Jeremy Soller [Tue, 2 Nov 2021 17:21:04 +0000 (11:21 -0600)]
ALSA: hda/realtek: Headset fixup for Clevo NH77HJQ
On Clevo NH77HJ, NH77HP, and their 15" variants, there is a headset
microphone input attached to 0x19 that does not have a jack detect. In
order to get it working, the pin configuration needs to be set
correctly, and a new ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE fixup is
applied. This is similar to the existing System76 quirk for ALC293, but
for ALC256.
Wang Wensheng [Wed, 3 Nov 2021 03:35:17 +0000 (03:35 +0000)]
ALSA: timer: Fix use-after-free problem
When the timer instance was add into ack_list but was not currently in
process, the user could stop it via snd_timer_stop1() without delete it
from the ack_list. Then the user could free the timer instance and when
it was actually processed UAF occurred.
This issue could be reproduced via testcase snd_timer01 in ltp - running
several instances of that testcase at the same time.
What I actually met was that the ack_list of the timer broken and the
kernel went into deadloop with irqoff. That could be detected by
hardlockup detector on board or when we run it on qemu, we could use gdb
to dump the ack_list when the console has no response.
To fix this issue, we delete the timer instance from ack_list and
active_list unconditionally in snd_timer_stop1().
Takashi Iwai [Tue, 2 Nov 2021 16:18:59 +0000 (17:18 +0100)]
ALSA: usb-audio: Add quirk for Audient iD14
Audient iD14 (2708:0002) may get a control message error that
interferes the operation e.g. with alsactl. Add the quirk to ignore
such errors like other devices.
Takashi Iwai [Mon, 1 Nov 2021 15:58:27 +0000 (16:58 +0100)]
Merge tag 'asoc-v5.16' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.16
This is an unusually large set of updates, mostly a large crop of
unusually big drivers coupled with extensive overhauls of existing code.
There's a SH change here for the DAI format terminology, the change is
straightforward and the SH maintainers don't seem very active.
- A new version of the audio graph card which supports a wider range of
systems.
- Move of the Cirrus DSP framework into drivers/firmware to allow for
future use by non-audio DSPs.
- Several conversions to YAML DT bindings.
- Continuing cleanups to the SOF and Intel code.
- A very big overhaul of the cs42l42 driver, correcting many problems.
- Support for AMD Vangogh and Yelow Cap, Cirrus CS35L41, Maxim
MAX98520 and MAX98360A, Mediatek MT8195, Nuvoton NAU8821, nVidia
Tegra210, NXP i.MX8ULP, Qualcomm AudioReach, Realtek ALC5682I-VS,
RT5682S, and RT9120 and Rockchip RV1126 and RK3568
Jason Ormes [Sat, 30 Oct 2021 20:04:05 +0000 (15:04 -0500)]
ALSA: usb-audio: Line6 HX-Stomp XL USB_ID for 48k-fixed quirk
Adding the Line6 HX-Stomp XL USB_ID as it needs this fixed frequency
quirk as well.
The device is basically just the HX-Stomp with some more buttons on
the face. I've done some recording with it after adding it, and it
seems to function properly with this fix. The Midi features appear to
be working as well.
[ a coding style fix and patch reformat by tiwai ]
Mark Brown [Fri, 29 Oct 2021 20:53:41 +0000 (21:53 +0100)]
Merge series "Multiple headphone codec driver support" from Brent Lu <brent.lu@intel.com>:
Support multiple headphone drivers in same machine driver. In this
case, both rt5682 and rt5682s are supported and enumerated by different
ACPI HID "10EC5682" and "RTL5682".
V2 Changes:
- remove useless 'NULL', 'false' in if-condition
- can use 'comp_ids' field alone to enumerate driver
- add comma to the end of entry in structure initialization
- keep the table of byt/cht/cml/icl untouched
V3 Changes:
- upstreamd from SOF github, PR#3200
- use new compatiable IDs to shrink the enumerate table of BYT and CHT
- add 'const' to snd_soc_acpi_codecs structures
V4 Changes:
- add signoff to patch 4~6
V5 Changes:
- none, just rebase for patch 3 conflict
Brent Lu (3):
ASoC: soc-acpi: add comp_ids field for machine driver matching
ASoC: Intel: sof_rt5682: detect codec variant in probe function
ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s
Pierre-Louis Bossart (3):
ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs
ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs
ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs
Mark Brown [Fri, 29 Oct 2021 20:53:39 +0000 (21:53 +0100)]
Merge series "ASoC: Sanity checks and soc-topology updates" from Cezary Rojewski <cezary.rojewski@intel.com>:
Couple of soc-topology related changes and a use-after-free fix. Said fix
and two sanity checks for soc-topology lead the way. While the
use-after-free is quite obvious, the sanity checks are here to cover for
cases where user malformed the topology file -or- access to filesystem
somehow got interrupted during copy operation. We shouldn't be reading
outside the file boundary.
Afterward a change to soc_tplg_add_kcontrol(): device being passed to
soc_tplg_add_dcontrol() from comp->dev to tplg->dev which corrects
dev_xxx() invoked later on.
Also, device used for topology memory allocations from component->dev to
component->card->dev so memory gets freed each time card device (usually
platform device) is removed rather than the component device what may
happen less frequently.
Dummy component gets smarter and no longer overrides hw_params if
there are other components accociated with related struct
snd_soc_pcm_runtime instance.
Amadeusz Sławiński (5):
ASoC: core: Remove invalid snd_soc_component_set_jack call
ASoC: topology: Check for dapm widget completeness
ASoC: topology: Use correct device for prints
ASoC: topology: Change topology device to card device
ASoC: Stop dummy from overriding hwparams
Uwe Kleine-König [Tue, 19 Oct 2021 07:41:25 +0000 (09:41 +0200)]
ASoC: tlv320aic3x: Make aic3x_remove() return void
Up to now aic3x_remove() returns zero unconditionally. Make it return
void instead which makes it easier to see in the callers that there is
no error to handle.
Also the return value of i2c and spi remove callbacks is ignored anyway.
Brent Lu [Fri, 29 Oct 2021 17:14:04 +0000 (01:14 +0800)]
ASoC: soc-acpi: add comp_ids field for machine driver matching
A machine driver needs to be enumerated by more than one ACPI HID if
it supports second headphone driver (i.e. rt5682 and rt5682s).
However, the id field in snd_soc_acpi_mach structure could contain
only one HID. By adding a 'comp_ids' field which can contain several
HIDs, we can enumerate a machine driver by multiple ACPI HIDs.
In case that there are other components assigned to runtime device,
depending on order dummy component can override their params with its
own, which shouldn't happen. Check if there are any other components
assigned to rtd and if so, skip setting hwparams.
Occurs when using topology where 'snd-soc-dummy' gets assigned by
default as codec and platform component.
Alternative approach would be to copy whole dummy handling and rename it
to "snd-soc-null" or something similar. And remove hwparams assignment
to make it really do nothing.
ASoC: topology: Change topology device to card device
Topology needs device for prints and resource allocation. So far,
component->dev is used. However, this may lead to high memory use in
model where card is an independent driver which can be reloaded and
topology is loaded from component's probe() method. Every time machine
driver is reloaded topology is being loaded anew, each time allocating
new memory. Said memory will only be freed when component itself is
being freed.
Address the problem by tying topology to component->card->dev instead,
so memory occupied by the topology is freed whenever related machine
device gets removed.
Mark Brown [Fri, 29 Oct 2021 13:51:28 +0000 (14:51 +0100)]
Merge series "ASoC: cs42l42: Fix definition and handling of jack switch invert" from Richard Fitzgerald <rf@opensource.cirrus.com>:
Summary: The driver applied the opposite of the DT setting to the
wrong register bit.
The jack plug detect hardware in cs42l42 is somewhat confusing,
compounded by an unclear description in the datasheet. This is most
likely the reason that the driver implemented a DT property for the
wrong register bit, that had the opposite effect of what was
described in the binding.
Changing the meaning of the property values isn't feasible; the
driver dates from 2016 and the risk of breaking out-of-tree configs
is too high (the property is also available to ACPI systems).
So the fix is to make the binding doc match the actual behaviour and
then fix the driver to apply it to the correct register bit.
As a bonus, patch #3 converts the binding to yaml.
Richard Fitzgerald (3):
ASoC: dt-bindings: cs42l42: Correct description of ts-inv
ASoC: cs42l42: Correct configuring of switch inversion from ts-inv
ASoC: dt-bindings: cs42l42: Convert binding to yaml
Julian Braha [Fri, 29 Oct 2021 00:12:25 +0000 (20:12 -0400)]
ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_RT1015P
When SND_SOC_MT8192_MT6359_RT1015_RT5682,
SND_SOC_MT8192_MT6359_RT1015_RT5682,
SND_SOC_MT8183_DA7219_MAX98357A, or
SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A is selected,
and GPIOLIB is not selected, Kbuild gives the following
warnings, respectively:
The same codec seems to have different personalities. ESSX8316 was
used for Baytrail/CherryTrail, ESSX8336 seems to be used for AppoloLake,
GeminiLake, JasperLake and TigerLake devices.
ASoC: cs42l42: Correct configuring of switch inversion from ts-inv
The setting from the cirrus,ts-inv property should be applied to the
TIP_SENSE_INV bit, as this is the one that actually affects the jack
detect block. The TS_INV bit only swaps the meaning of the PLUG and
UNPLUG interrupts and should always be 1 for the interrupts to have
the normal meaning.
Due to some misunderstanding the driver had been implemented to
configure the TS_INV bit based on the jack switch polarity. This made
the interrupts behave the correct way around, but left the jack detect
block, button detect and analogue circuits always interpreting an open
switch as unplugged.
The signal chain inside the codec is:
SENSE pin -> TIP_SENSE_INV -> TS_INV -> (invert) -> interrupts
|
v
Jack detect,
button detect and
analog control
As the TIP_SENSE_INV already performs the necessary inversion the
TS_INV bit never needs to change. It must always be 1 to yield the
expected interrupt behaviour.
Some extra confusion has arisen because of the additional invert in the
interrupt path, meaning that a value applied to the TS_INV bit produces
the opposite effect of applying it to the TIP_SENSE_INV bit. The ts-inv
property has therefore always had the opposite effect to what might be
expected (0 = inverted, 1 = not inverted). To maintain the meaning of
the ts-inv property it must be inverted when applied to TIP_SENSE_INV.
ASoC: dt-bindings: cs42l42: Correct description of ts-inv
This swaps the descriptions of the 0 and 1 values to match
what the driver actually does with this property.
The background here is somewhat confusing. The codec has two
invert bits for the tip sense. The DT property should have been
for the TIP_SENSE_INV bit, which is the one that controls the
detect block. Due to some misunderstanding of the hardware the
driver actually implemented setting of the TS_INV bit, which is
only for swapping the sense of the interrupt bits. The description
was taken from the datasheet and refers to TIP_SENSE_INV but
unfortunately TS_INV has a different purpose and the net effect
of changing it is the reverse of what was intended (this is not
clearly described in the datasheet). So the ts-inv settings have
always done the exact opposite of what the description said.
Given the age of the driver, it's too late now to swap the meanings
of the values, so the description is changed to match the behaviour.
They have been annotated with the terminology used in the datasheet
to avoid the confusion of which one corresponds to what the datasheet
calls "inverted tip sense".
Takashi Sakamoto [Thu, 28 Oct 2021 13:03:25 +0000 (22:03 +0900)]
ALSA: oxfw: fix functional regression for Mackie Onyx 1640i in v5.14 or later
A user reports functional regression for Mackie Onyx 1640i that the device
generates slow sound with ALSA oxfw driver which supports media clock
recovery. Although the device is based on OXFW971 ASIC, it does not
transfer isochronous packet with own event frequency as expected. The
device seems to adjust event frequency according to events in received
isochronous packets in the beginning of packet streaming. This is
unknown quirk.
This commit fixes the regression to turn the recovery off in driver
side. As a result, nominal frequency is used in duplex packet streaming
between device and driver. For stability of sampling rate in events of
transferred isochronous packet, 4,000 isochronous packets are skipped
in the beginning of packet streaming.
Clang is a little more pedantic than GCC, which permits implicit
fallthroughs to cases that contain just break or return. Clang's version
is more in line with the kernel's own stance in deprecated.rst, which
states that all switch/case blocks must end in either break,
fallthrough, continue, goto, or return. Add the missing break to fix
the warning.
Julian Braha [Wed, 27 Oct 2021 18:48:35 +0000 (14:48 -0400)]
ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_DMIC
When SND_SOC_AMD_RENOIR_MACH or SND_SOC_AMD_RV_RT5682_MACH
are selected, and GPIOLIB is not selected, Kbuild gives
the following warnings, respectively:
This is because SND_SOC_DMIC and SND_SOC_MAX98357A are
selected by SND_SOC_AMD_RV_RT5682_MACH and SND_SOC_AMD_RENOIR_MACH,
respectively. However, neither of the selectors depend on or select GPIOLIB,
despite their selectees depending on GPIOLIB.
These unmet dependency bugs were detected by Kismet,
a static analysis tool for Kconfig. Please advise if this
is not the appropriate solution.
Takashi Sakamoto [Wed, 27 Oct 2021 12:55:29 +0000 (21:55 +0900)]
ALSA: firewire-motu: export meter information to userspace as float value
In command DSP models, one meter information consists of 4 bytes for
IEEE 764 floating point (binary32). In previous patch, it is exported
to userspace as 32 bit storage since the storage is also handled in
ALSA firewire-motu driver as well in kernel space in which floating point
arithmetic is not preferable. On the other hand, ALSA firewire-motu driver
doesn't perform floating point calculation. The driver just gather meter
information from isochronous packets and fill structure fields for
userspace.
In 'header' target of Kbuild, UAPI headers are processed before installed.
In this timing, #ifdef macro with __KERNEL__ is removed. This mechanism
is useful in the case so that the 32 bit storage can be accessible as u32
type in kernel space and float type in user space. We can see the same
usage in ''struct acct_v3' in 'include/uapi/linux/acct.h'.
This commit is for the above idea. Additionally, due to message
protocol, meter information is filled with 0xffffffff in the end of
period but 0xffffffff is invalid as binary32. To avoid confusion in
userspace application, the last two elements are left without any
assignment.
Takashi Sakamoto [Wed, 27 Oct 2021 12:55:28 +0000 (21:55 +0900)]
ALSA: firewire-motu: refine parser for meter information in register DSP models
After further investigation, I realize that the total number of elements
in array is not enough to store all of related messages from device.
This commit refines meter array and message parser.
In terms of channel identifier, register DSP models are classified to
two categories:
1. the target of output is selectable
828mk2, 896hd, and Traveler are in the category. They transfer messages
with channel identifier between 0x00 and 0x13 for input meters,
therefore 20 elements are needed to store.
On the other hand, they transfer messages with channel identifier for one
pair of output meters. The selection is done by asynchronous write
transaction to offset 0x'ffff'f000'0b2c. The table for relationship
between written value and available identifiers is below:
Actually in the above three models, 0x96/0x97 pair is the maximum. Thus
the number of available output meter is 24.
2. all of output is available
8 pre, Ultralite, Audio Express, and 4 pre are in the category. They
transfer messages for output meters without any selection. The table for
available identifier for each direction is below:
============== ========= ==========
model input output
============== ========= ==========
8 pre 0x00-0x0f 0x82-0x8d
Ultralite 0x00-0x09 0x82-0x8f
Audio Express 0x00-0x09 0x80-0x8d
4 pre 0x00-0x09 0x80-0x8d
============== ========= ==========
Some of available identifiers might not be used for actual output meters.
Anyway, 24 plus 24 elements accommodate the input/output meters.
I note that isochronous packet from V3HD/V4HD deliver no message.
Notification by asynchronous transaction to registered address seems to be
used for the purpose as well as for change of mixer parameter.
Takashi Sakamoto [Wed, 27 Oct 2021 12:55:27 +0000 (21:55 +0900)]
ALSA: firewire-motu: fix null pointer dereference when polling hwdep character device
ALSA firewire-motu driver recently got support for event notification via
ALSA HwDep interface for register DSP models. However, when polling ALSA
HwDep cdev, the driver can cause null pointer dereference for the other
models due to accessing to unallocated memory or uninitialized memory.
This commit fixes the bug by check the type of model before accessing to
the memory.
Takashi Iwai [Thu, 28 Oct 2021 07:09:11 +0000 (09:09 +0200)]
ALSA: hda/realtek: Add a quirk for HP OMEN 15 mute LED
HP OMEN 15 laptop requires the quirk to fiddle with COEF 0x0b bit 2
for toggling the mute LED. It's already implemented for other HP
laptops, and we just need to add a proper fixup entry.
WARNING: unmet direct dependencies detected for SND_SOC_AMD_MACH_COMMON
Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && X86 && PCI [=n] && I2C [=m]
Selected by [m]:
- SND_SOC_AMD_LEGACY_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m]
Make this more consistent by adding the same checks everywhere.
Fixes: 9d8a7be88b33 ("ASoC: amd: acp: Add legacy sound card support for Chrome audio") Fixes: 9f84940f5004 ("ASoC: amd: acp: Add SOF audio support on Chrome board") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20211027082359.52248-1-arnd@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
Yang Yingliang [Wed, 27 Oct 2021 06:52:28 +0000 (14:52 +0800)]
ASoC: amd: acp: Fix return value check in acp_machine_select()
In case of error, platform_device_register_data() returns ERR_PTR()
and never returns NULL. The NULL test in the return value check
should be replaced with IS_ERR().
Reported-by: Hulk Robot <hulkci@huawei.com> Fixes: e646b51f5dd5 ("ASoC: amd: acp: Add callback for machine driver on ACP") Signed-off-by: Yang Yingliang <yangyingliang@huawei.com> Link: https://lore.kernel.org/r/20211027065228.833825-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: rockchip: i2s_tdm: improve return value handling
cppcheck reports the following warning:
sound/soc/rockchip/rockchip_i2s_tdm.c:599:9: warning: Identical
condition and return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/rockchip/rockchip_i2s_tdm.c:594:6: note: If condition 'ret'
is true, the function will return/exit
if (ret)
^
sound/soc/rockchip/rockchip_i2s_tdm.c:599:9: note: Returning identical
expression 'ret'
return ret;
^
While the code is not wrong, it's clearer to return 0 directly.
sound/soc/codecs/nau8821.c:390:17: error: Array 'dmic_speed_sel[4]'
accessed at index 4, which is out of bounds. [arrayIndexOutOfBounds]
dmic_speed_sel[i].param, dmic_speed_sel[i].val);
^
sound/soc/codecs/nau8821.c:378:2: note: After for loop, i has value 4
for (i = 0 ; i < 4 ; i++)
^
sound/soc/codecs/nau8821.c:390:17: note: Array index out of bounds
dmic_speed_sel[i].param, dmic_speed_sel[i].val);
^
While the code is not incorrect, we can deal with the out-of-bounds
check in a clearer way that makes static analysis happy.
Johan Hovold [Tue, 26 Oct 2021 09:54:01 +0000 (11:54 +0200)]
ALSA: ua101: fix division by zero at probe
Add the missing endpoint max-packet sanity check to probe() to avoid
division by zero in alloc_stream_buffers() in case a malicious device
has broken descriptors (or when doing descriptor fuzz testing).
Note that USB core will reject URBs submitted for endpoints with zero
wMaxPacketSize but that drivers doing packet-size calculations still
need to handle this (cf. commit 2548288b4fb0 ("USB: Fix: Don't skip
endpoint descriptors with maxpacket=0")).
Mark Brown [Tue, 26 Oct 2021 19:00:42 +0000 (20:00 +0100)]
Merge series "ASoC: cs42l42: Fixes to power-down" from Richard Fitzgerald <rf@opensource.cirrus.com>:
Driver probe and remove were inconsistent in what they did to power-down
and neither did all steps. In addition to that, neither function
prevented the interrupt handler from running during and after power-down.
Richard Fitzgerald (2):
ASoC: cs42l42: Reset and power-down on remove() and failed probe()
ASoC: cs42l42: free_irq() before powering-down on probe() fail
Mark Brown [Tue, 26 Oct 2021 19:00:40 +0000 (20:00 +0100)]
Merge series "Update Lpass digital codec macro drivers" from Srinivasa Rao Mandadapu <srivasam@codeaurora.org>:
This patch set is to add support for lpass sc7280 based targets.
Upadate compatible name and change of bulk clock voting to optional
clock voting in digital codecs va, rx, tx macro drivers.
Changes Since V3:
-- Removed fixes tag.
-- Change signedoff by sequence.
Changes Since V2:
-- Add Tx macro deafults for lpass sc7280
Changes Since V1:
-- Removed individual clock voting and used bulk clock optional.
-- Removed volatile changes and fixed default values.
-- Typo errors.
Srinivasa Rao Mandadapu (5):
ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for
sc7280
ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital
codecs
ASoC: codecs: tx-macro: Enable tx top soundwire mic clock
ASoC: codecs: tx-macro: Update tx default values
ASoC: codecs: Change bulk clock voting to optional voting in digital
codecs
--
Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc.,
is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.
Mark Brown [Tue, 26 Oct 2021 19:00:39 +0000 (20:00 +0100)]
Merge series "ASoC: qcom: Add AudioReach support" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
Hi Mark,
This version is a respin of v10 fixing a build error in 12/17 patch.
QCOM SoC relevant non-audio patches in this series has been merged into
the Qualcomm drivers-for-5.16 tree, as this series depends those patches
an immutable tag is available at:
https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git tags/20210927135559.738-6-srinivas.kandagatla@linaro.org
This patchset adds ASoC driver support to configure signal processing
framework ("AudioReach") which is integral part of Qualcomm next
generation audio SDK and will be deployed on upcoming Qualcomm chipsets.
It makes use of ASoC Topology to load graphs on to the DSP which is then
managed by APM (Audio Processing Manager) service to prepare/start/stop.
Here is simplified high-level block diagram of AudioReach:
AudioReach has constructs of sub-graph, container and modules.
Each sub-graph can have N containers and each Container can have N Modules
and connections between them can be linear or non-linear.
An audio function can be realized with one or many connected
sub-graphs. There are also control/event paths between modules that can
be wired up while building graph to achieve various control mechanism
between modules. These concepts of Sub-Graph, Containers and Modules
are represented in ASoC topology.
Here is simple I2S graph with a Write Shared Memory and a
Volume control module within a single Subgraph (1) with one Container (1)
and 5 modules.
This patchset adds very minimal support for AudioReach which includes
supporting sub-graphs containing CODEC DMA ports and simple PCM
Decoder/Encoder and Logger Modules. Additional capabilities will
be built over time to expose features offered by AudioReach.
This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5
and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via
VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio
via I2S.
First 10 Patches are mostly reorganization existing Old QDSP Audio
Framework code and bindings so that we could reuse them on AudioReach.
ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at
https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/
and Qualcomm AudioReach DSP headers are available at:
https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-headers
Note: There is one false positive warning in this patchset:
audioreach.c:80:45: warning: array of flexible structures
Thanks,
srini
Changes since v10:
- fix build error during arm64 defconfig build reported by Mark in 12/17 patch
for audioreach_tplg_init symbol
Srinivas Kandagatla (17):
ASoC: dt-bindings: move LPASS dai related bindings out of q6afe
ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe
ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h
ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file
ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file
ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible
ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible
ASoC: dt-bindings: add q6apm digital audio stream bindings
ASoC: qdsp6: audioreach: add basic pkt alloc support
ASoC: qdsp6: audioreach: add q6apm support
ASoC: qdsp6: audioreach: add module configuration command helpers
ASoC: qdsp6: audioreach: add Kconfig and Makefile
ASoC: qdsp6: audioreach: add topology support
ASoC: qdsp6: audioreach: add q6apm-dai support
ASoC: qdsp6: audioreach: add q6apm lpass dai support
ASoC: qdsp6: audioreach: add q6prm support
ASoC: qdsp6: audioreach: add support for q6prm-clocks
ASoC: amd: acp: SND_SOC_AMD_ACP_COMMON should depend on X86 && PCI
All configuration symbols for AMD Audio ACP conponents depend on X86 &&
PCI, except for SND_SOC_AMD_ACP_COMMON. Add a dependency on X86 && PCI
to SND_SOC_AMD_ACP_COMMON, to prevent asking the user about AMD Audio
ACP support when configuring a kernel without X86 or PCI support.
ASoC: amd: acp: Wrap AMD Audio ACP components in SND_SOC_AMD_ACP_COMMON
The build only descends into sound/soc/amd/acp/ if
CONFIG_SND_SOC_AMD_ACP_COMMON=y. Hence all later config symbols should
depend on SND_SOC_AMD_ACP_COMMON, to prevent asking the user about
config symbols for driver code that won't be build anyway.
ASoC: cs42l42: free_irq() before powering-down on probe() fail
Relying on devm to free the irq handler on probe failure leaves a
small window of opportunity for an interrupt to become pending and
then the handler to run after the chip has been reset and powered
off.
For safety cs42l42_probe() should free the irq in the error path.
As the irq is now disabled by the driver in probe() and remove()
there is no point allocating it as a devres-managed item, so
convert to plain non-devres.
ASoC: dt-bindings: add q6apm digital audio stream bindings
On AudioReach audio Framework, Audio Streams (PCM/Compressed) are managed by
Q6APM(Audio Process Manager) service. This patch adds bindings for this DAIs
exposed by the DSP.
On AudioReach audio Framework access to LPASS ports is via Q6PRM
(Proxy Resource Manager) service, so add a dedicated lpass-clock compatible
string for this.
ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file
Various Q6DSP frameworks will use LPASS Audio IP, so move all the hardware
specific details to a common file so that they could be reused across
multiple Q6DSP frameworks.
In this case all the audio ports definitions can be moved to a common file
to be able to reuse across multiple Q6DSP frameworks.
ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h
move all LPASS audio ports defines from q6afe.h to q6dsp-lpass-ports.h
as these belong to LPASS IP.
Also this move helps in reusing this header across multiple audio
frameworks on Qualcomm Audio DSP.
This patch is split out of the dt-bindings patch to enable easy review.
ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe
q6afe (Audio Front End) is one of the DSP service that handles both
LPASS (Low Power Audio SubSystem) Audio ports and LPASS clocks.
As LPASS is a hardwware IP and commonly used by Qualcomm Audio DSP.
In order to allow multiple DSP frameworks to use these bindings
its best to move it out from the dsp specific bindings.
For compatibility reasons and not breaking which is already working
we still maintain same compatible string "qcom,q6afe-clocks"
Also as part of this change convert these LPASS clocks related bindings
into yaml format.
ASoC: dt-bindings: move LPASS dai related bindings out of q6afe
q6afe (Audio Front End) is one of the DSP service that handles both
LPASS (Low Power Audio SubSystem) Audio ports and LPASS clocks.
As LPASS is a hardwware IP and commonly used by Qualcomm Audio DSP.
In order to allow multiple DSP frameworks to use these bindings
its best to move it out from the dsp specific bindings.
For compatibility reasons and not breaking which is already working
we still maintain same compatible string "qcom,q6afe-dais"
Also as part of this change convert these LPASS dai related bindings
into yaml format.
Mark Brown [Tue, 26 Oct 2021 12:49:20 +0000 (13:49 +0100)]
Merge tag '20210927135559.738-6-srinivas.kandagatla@linaro.org' of https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux into v11_20211026_srinivas_kandagatla_asoc_qcom_add_audioreach_support for audioreach support
This immutable branch is based on v5.15-rc1 and contains the following
patches extending the existig APR driver to also implement GPR: 20210927135559.738-2-srinivas.kandagatla@linaro.org 20210927135559.738-3-srinivas.kandagatla@linaro.org 20210927135559.738-4-srinivas.kandagatla@linaro.org 20210927135559.738-5-srinivas.kandagatla@linaro.org 20210927135559.738-6-srinivas.kandagatla@linaro.org
David Lin [Mon, 25 Oct 2021 11:38:57 +0000 (19:38 +0800)]
ASoC: nau8825: add set_jack coponment support
Use set_jack ops to set jack for new machine drivers. Meanwhile,
the old machine drivers can still call previous export function
"nau8825_enable_jack_detect".
Chengfeng Ye [Sun, 24 Oct 2021 11:17:36 +0000 (04:17 -0700)]
ALSA: usb-audio: fix null pointer dereference on pointer cs_desc
The pointer cs_desc return from snd_usb_find_clock_source could
be null, so there is a potential null pointer dereference issue.
Fix this by adding a null check before dereference.
Chengfeng Ye [Sun, 24 Oct 2021 10:46:11 +0000 (03:46 -0700)]
ALSA: gus: fix null pointer dereference on pointer block
The pointer block return from snd_gf1_dma_next_block could be
null, so there is a potential null pointer dereference issue.
Fix this by adding a null check before dereference.
Pavel Skripkin [Sun, 24 Oct 2021 14:03:15 +0000 (17:03 +0300)]
ALSA: mixer: fix deadlock in snd_mixer_oss_set_volume
In commit 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots")
added mutex protection in snd_mixer_oss_set_volume(). Second
mutex_lock() in same function looks like typo, fix it.
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