Fiero SC-01 is a USB sound card with two mono inputs and a single
stereo output. The inputs are composed into a single stereo stream.
The device uses a vendor-provided driver on Windows and does not work
at all without it. The driver mostly provides ASIO functionality, but
also alters the way the sound card is queried for sample rates and
clocks.
ALSA queries those failing with an EPIPE (same as Windows 10 does).
Presumably, the vendor-provided driver does not query it at all, simply
matching by VID:PID. Thus, I consider this a buggy firmware and adhere
to a set of fixed endpoint quirks instead.
The soundcard has an internal clock. Implicit feedback mode is required
for the playback.
I have updated my device to v1.1.0 from a Windows 10 VM using a vendor-
provided binary prior to the development, hoping for it to just begin
working. The device provides no obvious way to downgrade the firmware,
and regardless, there's no binary available for v1.0.0 anyway.
Thus, I will be getting another unit to extend the patch with support
for that. Expected to be a simple copy-paste of the existing one,
though.
There were no previous reports of that device in context of Linux
anywhere. Other issues have been reported though, but that's out of the
scope.
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-1-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
ASoC: Fixes for v5.19
A few more fixes for v5.19 which came in during the second half of the
merge window, again nothing that's really remarkable outside of the
individual drivers.
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: 7b0efea4ba ("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>
When ep_idx is already non-zero, it means usually a capture stream
that is set up explicity by a fixed-format quirk, and applying the
check for generic (non-implicit-fb) sync EPs might hit incorrectly,
resulting in a bogus sync endpoint for the capture stream.
This patch adds a check for the ep_idx and skip if it's a secondary
endpoint. It's a part of the fixes for regressions on Saffire 6.
Fixes: 7b0efea4ba ("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/20220606160910.6926-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
This code is supposed to propagate errors from simple_write_to_buffer()
or return -EFAULT if "size != count". However "size" needs to be signed
for the code to work correctly and the case where "size == 0" is not
handled correctly.
Fixes: 066c67624d ("ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages")
Fixes: 2f0b1b013b ("ASoC: SOF: debug: Add support for IPC message injection")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/Yph+Cd+JrfOH0i7z@kili
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Charles Keepax <ckeepax@opensource.cirrus.com>:
Mostly the usage of the SX controls seems to match the lowest gain
value + number of gain levels expected. The one notable exception
there being cs53l30 as David noted. However, there are a couple of
other places where the minimum value/TLVs are slightly incorrectly
specified.
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>
ASoC: Fixes for v5.19
A few more fixes that came in during the merge window - nothing
huge here, there is one core fix for DPCM from Pierre but mostly
driver changes.
Follow the pattern used by other selftests like memfd and fall back on the
standard toolchain options to build with a system installed alsa-lib if
we don't get anything from pkg-config. This reduces our build dependencies
a bit in the common case while still allowing use of pkg-config in case
there is a need for it.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20220531151337.2933810-1-broonie@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Maris found out that the quirk for TEAC devices to work around the
clock setup is needed to apply only when the base clock is changed,
e.g. from 48000-based clocks (48000, 96000, 192000, 384000) to
44100-based clocks (44100, 88200, 176400, 352800), or vice versa,
while switching to another clock with the same base clock doesn't need
the (forcible) interface setup.
This patch implements the optimization for the TEAC clock quirk to
avoid the unnecessary interface re-setup.
Fixes: 5ce0b06ae5 ("ALSA: usb-audio: Workaround for clock setup on TEAC devices")
Reported-by: Maris Abele <maris7abele@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220531130749.30357-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Adam Thomson <DLG-Adam.Thomson.Opensource@dm.renesas.com>:
This series contains 2 small fixes around the AAD part of DA7219, particularly
in relation to jack pole detection on certain active headsets, and tidy up
when a jack is removed.
