrong shift, we get only
half the gain we expect.
Signed-off-by: Hector Martin <marcan@marcan.st>
Fixes: 827ed8a0fa ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-4-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3e99e5697e1f7120b5abc755e8a560b22612d6ed ]
DVC value 0xc8 is -100dB and 0xc9 is mute; this needs to map to
-100.5dB as far as the dB scale is concerned. Fix that and enable
the mute flag, so alsamixer correctly shows the control as
<0 dB .. -100 dB, mute>.
Signed-off-by: Hector Martin <marcan@marcan.st>
Fixes: 827ed8a0fa ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-3-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d1a10f1b48202e2d183cce144c218a211e98d906 ]
Fix setting of FSYNC polarity in case of LEFT_J and DSP_A/B formats.
Do NOT set the SCFG field as was previously done, because that is not
correct and is also in conflict with the "ASI1 Source" control which
sets the same SCFG field!
Also add support for explicit polarity inversion.
Fixes: 827ed8a0fa ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-2-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cd10bb89b0d57bca98eb75e0444854a1c129a14e ]
Make sure there is at least 1 ms delay from reset to first command as
is specified in the datasheet. This is a fix similar to commit
307f31452078 ("ASoC: tas2770: Insert post reset delay").
Fixes: 827ed8a0fa ("ASoC: tas2764: Add the driver for the TAS2764")
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20220630075135.2221-1-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 040e3360af3736348112d29425bf5d0be5b93115 ]
Put the SGTL5000 in a silent/safe state on shutdown/remove, this is
required since the SGTL5000 produces a constant noise on its output
after it is configured and its clock is removed. Without this change
this is happening every time the module is unbound/removed or from
reboot till the clock is enabled again.
The issue was experienced on both a Toradex Colibri/Apalis iMX6, but can
be easily reproduced everywhere just playing something on the codec and
after that removing/unbinding the driver.
Fixes: 9b34e6cc3b ("ASoC: Add Freescale SGTL5000 codec support")
Signed-off-by: Francesco Dolcini <francesco.dolcini@toradex.com>
Reviewed-by: Fabio Estevam <festevam@denx.de>
Link: https://lore.kernel.org/r/20220624101301.441314-1-francesco.dolcini@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 40737057b48f1b4db67b0d766b95c87ba8fc5e03 ]
The .set_jack_detect() codec component callback is invoked during card
registration, which happens when the machine driver is probed.
The issue is that this callback can race with the bus suspend/resume,
and IO timeouts can happen. This can be reproduced very easily if the
machine driver is 'blacklisted' and manually probed after the bus
suspends. The bus and codec need to be re-initialized using pm_runtime
helpers.
Previous contributions tried to make sure accesses to the bus during
the .set_jack_detect() component callback only happen when the bus is
active. This was done by changing the regcache status on a component
remove. This is however a layering violation, the regcache status
should only be modified on device probe, suspend and resume. The
component probe/remove should not modify how the device regcache is
handled. This solution also didn't handle all the possible race
conditions, and the RT700 headset codec was not handled.
This patch tries to resume the codec device before handling the jack
initializations. In case the codec has not yet been initialized,
pm_runtime may not be enabled yet, so we don't squelch the -EACCES
error code and only stop the jack information. When the codec reports
as attached, the jack initialization will proceed as usual.
BugLink: https://github.com/thesofproject/linux/issues/3643
Fixes: 7ad4d237e7 ('ASoC: rt711-sdca: Add RT711 SDCA vendor-specific driver')
Fixes: 899b12542b ('ASoC: rt711: add snd_soc_component remove callback')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20220606203752.144159-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4ddef9c4d70aae0c9029bdec7c3f7f1c1c51ff8c ]
The USB audio device 0db0:a073 based on the Realtek ALC4080 chipset
exposes all playback volume controls as "PCM". This makes
distinguishing the individual functions hard.
The mapping already adopted for device 0db0:419c based on the same
chipset fixes the issue, apply it for this device too.
Signed-off-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/Yl1ykPaGgsFf3SnW@ryzen
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5762f980ca10dcfe5eead7c40d1c34cae61f409b ]
The USB audio device 0db0:419c based on the Realtek ALC4080 chip exposes
all playback volume controls as "PCM". This is makes distinguishing the
individual functions hard.
The added mapping distinguishes all playback volume controls as their
respective function:
- Speaker - for back panel output
- Frontpanel Headphone - for front panel output
- IEC958 - for digital output on the back panel
This clarifies the individual volume control functions for users.
Signed-off-by: Johannes Schickel <lordhoto@gmail.com>
Link: https://lore.kernel.org/r/20220115140257.8751-1-lordhoto@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c8618d65007ba68d7891130642d73e89372101e8 ]
The bug is here:
if (!dai) {
The list iterator value 'dai' will *always* be set and non-NULL
by for_each_component_dais(), so it is incorrect to assume that
the iterator value will be NULL if the list is empty or no element
is found (In fact, it will be a bogus pointer to an invalid struct
object containing the HEAD). Otherwise it will bypass the check
'if (!dai) {' (never call dev_err() and never return -ENODEV;)
and lead to invalid memory access lately when calling
'rt5682_set_bclk1_ratio(dai, factor);'.
To fix the bug, just return rt5682_set_bclk1_ratio(dai, factor);
when found the 'dai', otherwise dev_err() and return -ENODEV;
Cc: stable@vger.kernel.org
Fixes: ebbfabc16d ("ASoC: rt5682: Add CCF usage for providing I2S clks")
Signed-off-by: Xiaomeng Tong <xiam0nd.tong@gmail.com>
Link: https://lore.kernel.org/r/20220327081002.12684-1-xiam0nd.tong@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4045daf0fa87846a27f56329fddad2deeb5ca354 ]
On resume from suspend the following chain of events can happen:
A rt5682_resume() -> mod_delayed_work() for jack_detect_work
B DAPM sequence starts ( DAPM is locked now)
A1. rt5682_jack_detect_handler() scheduled
- Takes both jdet_mutex and calibrate_mutex
- Calls in to rt5682_headset_detect() which tries to take DAPM lock, it
starts to wait for it as B path took it already.
B1. DAPM sequence reaches the "HP Amp", rt5682_hp_event() tries to take
the jdet_mutex, but it is locked in A1, so it waits.
Deadlock.
To solve the deadlock, drop the jdet_mutex, use the jack_detect_work to do
the jack removal handling, move the dapm lock up one level to protect the
most of the rt5682_jack_detect_handler(), but not the jack reporting as it
might trigger a DAPM sequence.
The rt5682_headset_detect() can be changed to static as well.
Fixes: 8deb34a90f063 ("ASoC: rt5682: fix the wrong jack type detected")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20220126100325.16513-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit c5e58c4545a69677d078b4c813b5d10d3481be9c upstream.
The previous cleanup with devres may lead to the incorrect release
orders at the probe error handling due to the devres's nature. Until
we register the card, snd_card_free() has to be called at first for
releasing the stuff properly when the driver tries to manage and
release the stuff via card->private_free().
This patch fixes it by calling snd_card_free() manually on the error
from the probe callback.
Fixes: 5bff69b364 ("ALSA: cs46xx: Allocate resources with device-managed APIs")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Jan Engelhardt <jengelh@inai.de>
Link: https://lore.kernel.org/r/p2p1s96o-746-74p4-s95-61qo1p7782pn@vanv.qr
Link: https://lore.kernel.org/r/20220705152336.350-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ae8b1631561a3634cc09d0c62bbdd938eade05ec upstream.
Both Behringer UMC 202 HD and 404 HD need explicit quirks to enable
the implicit feedback mode and start the playback stream primarily.
The former seems fixing the stuttering and the latter is required for
a playback-only case.
Note that the "clock source 41 is not valid" error message still
appears even after this fix, but it should be only once at probe.
The reason of the error is still unknown, but this seems to be mostly
harmless as it's a one-off error and the driver retires the clock
setup and it succeeds afterwards.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220624101132.14528-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 56ec3e755bd1041d35bdec020a99b327697ee470 upstream.
It turned out that Lenovo shipped two completely different products
with the very same PCI SSID, where both require different quirks;
namely, Lenovo C940 has already the fixup for its speaker
(ALC298_FIXUP_LENOVO_SPK_VOLUME) with the PCI SSID 17aa:3818, while
Yoga Duet 7 has also the very same PCI SSID but requires a different
quirk, ALC287_FIXUP_YOGA7_14TIL_SPEAKERS.
Fortunately, both are with different codecs (C940 with ALC298 and Duet
7 with ALC287), hence we can apply different fixes by checking the
codec ID. This patch implements that special fixup function.
For easier handling, the internal function for applying a specific
fixup entry is exported as __snd_hda_apply_fixup(), so that it can be
called from the codec driver. The rest is simply calling it with a
different fixup ID depending on the codec ID.
Reported-by: Hans de Goede <hdegoede@redhat.com>
Tested-by: nikitashvets@flyium.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/5ca147d1-3a2d-60c6-c491-8aa844183222@redhat.com
Link: https://lore.kernel.org/r/20220614054831.14648-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 5faa0bc69102f3a4c605581564c367be5eb94dfa upstream.
Currently the Conexant codec driver sets up the beep NID after calling
snd_hda_gen_parse_auto_config(). It turned out that this results in
the insufficient setup for the beep control, as the generic parser
handles the fake path in snd_hda_gen_parse_auto_config() only if the
beep_nid is set up beforehand.
For dealing with the beep widget properly, call cx_auto_parse_beep()
before snd_hda_gen_parse_auto_config() call.
Fixes: 51e19ca5f7 ("ALSA: hda/conexant - Clean up beep code")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216152
Link: https://lore.kernel.org/r/20220620104008.1994-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ]
Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value
of the control, meaning that no event is generated to userspace. Fix this
by setting the default return value to 1, the code already exits early with
a return value of 0 if the value is unchanged.
Signed-off-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ]
Currently the put() method for the deemphasis control returns 0 when a new
value is written to the control even if the value changed, meaning events
are not generated. Fix this, skip the work of updating the value when it is
unchanged and then return 1 after having done so.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ]
If the audio CODEC is playing sound when the system is suspended,
it can be left in a state which throws the following error:
wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16
Once this error has occurred, the audio will not work again until rebooted.
Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS.
Signed-off-by: Adam Ford <aford173@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ]
The minimum value for the PGA Volume is given as 0x1A, however the
values from there to 0x19 are all the same volume and this is not
represented in the TLV structure. The number of volumes given is correct
so this leads to all the volumes being shifted. Move the minimum value
up to 0x19 to fix this.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ]
We tried to enable the audio on an imx6sx EVB with the codec nau8822,
after setting the internal PLL fractional parameters, the audio still
couldn't work and the there was no sdma irq at all.
After checking with the section "8.1.1 Phase Locked Loop (PLL) Design
Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to
turn off the PLL before programming fractional parameters and turn on
the PLL after programming.
After this change, the audio driver could record and play sound and
the sdma's irq is triggered when playing or recording.
Cc: David Lin <ctlin0@nuvoton.com>
Cc: John Hsu <kchsu0@nuvoton.com>
Cc: Seven Li <wtli@nuvoton.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e0469d6581aecb0e34e2ec64f39f88e6985cc52f upstream.
Focusrite Saffire 6 has fixed audioformat quirks with multiple
endpoints assigned to a single altsetting. Unfortunately the generic
parser couldn't detect the sync endpoint correctly as the implicit
sync due to the missing EP attribute bits. In the former kernels, it
used to work somehow casually, but it's been broken for a while after
the large code change in 5.11.
This patch cures the regression by the following:
- Allow the static quirk table to provide the sync EP information;
we just need to fill the fields and let the generic parser skipping
parsing if sync_ep is already set.
- Add the sync endpoint information to the entry for Saffire 6.
Fixes: 7b0efea4baf0 ("ALSA: usb-audio: Add missing ep_idx in fixed EP quirks")
Reported-and-tested-by: André Kapelrud <a.kapelrud@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220606160910.6926-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
