Dell has new platform for ALC274.
This will support to enable headset mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In `create_composite_quirk`, the terminating condition of for loops is
`quirk->ifnum < 0`. So any composite quirks should end with `struct
snd_usb_audio_quirk` object with ifnum < 0.
for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) {
.....
}
the data field of Bower's & Wilkins PX headphones usb device device quirks
do not end with {.ifnum = -1}, wihch may result in out-of-bound read.
This Patch fix the bug by adding an ending quirk object.
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places where we access the data without checking the
actual object size from the USB audio descriptor. This may result in
OOB access, as recently reported.
This patch addresses these missing checks. Most of added codes are
simple bLength checks in the caller side. For the input and output
terminal parsers, we put the length check in the parser functions.
For the input terminal, a new argument is added to distinguish between
UAC1 and the rest, as they treat different objects.
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Reported-by: Hui Peng <benquike@163.com>
Tested-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had some sanity checks of the mixer unit descriptors but they
are too loose and some corner cases are overlooked. Add more strict
checks in uac_mixer_unit_get_channels() for avoiding possible OOB
accesses by malformed descriptors.
This also changes the semantics of uac_mixer_unit_get_channels()
slightly. Now it returns zero for the cases where the descriptor
lacks of bmControls instead of -EINVAL. Then the caller side skips
the mixer creation for such unit while it keeps parsing it.
This corresponds to the case like Maya44.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser for the processing unit reads bNrInPins field before the
bLength sanity check, which may lead to an out-of-bound access when a
malformed descriptor is given. Fix it by assignment after the bLength
check.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nobody has actually used the type (VERIFY_READ vs VERIFY_WRITE) argument
of the user address range verification function since we got rid of the
old racy i386-only code to walk page tables by hand.
It existed because the original 80386 would not honor the write protect
bit when in kernel mode, so you had to do COW by hand before doing any
user access. But we haven't supported that in a long time, and these
days the 'type' argument is a purely historical artifact.
A discussion about extending 'user_access_begin()' to do the range
checking resulted this patch, because there is no way we're going to
move the old VERIFY_xyz interface to that model. And it's best done at
the end of the merge window when I've done most of my merges, so let's
just get this done once and for all.
This patch was mostly done with a sed-script, with manual fix-ups for
the cases that weren't of the trivial 'access_ok(VERIFY_xyz' form.
There were a couple of notable cases:
- csky still had the old "verify_area()" name as an alias.
- the iter_iov code had magical hardcoded knowledge of the actual
values of VERIFY_{READ,WRITE} (not that they mattered, since nothing
really used it)
- microblaze used the type argument for a debug printout
but other than those oddities this should be a total no-op patch.
I tried to fix up all architectures, did fairly extensive grepping for
access_ok() uses, and the changes are trivial, but I may have missed
something. Any missed conversion should be trivially fixable, though.
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Among a few HD-audio fixes, the only significant one is the
regression fix on some machines like Dell XPS due to the default
binding changes. We ended up reverting the whole since the fix for
ASoC HD-audio driver won't be available immediately.
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Merge tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Among a few HD-audio fixes, the only significant one is the regression
fix on some machines like Dell XPS due to the default binding changes.
We ended up reverting the whole since the fix for ASoC HD-audio driver
won't be available immediately"
* tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Revert DSP detection on legacy HD-audio driver
ALSA: hda/tegra: clear pending irq handlers
ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptops
This essentially reverts the commits
c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present
and Skylake driver selected")
and
d82b51c855 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+
driver selection")
for the path of legacy HD-audio controller (snd-hda-intel).
The automatic DSP detection and skip of binding with the legacy driver
caused regressions on several machines like Dell XPS13. They give the
PCI class 0x40380 indicating the availability of DSP while they don't
work with ASoC SKL driver (yet).
As the support of ASoC driver for such devices isn't available, it's
better to revert the whole DSP-detection-and-skip behavior of the
legacy driver, so that we can get the old good driver working on such
devices.
The pci_binding option for ASoC SKL driver is still kept so that it
can work without blacklisting.
Fixes: c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Reported-by: Hans de Goede <hdegoede@redhat.com>
Reported-by: Azat Khuzhin <dohardgopro@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after disabling interrupts on the module, it could be possible
that irq handlers are still running. System hang is seen during
suspend path. It was found that, there were pending writes on the
HDA bus and clock was disabled by that time.
Above mentioned issue is fixed by clearing any pending irq handlers
before disabling clocks and returning from hda suspend.
Suggested-by: Mohan Kumar <mkumard@nvidia.com>
Suggested-by: Dara Ramesh <dramesh@nvidia.com>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose
CODEC is Realtek ALC294 has jack auto detection feature. This patch
enables the feature.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'for-linus-4.21-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/xen/tip
Pull xen updates from Juergen Gross:
"Xen features and fixes:
- a series to enable KVM guests to be booted by qemu via the Xen PVH
boot entry for speeding up KVM guest tests
- a series for a common driver to be used by Xen PV frontends (right
now drm and sound)
- two other fixes in Xen related code"
* tag 'for-linus-4.21-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/xen/tip:
ALSA: xen-front: Use Xen common shared buffer implementation
drm/xen-front: Use Xen common shared buffer implementation
xen: Introduce shared buffer helpers for page directory...
xen/pciback: Check dev_data before using it
kprobes/x86/xen: blacklist non-attachable xen interrupt functions
KVM: x86: Allow Qemu/KVM to use PVH entry point
xen/pvh: Add memory map pointer to hvm_start_info struct
xen/pvh: Move Xen code for getting mem map via hcall out of common file
xen/pvh: Move Xen specific PVH VM initialization out of common file
xen/pvh: Create a new file for Xen specific PVH code
xen/pvh: Move PVH entry code out of Xen specific tree
xen/pvh: Split CONFIG_XEN_PVH into CONFIG_PVH and CONFIG_XEN_PVH
Pull sparc updates from David Miller:
- Automatic system call table generation, from Firoz Khan.
- Clean up accesses to the OF device names by using full_name instead
of path_component_name.
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-next:
ALSA: sparc: Use of_node_name_eq for node name comparisons
sbus: Use of_node_name_eq for node name comparisons
sparc: generate uapi header and system call table files
sparc: add system call table generation support
sparc: add __NR_syscalls along with NR_syscalls
sparc: move __IGNORE* entries to non uapi header
sparc: Use DT node full_name instead of name for resources
sparc: Remove unused leon_trans_init
sparc: Use device_type helpers to access the node type
sparc: Use of_node_name_eq for node name comparisons
sparc: Convert to using %pOFn instead of device_node.name
sparc: prom: use property "name" directly to construct node names
of: Drop full path from full_name for PDT systems
sparc: Convert to using %pOF instead of full_name
fs/openpromfs: Use of_node_name_eq for node name comparisons
fs/openpromfs: use full_name instead of path_component_name
There are no intensive changes in both ALSA and ASoC core parts while
rather most of changes are a bunch of driver fixes and updates.
A large diff pattern appears in ASoC TI part which now merges both
OMAP and DaVinci stuff, but the rest spreads allover the places.
Note that this pull request includes also some updates for LED trigger
and platform drivers for mute LEDs, appearing in the diffstat as well.
Some highlights:
ASoC:
- Preparatory work for merging the audio-graph and audio-graph-scu
cards
- A merge of TI OMAP and DaVinci directories, as both product lines
get merged together. Also including a few architecture changes as
well.
- Major cleanups of the Maxim MAX9867 driver
- Small fixes for tablets & co with Intel BYT/CHT chips
- Lots of rsnd updates as usual
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers
HD-audio:
- Introduce audio-mute LED trigger for replacing the former hackish
dynamic binding
- Huawei WMI hotkey and mute LED support
- Refactoring of PM code and display power controls
- Headset button support in the generic jack code
- A few updates for Tegra
- Fixups for HP EliteBook and ASUS UX391UA
- Lots of updates for Intel ASoC HD-audio, including the improved DSP
detection and the fallback binding from ASoC SST to legacy HD-audio
controller drivers
Others:
- Updates for FireWire TASCAM and Fireface devices, some other fixes
- A few potential Spectre v1 fixes that are all trivial
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Merge tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There are no intensive changes in both ALSA and ASoC core parts while
rather most of changes are a bunch of driver fixes and updates. A
large diff pattern appears in ASoC TI part which now merges both OMAP
and DaVinci stuff, but the rest spreads allover the places.
Note that this pull request includes also some updates for LED trigger
and platform drivers for mute LEDs, appearing in the diffstat as well.
Some highlights:
ASoC:
- Preparatory work for merging the audio-graph and audio-graph-scu
cards
- A merge of TI OMAP and DaVinci directories, as both product lines
get merged together. Also including a few architecture changes as
well.
- Major cleanups of the Maxim MAX9867 driver
- Small fixes for tablets & co with Intel BYT/CHT chips
- Lots of rsnd updates as usual
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx
I2S controllers
HD-audio:
- Introduce audio-mute LED trigger for replacing the former hackish
dynamic binding
- Huawei WMI hotkey and mute LED support
- Refactoring of PM code and display power controls
- Headset button support in the generic jack code
- A few updates for Tegra
- Fixups for HP EliteBook and ASUS UX391UA
- Lots of updates for Intel ASoC HD-audio, including the improved DSP
detection and the fallback binding from ASoC SST to legacy HD-audio
controller drivers
Others:
- Updates for FireWire TASCAM and Fireface devices, some other fixes
- A few potential Spectre v1 fixes that are all trivial"
* tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection
ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected
ALSA: HDA: export process_unsol_events()
ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294
ALSA: bebob: fix model-id of unit for Apogee Ensemble
ALSA: emu10k1: Fix potential Spectre v1 vulnerabilities
ALSA: rme9652: Fix potential Spectre v1 vulnerability
ASoC: ti: Kconfig: Remove the deprecated options
ARM: davinci_all_defconfig: Update the audio options
ARM: omap1_defconfig: Do not select ASoC by default
ARM: omap2plus_defconfig: Update the audio options
ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options
ARM: OMAP2: Update for new MCBSP Kconfig option
ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option
MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support
ASoC: ti: Merge davinci and omap directories
ALSA: hda: add mute LED support for HP EliteBook 840 G4
ALSA: fireface: code refactoring to handle model-specific registers
ALSA: fireface: add support for packet streaming on Fireface 800
ALSA: fireface: allocate isochronous resources in mode-specific implementation
...
For HDaudio and Skylake drivers, add module parameter "pci_binding"
When pci_binding == 0 (AUTO), the PCI class/subclass info is used to
select drivers based on the presence of the DSP.
pci_binding == 1 (LEGACY) forces the use of the HDAudio legacy driver,
even if the DSP is present.
pci_binding == 2 (ASOC) forces the use of the ASOC driver. The
information on the DSP presence is bypassed.
The value for the module parameter needs to be identical for both
drivers. This parameter is intended as a back-up solution if the
automatic detection fails or when the DSP usage fails. Such cases
should be reported on the alsa-devel mailing list for analysis.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the SST/Skylake driver supports per platform selectors, we
can add logic to automatically select the right driver.
If the Skylake driver is selected for a specific platform, and the DSP
is detected at run-time based on the PCI class/subclass/prog-if
information, the legacy HDaudio driver aborts the probe. This will
result in a single driver probing and remove the need for modprobe
blacklists.
Follow-up patches will add a module parameter to bypass the logic if
this automatic detection fails, or if the Skylake driver is unable to
actually support the platform (firmware authentication, missing
topology file, hardware issue, etc).
The same mechanism will be used to conflicts generated by the same PCI
ID being registered by both legacy HDAuudio and SOF drivers for Intel
platforms. In other words SOF will not require changes to the HDaudio
legacy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SOF implementation does not rely on the hdac_bus library, however
for HDMI and HDaudio codec support it does need to deal with
unsolicited events. Instead of re-inventing the wheel, export this
symbol to reuse this part of the library directly.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By default, there is no sound on Asus UX391UA on Linux.
This patch adds sound support on Asus UX391UA. Tested working by three
different users.
The problem has also been described at
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1784485
Signed-off-by: Wandrille RONCE <w@ndrille.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ipcm->substream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/emu10k1/emufx.c:1031 snd_emu10k1_ipcm_poke() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
sound/pci/emu10k1/emufx.c:1075 snd_emu10k1_ipcm_peek() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
Fix this by sanitizing ipcm->substream before using it to index emu->fx8010.pcm
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->channel is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/rme9652/hdsp.c:4100 snd_hdsp_channel_info() warn: potential spectre issue 'hdsp->channel_map' [r] (local cap)
Fix this by sanitizing info->channel before using it to index hdsp->channel_map
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
Also, notice that I refactored the code a bit in order to get rid of the
following checkpatch warning:
ERROR: do not use assignment in if condition
FILE: sound/pci/rme9652/hdsp.c:4103:
if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use page directory based shared buffer implementation
now available as common code for Xen frontend drivers.
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Boris Ostrovsky <boris.ostrovsky@oracle.com>
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
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Merge tag 'asoc-v4.21' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.21
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
We no longer have these options used anywhere.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Create new directory to contain all Texas Instruments specific DAI,
platform and machine drivers instead of scattering them under davinci and
omap directories.
There is already inter dependency between the two directories becasue of
McASP (on dra7x it is serviced by sDMA, not EDMA).
With the upcoming AM654 we will need to introduce new platform driver for
UDMA and it does not fit under davinci, nor under omap.
With the move I have restructured the Kconfig to be more usable in the era
of simple-sound-card:
CPU DAIs can be selected individually and they will select the platform
driver they can be served with.
To avoid breakage, I have moved over deprecated Kconfig options so
defconfig builds will work without regression.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
For sound/soc/{omap => ti}:
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tested with 4.19.9.
v2: Changed from CXT_FIXUP_MUTE_LED_GPIO to CXT_FIXUP_HP_DOCK because
that's what the existing fixups for EliteBooks use.
Signed-off-by: Mantas Mikulėnas <grawity@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a result of investigation for Fireface 800, 'struct snd_ff_spec.regs'
is just for higher address to receive tx asynchronous packets of MIDI
messages, thus it can be simplified.
This commit simplifies it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to multiplex PCM frames into isochronous
packets and demultiplex PCM frames from isochronous packets for ALSA PCM
applications.
Fireface 800 voluntarily maintains resources for tx isochronous
communication. It performs reservation of isochronous channel and
allocation/update of bandwidth in some cases below:
- at a first request to allocation after bus resets
- at requests to allocation when further bandwidth is required
When request is grant and the unit is prepared, read data from
0x0000801c0008 represents isochronous channel for tx stream, then
the unit can handle requests to start communication. If driver
send the request without checking the register, the unit takes
panic to continue bus resets. The unit starts transmission of
tx packets after receiving several rx packets from driver.
I note that the unit can process tx/rx packets and generate/record
sound regardless of HOST LED.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The way to maintain isochronous resources on bus is different between
Fireface 400/800.
This commit is a preparation. This commit moves a function to allocate resource to
model-dependent implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 400/800 use three modes against the number of data channels in
data block for both tx/rx packets.
This commit adds refactoring for it. Some enumerators are added to
represent each of mode and a function is added to calculate the mode
from sampling frequency code (sfc).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both of Fireface 400/800 have the same register to switch frame fetching
mode regardless of difference of available number of PCM frames in
rx isochronous packet.
This commit moves a helper function from model-dependent implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to my memo at hand and saved records, writing 0x00000001 to
SND_FF_REG_FETCH_PCM_FRAMES disables fetching PCM frames in corresponding
channel, however current implement uses reversed logic. This results in
muted volume in device side during playback.
This commit corrects the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e1 ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An initial commit to add tracepoints for packets without CIP headers
uses different print formats for added tracepoints. However this is not
convenient for users/developers to prepare debug tools.
This commit uses the same format for the two tracepoints.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: b164d2fd6e ('ALSA: firewire_lib: add tracepoints for packets without CIP headers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An initial commit to add tracepoints for packets without CIP headers
introduces a wrong assignment to 'data_blocks' value of
'out_packet_without_header' tracepoint.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: b164d2fd6e ('ALSA: firewire_lib: add tracepoints for packets without CIP headers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-1/6 engine of ALSA firewire stack, a packet handler has a
second argument for 'the number of bytes in payload of isochronous
packet'. However, an incoming packet handler without CIP header uses the
value as 'the number of quadlets in the payload'. This brings userspace
applications to receive the number of PCM frames as four times against
real time.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 3b196c394d ('ALSA: firewire-lib: add no-header packet processing')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add support to Display_port_rx mixers required to
select path between ASM stream and AFE ports.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support of AFE DAI for Display_port_rx port.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for Display_Port_Rx
port in AFE.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds MP3 playback support in q6asm dais, adding other codec
support should be pretty trivial.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to mp3 format in ASM module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Default copy function uses kmalloc to allocate buffers, lets check
if the runtime buffers are setup before making this allocations.
This can be useful if the buffers are dma buffers.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current SKYLAKE kconfig is a all-you-can-eat selection that will
support all known plaforms. This is however not necessarily a good
thing: most platforms for SKL and KBL don't support the DSP, but a
number of CNL/WHL ones do. Selecting this driver in all cases isn't
really smart and will require users to muck with blacklists.
Partition the configs to allow distributions to select on which
platform this driver is used. Keep the existing SND_SOC_INTEL_SKYLAKE
config to select everything for backwards compatibility. This patch does
not provide new functionality, only finer-grained choices in supported
platforms.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "prefix" by many ways.
But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch supports it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,prefix = "xxx"; // initial
simple-audio-card,dai-link {
prefix = "xxx"; // overwrite
cpu {
...
};
codec {
prefix = "xxx"; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "convert_rate/channel"
by many ways. But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,convert_channels = <xxx>; // initial
simple-audio-card,dai-link {
convert_channels = <xxx>; // overwrite
cpu {
convert_channels = <xxx>; // overwrite
};
codec {
convert_channels = <xxx>; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "mclk-fs" by many way.
But, it is not useful and readable.
We want to do is that allow having mclk-fs everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,mclk-fs = <xxx>; // for initial
simple-audio-card,dai-link {
mclk-fs = <xxx>; // overwrite
cpu {
mclk-fs = <xxx>; // overwrite
};
codec {
mclk-fs = <xxx>; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card and simple-scu-card are very similar driver,
but the former is supporting normal sound card,
the latter is supporting DPCM sound card.
We couldn't use normal sound and DPCM sound in same time by
one sound card. This patch merges both sound card into
simple-card. Now we can use both feature on same driver.
simple-card is now supporting .compatible = "simple-scu-audio-card".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "prefix" by many ways.
But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch supports it.
It will be overwrote if lower node has it.
sound {
prefix = "xxx"; // initial
};
codec {
audio-graph-card,prefix = "xxx"; // overwrite
ports {
prefix = "xxx"; // overwrite
port {
prefix = "xxx"; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
