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* Merge tag 'remove-dma_zalloc_coherent-5.0' of ↵Linus Torvalds2019-01-122-4/+4
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.infradead.org/users/hch/dma-mapping Pull dma_zalloc_coherent() removal from Christoph Hellwig: "We've always had a weird situation around dma_zalloc_coherent. To safely support mapping the allocations to userspace major architectures like x86 and arm have always zeroed allocations from dma_alloc_coherent, but a couple other architectures were missing that zeroing either always or in corner cases. Then later we grew anothe dma_zalloc_coherent interface to explicitly request zeroing, but that just added __GFP_ZERO to the allocation flags, which for some allocators that didn't end up using the page allocator ended up being a no-op and still not zeroing the allocations. So for this merge window I fixed up all remaining architectures to zero the memory in dma_alloc_coherent, and made dma_zalloc_coherent a no-op wrapper around dma_alloc_coherent, which fixes all of the above issues. dma_zalloc_coherent is now pointless and can go away, and Luis helped me writing a cocchinelle script and patch series to kill it, which I think we should apply now just after -rc1 to finally settle these issue" * tag 'remove-dma_zalloc_coherent-5.0' of git://git.infradead.org/users/hch/dma-mapping: dma-mapping: remove dma_zalloc_coherent() cross-tree: phase out dma_zalloc_coherent() on headers cross-tree: phase out dma_zalloc_coherent()
| * cross-tree: phase out dma_zalloc_coherent()Luis Chamberlain2019-01-082-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We already need to zero out memory for dma_alloc_coherent(), as such using dma_zalloc_coherent() is superflous. Phase it out. This change was generated with the following Coccinelle SmPL patch: @ replace_dma_zalloc_coherent @ expression dev, size, data, handle, flags; @@ -dma_zalloc_coherent(dev, size, handle, flags) +dma_alloc_coherent(dev, size, handle, flags) Suggested-by: Christoph Hellwig <hch@lst.de> Signed-off-by: Luis Chamberlain <mcgrof@kernel.org> [hch: re-ran the script on the latest tree] Signed-off-by: Christoph Hellwig <hch@lst.de>
* | ALSA: hda/realtek - Disable headset Mic VREF for headset mode of ALC225Kailang Yang2019-01-091-1/+15
| | | | | | | | | | | | | | | | | | | | | | Disable Headset Mic VREF for headset mode of ALC225. This will be controlled by coef bits of headset mode functions. [ Fixed a compile warning and code simplification -- tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - Add unplug function into unplug state of Headset Mode ↵Kailang Yang2019-01-091-0/+1
| | | | | | | | | | | | | | | | | | | | | | for ALC225 Forgot to add unplug function to unplug state of headset mode for ALC225. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: fix CM6206 register definitionsAmadeusz Sławiński2019-01-081-1/+1
| | | | | | | | | | | | | | | | fix typo after a recent commit causing headphones to have no sound Fixes: ad43d528a7ac (ALSA: usb-audio: Define registers for CM6206) Signed-off-by: Amadeusz Sławiński <amade@asmblr.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: cs46xx: Potential NULL dereference in probeDan Carpenter2019-01-081-0/+3
| | | | | | | | | | | | | | | | | | | | The "chip->dsp_spos_instance" can be NULL on some of the ealier error paths in snd_cs46xx_create(). Reported-by: "Yavuz, Tuba" <tuba@ece.ufl.edu> Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - Support Dell headset mode for New AIO platformKailang Yang2019-01-071-0/+1
| | | | | | | | | | | | | | | | | | 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>
* | ALSA: usb-audio: Fix an out-of-bound read in create_composite_quirksHui Peng2019-01-071-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: 240a8af929c7 ("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>
* | ALSA: usb-audio: Always check descriptor sizes in parser codeTakashi Iwai2019-01-073-12/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | ALSA: usb-audio: Check mixer unit descriptors more strictlyTakashi Iwai2019-01-071-4/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | ALSA: usb-audio: Avoid access before bLength check in build_audio_procunit()Takashi Iwai2019-01-071-2/+8
|/ | | | | | | | | | 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>
* Remove 'type' argument from access_ok() functionLinus Torvalds2019-01-032-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* Merge tag 'sound-fix-4.21-rc1' of ↵Linus Torvalds2019-01-036-117/+11
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
| * ALSA: hda - Revert DSP detection on legacy HD-audio driverTakashi Iwai2019-01-014-116/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This essentially reverts the commits c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") and d82b51c855a2 ("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: c337104b1a16 ("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>
| * ALSA: hda/tegra: clear pending irq handlersSameer Pujar2019-01-011-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptopsJian-Hong Pan2019-01-011-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | 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: 4e051106730d ("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>
* | Merge tag 'for-linus-4.21-rc1-tag' of ↵Linus Torvalds2018-12-267-261/+84
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
| * | ALSA: xen-front: Use Xen common shared buffer implementationOleksandr Andrushchenko2018-12-187-261/+84
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-nextLinus Torvalds2018-12-261-3/+3
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
| * | | ALSA: sparc: Use of_node_name_eq for node name comparisonsRob Herring2018-12-051-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. Cc: "David S. Miller" <davem@davemloft.net> Cc: sparclinux@vger.kernel.org Signed-off-by: Rob Herring <robh@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
* | | | Merge tag 'sound-4.21-rc1' of ↵Linus Torvalds2018-12-25204-4833/+10461
|\ \ \ \ | | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 ...
| * | | ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selectionPierre-Louis Bossart2018-12-192-18/+56
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selectedPierre-Louis Bossart2018-12-194-8/+96
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: HDA: export process_unsol_events()Keyon Jie2018-12-191-4/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294Wandrille RONCE2018-12-191-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: bebob: fix model-id of unit for Apogee EnsembleTakashi Sakamoto2018-12-191-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit fixes hard-coded model-id for an unit of Apogee Ensemble with a correct value. This unit uses DM1500 ASIC produced ArchWave AG (formerly known as BridgeCo AG). I note that this model supports three modes in the number of data channels in tx/rx streams; 8 ch pairs, 10 ch pairs, 18 ch pairs. The mode is switched by Vendor-dependent AV/C command, like: $ cd linux-firewire-utils $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0600000000 (8ch pairs) $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0601000000 (10ch pairs) $ ./firewire-request /dev/fw1 fcp 0x00ff000003dbeb0602000000 (18ch pairs) When switching between different mode, the unit disappears from IEEE 1394 bus, then appears on the bus with different combination of stream formats. In a mode of 18 ch pairs, available sampling rate is up to 96.0 kHz, else up to 192.0 kHz. $ ./hinawa-config-rom-printer /dev/fw1 { 'bus-info': { 'adj': False, 'bmc': True, 'chip_ID': 21474898341, 'cmc': True, 'cyc_clk_acc': 100, 'generation': 2, 'imc': True, 'isc': True, 'link_spd': 2, 'max_ROM': 1, 'max_rec': 512, 'name': '1394', 'node_vendor_ID': 987, 'pmc': False}, 'root-directory': [ ['HARDWARE_VERSION', 19], [ 'NODE_CAPABILITIES', { 'addressing': {'64': True, 'fix': True, 'prv': False}, 'misc': {'int': False, 'ms': False, 'spt': True}, 'state': { 'atn': False, 'ded': False, 'drq': True, 'elo': False, 'init': False, 'lst': True, 'off': False}, 'testing': {'bas': False, 'ext': False}}], ['VENDOR', 987], ['DESCRIPTOR', 'Apogee Electronics'], ['MODEL', 126702], ['DESCRIPTOR', 'Ensemble'], ['VERSION', 5297], [ 'UNIT', [ ['SPECIFIER_ID', 41005], ['VERSION', 65537], ['MODEL', 126702], ['DESCRIPTOR', 'Ensemble']]], [ 'DEPENDENT_INFO', [ ['SPECIFIER_ID', 2037], ['VERSION', 1], [(58, 'IMMEDIATE'), 16777159], [(59, 'IMMEDIATE'), 1048576], [(60, 'IMMEDIATE'), 16777159], [(61, 'IMMEDIATE'), 6291456]]]]} Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: emu10k1: Fix potential Spectre v1 vulnerabilitiesGustavo A. R. Silva2018-12-191-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: rme9652: Fix potential Spectre v1 vulnerabilityGustavo A. R. Silva2018-12-191-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | Merge tag 'asoc-v4.21' of ↵Takashi Iwai2018-12-18155-3827/+8807
| |\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
| | * \ \ Merge branch 'asoc-4.21' into asoc-nextMark Brown2018-12-18153-3812/+8769
| | |\ \ \
| | | * | | ASoC: ti: Kconfig: Remove the deprecated optionsPeter Ujfalusi2018-12-181-159/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: ti: Merge davinci and omap directoriesPeter Ujfalusi2018-12-1834-289/+427
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: qdsp6: q6routing: add display_port_rx port routingRohit kumar2018-12-141-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: qdsp6: q6afe-dai: add support to Display port RX daisRohit kumar2018-12-141-0/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: qdsp6: qdafe: add support for display_port_rxRohit kumar2018-12-141-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: qdsp6: q6asm-dai: Add support to compress offloadSrinivas Kandagatla2018-12-142-1/+372
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: qdsp6: q6asm: add support to MP3 formatSrinivas Kandagatla2018-12-141-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ALSA: compress: make use of runtime buffer for copySrinivas Kandagatla2018-12-141-3/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: Intel: Skylake: Add more platform granularityPierre-Louis Bossart2018-12-143-9/+92
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: simple-card: tidyup prefix for snd_soc_codec_confKuninori Morimoto2018-12-141-11/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: simple-card: tidyup convert_rate/channel methodKuninori Morimoto2018-12-141-6/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: simple-card: tidyup mclk-fs methodKuninori Morimoto2018-12-141-13/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: simple-card: merge simple-scu-cardKuninori Morimoto2018-12-142-46/+284
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: audio-graph-card: tidyup prefix for snd_soc_codec_confKuninori Morimoto2018-12-141-11/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | | * | | ASoC: audio-graph-card: tidyup convert_rate/channel methodKuninori Morimoto2018-12-141-8/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current audio-graph-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 { convert-channels = <xxx>; // initial }; codec { audio-graph-card,convert-channels = <xxx>; // overwrite ports { convert_channels = <xxx>; // overwrite port { convert_channels = <xxx>; // overwrite endpoint { convert_channels = <xxx>; // overwrite }; }; }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | | ASoC: audio-graph-card: tidyup mclk-fs methodKuninori Morimoto2018-12-141-17/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current audio-graph-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 { mclk-fs = <xxx>; // initial }; codec { ports { mclk-fs = <xxx>; // overwrite port { mclk-fs = <xxx>; // overwrite endpoint { mclk-fs = <xxx>; // overwrite }; }; }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | | ASoC: audio-graph-card: merge audio-graph-scu-cardKuninori Morimoto2018-12-142-47/+320
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | audio-graph-card and audio-graph-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 sound card by audio-graph-card. This patch merges both sound card into it. Now we can use both feature on same driver. audio-grap-card is now supporting .compatible = "audio-graph-scu-card". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | | ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() ID methodKuninori Morimoto2018-12-141-4/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") fixuped getting DAI ID method. It will get DAI ID from OF graph "port", but, we want to consider about "endpoint", too. And, we also want to keep compatibility. This patch fixup it as if (driver has specified DAI ID) use it as DAI ID else if (OF graph endpoint has reg) use it as DAI ID else if (OF graph port has reg) use it as DAI ID else use endpoint count as DAI ID Fixes: commit b6f3fc005a2c8 ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") Reported-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | | ASoC: core: Invoke pcm_new() for all DAI-linkRohit kumar2018-12-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove no_pcm check to invoke pcm_new() for backend dai-links too. This fixes crash in hdmi codec driver during hdmi_codec_startup() while accessing chmap_info struct. chmap_info struct memory is allocated in pcm_new() of hdmi codec driver which is not invoked in case of DPCM when hdmi codec driver is part of backend dai-link. Below is the crash stack: [ 61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018 .. [ 61.666696] CM = 0, WnR = 1 [ 61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000 [ 61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000 [ 61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP [ 61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21 .. [ 61.740269] PC is at hdmi_codec_startup+0x124/0x164 [ 61.745308] LR is at hdmi_codec_startup+0xe4/0x164 Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | | ASoC: pcm512x: Implement the digital_mute interfaceDimitris Papavasiliou2018-12-132-2/+121
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Clicks and pops of various volumes can be produced while the device is opened, closed, put into and taken out of standby, or reconfigured. Fix this, by implementing the digital_mute interface, so that the output is muted during such operations. Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org>
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-09 11:23:48 +0000