Takashi Iwai [Mon, 20 Jan 2020 10:41:27 +0000 (11:41 +0100)]
ALSA: hda: Apply aligned MMIO access only conditionally
It turned out that the recent simplification of HD-audio bus access
helpers caused a regression on the virtual HD-audio device on QEMU
with ARM platforms. The driver got a CORB/RIRB timeout and couldn't
probe any codecs.
The essential difference that caused a problem was the enforced
aligned MMIO accesses by simplification. Since snd-hda-tegra driver
is enabled on ARM, it enables CONFIG_SND_HDA_ALIGNED_MMIO, which makes
the all HD-audio drivers using the aligned MMIO accesses. While this
is mandatory for snd-hda-tegra, it seems that snd-hda-intel on ARM
gets broken by this access pattern.
For addressing the regression, this patch introduces a new flag,
aligned_mmio, to hdac_bus object, and applies the aligned MMIO only
when this flag is set. This change affects only platforms with
CONFIG_SND_HDA_ALIGNED_MMIO set, i.e. mostly only for ARM platforms.
Unfortunately the patch became a big bigger than it should be, just
because the former calls didn't take hdac_bus object in the argument,
hence we had to extend the call patterns.
Takashi Iwai [Wed, 15 Jan 2020 10:00:35 +0000 (11:00 +0100)]
ALSA: hda/analog - Minor optimization for SPDIF mux connections
AD HD-audio codec driver has a few code lines invoking
snd_get_num_conns() and using its return value as the array index
without checking. This is basically safe in all those places; at the
second and later calls snd_get_num_conns() returns the value cached
from the first invocation, hence the value is always consistent.
However, it looks a bit confusing as if a lack of the proper check.
This patch introduces a new field num_smux_conns in ad198x_spec for
simplifying the code. Now we store and refer to the value more
locally without invoking the extra function at each time.
Takashi Iwai [Thu, 16 Jan 2020 13:14:26 +0000 (14:14 +0100)]
Merge tag 'asoc-fix-v5.5-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.5
This is mostly driver specific fixes, plus an error handling fix
in the core. There is a rather large diffstat for the stm32 SAI
driver, this is a very large but mostly mechanical update which
wraps every register access in the driver to allow a fix to the
locking which avoids circular locks, the active change is much
smaller and more reasonably sized.
Alexander Tsoy [Wed, 15 Jan 2020 15:13:58 +0000 (18:13 +0300)]
ALSA: usb-audio: add implicit fb quirk for MOTU M Series
This fixes crackling sound during playback.
Further note: MOTU is known for reusing Product IDs for different
devices or different generations of the device (e.g. MicroBook
I/II/IIc shares a single Product ID). This patch was only tested with
M4 audio interface, but the same Product ID is also used by M2. Hope
it will work for M2 as well.
Takashi Iwai [Wed, 15 Jan 2020 20:37:33 +0000 (21:37 +0100)]
ALSA: seq: Fix racy access for queue timer in proc read
snd_seq_info_timer_read() reads the information of the timer assigned
for each queue, but it's done in a racy way which may lead to UAF as
spotted by syzkaller.
This patch applies the missing q->timer_mutex lock while accessing the
timer object as well as a slight code change to adapt the standard
coding style.
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was
checking for there to be at least one altsetting but then went on to
access the second one, which may not exist.
This could lead to random slab data being used to initialise the sync
endpoint in snd_usb_add_endpoint().
Fixes: c75a8a7ae565 ("ALSA: snd-usb: add support for implicit feedback") Fixes: ca10a7ebdff1 ("ALSA: usb-audio: FT C400 sync playback EP to capture EP") Fixes: 5e35dc0338d8 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204") Fixes: 17f08b0d9aaf ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II") Fixes: 103e9625647a ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk") Cc: stable <[email protected]> # 3.5 Signed-off-by: Johan Hovold <[email protected]> Link: https://lore.kernel.org/r/[email protected] Signed-off-by: Takashi Iwai <[email protected]>
ALSA: hda: patch_hdmi: remove warnings with empty body
make W=1 reports the following warnings, fix as suggested
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_non_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:824:3: warning: suggest braces around empty
body in an ‘if’ statement [-Wempty-body]
824 | ;
| ^
sound/pci/hda/patch_hdmi.c:826:3: warning: suggest braces around empty
body in an ‘if’ statement [-Wempty-body]
826 | ;
| ^
Stephan Gerhold [Sun, 5 Jan 2020 10:27:53 +0000 (11:27 +0100)]
ASoC: msm8916-wcd-digital: Reset RX interpolation path after use
For some reason, attempting to route audio through QDSP6 on MSM8916
causes the RX interpolation path to get "stuck" after playing audio
a few times. In this situation, the analog codec part is still working,
but the RX path in the digital codec stops working, so you only hear
the analog parts powering up. After a reboot everything works again.
So far I was not able to reproduce the problem when using lpass-cpu.
The downstream kernel driver avoids this by resetting the RX
interpolation path after use. In mainline we do something similar
for the TX decimator (LPASS_CDC_CLK_TX_RESET_B1_CTL), but the
interpolator reset (LPASS_CDC_CLK_RX_RESET_CTL) got lost when the
msm8916-wcd driver was split into analog and digital.
Fix this problem by adding the reset to
msm8916_wcd_digital_enable_interpolator().
Stephan Gerhold [Sat, 11 Jan 2020 16:40:04 +0000 (17:40 +0100)]
ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1
MIC BIAS Internal1 is broken at the moment because we always
enable the internal rbias resistor to the TX2 line (connected to
the headset microphone), rather than enabling the resistor connected
to TX1.
Move the RBIAS code to pm8916_wcd_analog_enable_micbias_int1/2()
to fix this.
Stephan Gerhold [Sat, 11 Jan 2020 16:40:03 +0000 (17:40 +0100)]
ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1
MIC BIAS External1 sets pm8916_wcd_analog_enable_micbias_ext1()
as event handler, which ends up in pm8916_wcd_analog_enable_micbias_ext().
But pm8916_wcd_analog_enable_micbias_ext() only handles the POST_PMU
event, which is not specified in the event flags for MIC BIAS External1.
This means that the code in the event handler is never actually run.
Set SND_SOC_DAPM_POST_PMU as the only event for the handler to fix this.
Kai Vehmanen [Fri, 10 Jan 2020 23:57:51 +0000 (17:57 -0600)]
ASoC: hdac_hda: Fix error in driver removal after failed probe
In case system has multiple HDA codecs, and codec probe fails for
at least one but not all codecs, driver will end up cancelling
a non-initialized timer context upon driver removal.
In case system has multiple HDA controllers, it can happen that
same HDA codec driver is used for codecs of multiple controllers.
In this case, SOF may fail to probe the HDA driver and SOF
initialization fails.
SOF HDA code currently relies that a call to request_module() will
also run device matching logic to attach driver to the codec instance.
However if driver for another HDA controller was already loaded and it
already loaded the HDA codec driver, this breaks current logic in SOF.
In this case the request_module() SOF does becomes a no-op and HDA
Codec driver is not attached to the codec instance sitting on the HDA
bus SOF is controlling. Typical scenario would be a system with both
external and internal GPUs, with driver of the external GPU loaded
first.
Fix this by adding similar logic as is used in legacy HDA driver
where an explicit device_attach() call is done after request_module().
Also add logic to propagate errors reported by device_attach() back
to caller. This also works in the case where drivers are not built
as modules.
Bard liao [Fri, 10 Jan 2020 23:57:48 +0000 (17:57 -0600)]
ASoC: SOF: Intel: lower print level to dbg if we will reinit DSP
We will reinit DSP in a loop when it fails to initialize the first
time, as recommended. So, it is not an error before we finally give
up. And reorder the trace to make it more readable.
Takashi Iwai [Thu, 9 Jan 2020 09:01:04 +0000 (10:01 +0100)]
ALSA: hda: Manage concurrent reg access more properly
In the commit 8e85def5723e ("ALSA: hda: enable regmap internal
locking"), we re-enabled the regmap lock due to the reported
regression that showed the possible concurrent accesses. It was a
temporary workaround, and there are still a few opened races even
after the revert. In this patch, we cover those still opened windows
with a proper mutex lock and disable the regmap internal lock again.
First off, the patch introduces a new snd_hdac_device.regmap_lock
mutex that is applied for each snd_hdac_regmap_*() call, including
read, write and update helpers. The mutex is applied carefully so
that it won't block the self-power-up procedure in the helper
function. Also, this assures the protection for the accesses without
regmap, too.
The snd_hdac_regmap_update_raw() is refactored to use the standard
regmap_update_bits_check() function instead of the open-code. The
non-regmap case is still open-coded but it's an easy part. The all
read and write operations are in the single mutex protection, so it's
now race-free.
In addition, a couple of new helper functions are added:
snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both
are called from HD-audio legacy driver. The former is to initialize
the given verb bits but only once when it's not initialized yet. Due
to this condition, the function invokes regcache_cache_only(), and
it's now performed inside the regmap_lock (formerly it was racy) too.
The latter function is for simply invoking regcache_sync() inside the
regmap_lock, which is called from the codec resume call path.
Along with that, the HD-audio codec driver code is slightly modified /
simplified to adapt those new functions.
And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are
rewritten with the helper macro. It's just for simplification because
the code logic is identical among all those functions.
Alexander Tsoy [Sun, 12 Jan 2020 10:23:58 +0000 (13:23 +0300)]
ALSA: usb-audio: Add boot quirk for MOTU M Series
Add delay to make sure that audio urbs are not sent too early.
Otherwise the device hangs. Windows driver makes ~2s delay, so use
about the same time delay value.
snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which
is an overkill. Thus a quirk that is called only once is implemented.
Also send two vendor-specific control messages before and after
the delay. This behaviour is blindly copied from the Windows driver.
Takashi Sakamoto [Mon, 13 Jan 2020 08:46:30 +0000 (17:46 +0900)]
ALSA: dice: add support for Alesis MasterControl
Alesis MasterControl was shipped 2009 and already discontinued. This model
consists of:
* TSB41AB2 for physical layer of IEEE 1394
* WaveFront Dice II STD for link layer and protocol implementation
* FreeScale DSPB56374AE
Although the firmware of this model can respond against read transaction
to address space for TCAT extension protocol, the content is not valid
for protocol extension. This results in sound card without any PCM/MIDI
interfaces.
Takashi Sakamoto [Mon, 13 Jan 2020 08:46:29 +0000 (17:46 +0900)]
ALSA: dice: loosen stream format check for MIDI conformant data channel
ALSA dice driver expects devices to multiplex MIDI messages into first
port of isochronous communication. Actually devices perform for it.
However, check of stream format is invalid for second port of isochronous
communication. As a result, when the device supports two ports for
isochronous communication and the stream format is hard-coded, ALSA
dice driver fails to start packet streaming.
This commit loosens stream format check for MIDI conformant data channel.
Takashi Sakamoto [Mon, 13 Jan 2020 07:34:18 +0000 (16:34 +0900)]
ALSA: oxfw: fix for Stanton SCS.1d
Stanton SCS.1d uses Oxford Semiconductor FW 971 ASIC (FW971) for
communication. Although the unit is bound to ALSA oxfw driver, the instance
of sound card can not be added due to its quirk of plug information. This
bug was added when snd-scs1x is merged into snd-oxfw at commit 9e2004f9cedf ("ALSA: oxfw: obsolete scs1x module").
This commit fixes the driver for the quirk. In cases that the unit returns
NOT IMPLEMENTED for some AV/C commands, the sound card is added without any
PCM/MIDI interfaces for packet streaming. For SCS.1d, model dependent
operation adds MIDI interface and applications can use it to operate
according to HSS1394 protocol from reverse-engineering work by Sean M.
Pappalardo.
Plug Control Register (PCR) has information that the unit has a pair of
plugs for isochronous communication:
Takashi Sakamoto [Mon, 13 Jan 2020 07:34:17 +0000 (16:34 +0900)]
ALSA: oxfw: don't add MIDI/PCM interface when packet streaming is unavailable
Stanton SCS.1d doesn't support packet streaming even if it has plugs for
isochronous communication.
This commit is a preparation for this case. The 'has_input' member is
added to specific structure, and MIDI/PCM interfaces are not added when
the member is false.
Takashi Sakamoto [Mon, 13 Jan 2020 07:34:16 +0000 (16:34 +0900)]
ALSA: oxfw: use ENXIO for not-supported cases
When AV/C command returns 'NOT IMPLEMENTED' status in its response, ALSA
oxfw driver uses ENOSYS as error code. However, it's expected just to be
used for missing system call number.
sound/usb/mixer_quirks.c: In function ‘snd_microii_controls_create’:
sound/usb/mixer_quirks.c:1694:2: warning: ‘static’ is not at beginning
of declaration [-Wold-style-declaration]
1694 | const static usb_mixer_elem_resume_func_t resume_funcs[] = {
| ^~~~~
ALSA: hda: patch_realtek: fix empty macro usage in if block
GCC reports the following warning with W=1
sound/pci/hda/patch_realtek.c: In function ‘alc269_suspend’:
sound/pci/hda/patch_realtek.c:3616:29: warning: suggest braces around
empty body in an ‘if’ statement [-Wempty-body]
3616 | alc5505_dsp_suspend(codec);
| ^
sound/pci/hda/patch_realtek.c: In function ‘alc269_resume’:
sound/pci/hda/patch_realtek.c:3651:28: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
3651 | alc5505_dsp_resume(codec);
| ^
This is a classic macro problem and can indeed lead to bad program
flows.
Takashi Iwai [Thu, 9 Jan 2020 08:20:00 +0000 (09:20 +0100)]
ALSA: hda: Rename back to dmic_detect option
We've got quite a few bug reports showing the SOF driver being loaded
unintentionally recently, and the reason seems to be that users didn't
know the module option change: with the recent kernel, a new option
dsp_driver=1 has to be passed to a new module snd-intel-dspcfg
instead of snd_hda_intel.dmic_detect=0 option.
That is, actually there are two tricky things here:
- We changed the whole detection in another module and another
option semantics.
- The existing option for skipping the DSP probe was also renamed.
For avoiding the confusion and giving user more hint, this patch
reverts the renamed option dsp_driver back to dmic_detect for
snd-hda-intel module, and show the warning about the module option
change when the non-default value is passed.
Olivier Moysan [Fri, 10 Jan 2020 13:11:31 +0000 (14:11 +0100)]
ASoC: stm32: dfsdm: fix 16 bits record
In stm32_afsdm_pcm_cb function, the transfer size is provided in bytes.
However, samples are copied as 16 bits words from iio buffer.
Divide by two the transfer size, to copy the right number of samples.
Olivier Moysan [Thu, 9 Jan 2020 08:32:54 +0000 (09:32 +0100)]
ASoC: stm32: sai: fix possible circular locking
In current driver, locks can be taken as follows:
- Register access: take a lock on regmap config and then on clock.
- Master clock provider: take a lock on clock and then on regmap config.
This can lead to the circular locking summarized below.
Remove peripheral clock management through regmap framework, and manage
peripheral clock in driver instead. On register access, lock on clock
is taken first, which allows to avoid possible locking issue.
Kai Vehmanen [Wed, 8 Jan 2020 18:08:56 +0000 (20:08 +0200)]
ALSA: hda: enable regmap internal locking
This reverts commit 42ec336f1f9d ("ALSA: hda: Disable regmap
internal locking").
Without regmap locking, there is a race between snd_hda_codec_amp_init()
and PM callbacks issuing regcache_sync(). This was caught by
following kernel warning trace:
YueHaibing [Wed, 8 Jan 2020 12:58:03 +0000 (20:58 +0800)]
ALSA: pci: echoaudio: remove set but not used variable 'chip'
sound/pci/echoaudio/echoaudio.c: In function snd_echo_mixer_info:
sound/pci/echoaudio/echoaudio.c:1233:20: warning: variable chip set but not used [-Wunused-but-set-variable]
sound/pci/echoaudio/echoaudio.c: In function 'snd_echo_vmixer_info':
sound/pci/echoaudio/echoaudio.c:1300:20: warning: variable 'chip' set but not used [-Wunused-but-set-variable]
commit e67c3f0fd44c ("ALSA: pci: echoaudio: remove usage
of dimen menber of elem_value structure") left behind this
unused variable.
Kailang Yang [Wed, 8 Jan 2020 08:47:56 +0000 (16:47 +0800)]
ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen
Add quirk to ALC285_FIXUP_SPEAKER2_TO_DAC1, which is the same fixup
applied for X1 Carbon 7th gen in commit d2cd795c4ece ("ALSA: hda -
fixup for the bass speaker on Lenovo Carbon X1 7th gen").
Takashi Iwai [Tue, 7 Jan 2020 07:09:56 +0000 (08:09 +0100)]
ASoC: Fix NULL dereference at freeing
When an ASoC driver with pcm_destruct component ops is freed before
the PCM object instantiation (e.g. deferring the probe), it hits an
Oops at snd_soc_pcm_component_free() that calls the pcm_destruct ops
unconditionally.
Fix it by adding a NULL-check of rtd->pcm before calling callbacks.
When a quirk for the Irbis NB41 netbook was added, to override the defaults
for this device, I forgot to add/keep the BYT_CHT_ES8316_SSP0 part of the
defaults, completely breaking audio on this netbook.
This commit adds the BYT_CHT_ES8316_SSP0 flag to the Irbis NB41 netbook
quirk, making audio work again.
Takashi Iwai [Mon, 6 Jan 2020 16:39:15 +0000 (17:39 +0100)]
Merge tag 'asoc-fix-v5.5-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.5
More fixes that have been collected, nothing super remarkable here - the
few core fixes are mainly error handling related as are many of the
driver fixes.
Takashi Iwai [Sun, 5 Jan 2020 14:48:23 +0000 (15:48 +0100)]
ALSA: sh: Fix compile warning wrt const
A long-standing compile warning was seen during build test:
sound/sh/aica.c: In function 'load_aica_firmware':
sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
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