- 26 Jan, 2021 4 commits
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Dmitry Osipenko authored
The "chip" can't be NULL in hda_tegra_runtime_resume() because code would crash otherwise. Let's remove the unnecessary check in order to clean up code a tad. Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested Suggested-by:
Thierry Reding <treding@nvidia.com> Signed-off-by:
Dmitry Osipenko <digetx@gmail.com> Link: https://lore.kernel.org/r/20210120003154.26749-4-digetx@gmail.comSigned-off-by:
Takashi Iwai <tiwai@suse.de>
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Dmitry Osipenko authored
Reset hardware on RPM-resume in order to bring it into a predictable state. Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested Signed-off-by:
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Dmitry Osipenko authored
Use clk_bulk helpers to make code cleaner. Note that this patch changed the order in which clocks are enabled to make code look nicer, but this doesn't matter in terms of hardware. Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested Acked-by:
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Takashi Sakamoto authored
Drivers in ALSA firewire stack supports eventing to userspace applications via ALSA hwdep interface. All of the drivers supports stream lock events. Some of them supports their unique events according to specification of target device. ALSA bebob driver supports the stream lock event only. In the case, it's enough to check condition only in loop with process blocking. However, current implementation check it again after breaking the loop. This commit removes the redundant check. Reported-by:
Dan Carpenter <dan.carpenter@oracle.com> Reported-by:
Christophe JAILLET <christophe.jaillet@wanadoo.fr> Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20210125140208.26318-1-o-takashi@sakamocchi.jpSigned-off-by:
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- 25 Jan, 2021 3 commits
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Dan Carpenter authored
Smatch complains that "count" is not clamped when "ff->dev_lock_changed" and it leads to an information leak. Fortunately, that's not actually possible and the condition can be deleted. Signed-off-by:
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Dan Carpenter authored
Smatch complains that "count" isn't clamped properly and "oxfw->dev_lock_changed" is false then it leads to an information leak. But it turns out that "oxfw->dev_lock_changed" is always set and the condition can be removed. Signed-off-by:
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Jiapeng Zhong authored
Fix the following coccicheck warnings: ./sound/pci/hda/hda_intel.c:2309:3-23: WARNING: Assignment of 0/1 to bool variable. Reported-by:
Abaci Robot <abaci@linux.alibaba.com> Signed-off-by:
Jiapeng Zhong <abaci-bugfix@linux.alibaba.com> Link: https://lore.kernel.org/r/1611559047-106928-1-git-send-email-abaci-bugfix@linux.alibaba.comSigned-off-by:
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- 24 Jan, 2021 1 commit
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Jiapeng Zhong authored
Fix the following coccicheck warnings: ./sound/pci/hda/patch_conexant.c:570:2-20: WARNING: Assignment of 0/1 to bool variable. Reported-by:
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- 21 Jan, 2021 2 commits
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Takashi Iwai authored
At probing a UAC2/UAC3 device like NUX MG-300 USB interface, we get error messages "RANGE setting not yet supported". It comes the place where the driver tries to determine the resolution of mixer volumes via SET_CUR_RES and GET_CUR_RES verbs. Those verbs aren't supported on UAC2 and UAC3, hence the driver warns like the above. Although the driver handles this error and works as expected, it's still ugly to show such errors unnecessarily. This patch papers over the errors by applying the resolution detection only for UAC1 and skipping it for UAC2/UAC3. Reported-by:
Mike Oliphant <oliphant@nostatic.org> Link: https://lore.kernel.org/r/20210120213932.1971-2-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
The current USB-audio driver gets an error at probing NUX MG-300 about parsing the clocks. This is because the firmware doesn't return the proper connection of the clock selector that is connected to a single clock; it's likely that the firmware was lazy^w optimized and the inquiry wasn't handled. Actually it makes little sense to inquire and set up the single connection explicitly. This patch fixes the issue by simply skipping the clock selector inquiry if it's a single connection. Reported-by:
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- 18 Jan, 2021 5 commits
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Olivia Mackintosh authored
This adds the Pioneer DJ DJM-750 to the quirks table and ensures skip_pioneer_sync_ep() is (also) called: this device uses the vendor ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many just like to be awkward it seems). Playback on all 8 channels works. I'll likely keep this working in the future and submit futher patches and improvements as necessary. Signed-off-by:
Olivia Mackintosh <livvy@base.nu> Link: https://lore.kernel.org/r/20210118130621.77miiie47wp7mump@base.nuSigned-off-by:
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Takashi Iwai authored
Back-merge of 5.11 devel branch for more works on USB-audio. Signed-off-by: -
Takashi Iwai authored
For addressing the regression on Pioneer devices, we recently corrected the quirk code to enable the implicit feedback mode on those devices properly. However, the devices still showed problems with the full duplex operations with JACK, and after debug sessions, we figured out that the older kernels that had worked with JACK also didn't use the implicit feedback mode at all although they had the quirk code to enable it; instead, the old code worked just to skip the normal sync endpoint setup that would have been detected without it. IOW, what broke without the implicit-fb quirk in the past was the application of the normal sync endpoint that is actually the capture data endpoint on these devices. This patch covers the overseen piece: it modifies the quirk code again not to enable the implicit feedback mode but just to make the driver skipping the sync endpoint detection. This made the driver working with JACK full-duplex mode again. Still it's not quite clear why the implicit feedback doesn't work on those devices yet; maybe it's about some issues in the URB setup. But at least, with this patch, the driver should work in the level of the older kernels again. Fixes: 167c9dc8 ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices") Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
The UAC2/3 sample rate setup is based on the clock node, which is usually shared in the interface, and can't be re-setup without deselecting the interface once, and that's how the current code behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence we basically need to call for each endpoint usage even if those share the same interface. This patch fixes the behavior of UAC1 to call always snd_usb_init_sample_rate() in snd_usb_endpoint_configure(). Fixes: bf6313a0 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
The current sample rate setup function for UAC1 assumes only the first endpoint retrieved from the interface:altset pair, but the rate set up may be needed also for the secondary endpoint. Also, retrieving the endpoint number from the interface descriptor is redundant; we have already the target endpoint in the given audioformat object. This patch simplifies the code and corrects the target endpoint as described in the above. It simply refers to fmt->endpoint directly. Also, this patch drops the pioneer_djm_set_format_quirk() that is caleld from snd_usb_set_format_quirk(); this function does the sample rate setup but for the capture endpoint (0x82), and that's exactly what the change above fixes. Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.deSigned-off-by:
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- 15 Jan, 2021 7 commits
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Takashi Iwai authored
LPE driver still has explicit calls of dma_set_mask() and dma_set_coherent_mask(). Let's simplify with dma_set_mask_and_coherent(). Link: https://lore.kernel.org/r/20210114125412.993-3-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
Many PCI drivers still have two explicit calls of dma_set_mask() and dma_set_coherent_mask(). Let's simplify with dma_set_mask_and_coherent(). Link: https://lore.kernel.org/r/20210114125412.993-2-tiwai@suse.deSigned-off-by:
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Kai-Chuan Hsieh authored
Add HD Audio Device PCI ID for the Intel Cometlake-R platform Reviewed-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by:
Kai-Chuan Hsieh <kaichuan.hsieh@canonical.com> Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.comSigned-off-by:
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Takashi Sakamoto authored
I-ONIX FW810s was shipped in Lexicon brand of HARMAN International industries, Inc 2009. The model uses TCD2220 ASIC as its communication engine. TCAT general protocol is supported, its extension isn't. This patch adds support for the model with hard-coded stream formats. $ python3 ~/git/linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04042b91 bus_info_length 4, crc_length 4, crc 11153 404 31333934 bus_name "1394" 408 e0008102 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512) 40c 000fd720 company_id 000fd7 | 410 007d7ecf device_id 20007d7ecf | EUI-64 000fd720007d7ecf root directory ----------------------------------------------------------------- 414 00064c2d directory_length 6, crc 19501 418 03000fd7 vendor 41c 8100000a --> descriptor leaf at 444 420 17000001 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 000438f2 directory_length 4, crc 14578 434 12000fd7 specifier id 438 13000001 version 43c 17000001 model 440 8100000d --> descriptor leaf at 474 descriptor leaf at 444 ----------------------------------------------------------------- 444 000489d5 leaf_length 4, crc 35285 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4c657869 "Lexi" 454 636f6e00 "con" descriptor leaf at 458 ----------------------------------------------------------------- 458 0006594b leaf_length 6, crc 22859 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 492d4f4e "I-ON" 468 49585f46 "IX_F" 46c 57383130 "W810" 470 53000000 "S" descriptor leaf at 474 ----------------------------------------------------------------- 474 0006594b leaf_length 6, crc 22859 478 00000000 textual descriptor 47c 00000000 minimal ASCII 480 492d4f4e "I-ON" 484 49585f46 "IX_F" 488 57383130 "W810" 48c 53000000 "S" Signed-off-by: -
Takashi Iwai authored
The last remaining usage of strlcpy() in USB-audio driver is the setup of the card longname string. Basically we need to know whether any non-empty string is set or not, and no real length is needed. Refactor the code and use strscpy() instead. After this change, strlcpy() is gone from all sound/* code. Link: https://lore.kernel.org/r/20210115100437.20906-1-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
USB-audio driver still contains two calls of strlcpy() because the return size is evaluated. Basically it just checks whether the string is copied or not, but since strcpy() may return a negative error code, we should check the negative value and treat as filled. Link: https://lore.kernel.org/r/20210115095758.19707-1-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
snd_seq_oss_synth_make_info() didn't check the error code from snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy() with the uninitialized string as the source, which may lead to the access over the limit. Add the proper error check for avoiding the failure. Reported-by: syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.deSigned-off-by:
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- 14 Jan, 2021 2 commits
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Takashi Iwai authored
It turned out that VIA codecs also mute the sound in the lowest mixer level. Turn on the dac_min_mute flag to indicate the mute-as-minimum in TLV like already done in Conexant and IDT codecs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.deSigned-off-by:
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Chris Chiu authored
The Acer Apire E5-575T laptop with codec ALC255 has a terrible background noise comes from internal mic capture. And the jack sensing dose not work for headset like some other Acer laptops. This patch limits the internal mic boost on top of the existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T. Signed-off-by:
Chris Chiu <chiu@endlessos.org> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.orgSigned-off-by:
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- 13 Jan, 2021 2 commits
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Takashi Iwai authored
Since the commit 5a6c3e11 ("ALSA: usb-audio: Add hw constraint for implicit fb sync"), we apply the hw constraints for the implicit feedback sync to make the secondary open aligned with the already opened stream setup. This change assumed that the secondary open is performed after the first stream has been already set up, and adds the hw constraints to sync with the first stream's parameters only when the EP setup for the first stream was confirmed at the open time. However, most of applications handling the full-duplex operations do open both playback and capture streams at first, then set up both streams. This results in skipping the additional hw constraints since the counter-part stream hasn't been set up yet at the open of the second stream, and it eventually leads to "incompatible EP" error in the end. This patch corrects the behavior by always applying the hw constraints for the implicit fb sync. The hw constraint rules are defined so that they check the sync EP dynamically at each invocation, instead. This covers the concurrent stream setups better and lets the hw refine calls resolving to the right configuration. Also this patch corrects a minor error that has existed in the debug print that isn't built as default. Fixes: 5a6c3e11 ("ALSA: usb-audio: Add hw constraint for implicit fb sync") Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.deSigned-off-by:
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Kai Vehmanen authored
Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P. Signed-off-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by:
Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel.comSigned-off-by:
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- 12 Jan, 2021 6 commits
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Alex Deucher authored
We are able to power down the GPU and audio via the GPU driver so flag these asics as supporting runtime pm. Reviewed-by:
Evan Quan <evan.quan@amd.com> Signed-off-by:
Alex Deucher <alexander.deucher@amd.com> Link: https://lore.kernel.org/r/20210105175245.963451-1-alexander.deucher@amd.comSigned-off-by:
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Geert Uytterhoeven authored
As snd_fw_async_midi_port.consume_bytes is unsigned int, and NSEC_PER_SEC is 1000000000L, the second multiplication in port->consume_bytes * 8 * NSEC_PER_SEC / 31250 always overflows on 32-bit platforms, truncating the result. Fix this by precalculating "NSEC_PER_SEC / 31250", which is an integer constant. Note that this assumes port->consume_bytes <= 16777. Fixes: 531f4718 ("ALSA: firewire-lib/firewire-tascam: localize async midi port") Reviewed-by:Geert Uytterhoeven <geert+renesas@glider.be> Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.beSigned-off-by:
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Geert Uytterhoeven authored
As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L, the second multiplication in ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250 always overflows on 32-bit platforms, truncating the result. Fix this by precalculating "NSEC_PER_SEC / 31250", which is an integer constant. Note that this assumes ff->rx_bytes[port] <= 16777. Fixes: 19174295 ("ALSA: fireface: add transaction support") Reviewed-by: -
Peter Geis authored
Currently hda on tegra30 fails to open a stream with an input/output error. For example: speaker-test -Dhw:0,3 -c 2 speaker-test 1.2.2 Playback device is hw:0,3 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 16384 Period size range from 32 to 8192 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error The tegra-hda device was introduced in tegra30 but only utilized in tegra124 until recent chips. Tegra210/186 work only due to a hardware change. For this reason it is unknown when this issue first manifested. Discussions with the hardware team show this applies to all current tegra chips. It has been resolved in the tegra234, which does not have hda support at this time. The explanation from the hardware team is this: Below is the striping formula referenced from HD audio spec. { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } The current issue is seen because Tegra HW has a problem with boundary condition (= 8) for striping. The reason why it is not seen on Tegra210/Tegra186 is because it uses max 2SDO lines. Max SDO lines is read from GCAP register. For the given stream (channels = 2, bps = 16); ratio = (channels * bps) / NSDO = 32 / NSDO; On Tegra30, ratio = 32/4 = 8 (FAIL) On Tegra210/186, ratio = 32/2 = 16 (PASS) On Tegra194, ratio = 32/4 = 8 (FAIL) ==> Earlier workaround was applied for it If Tegra210/186 is forced to use 4SDO, it fails there as well. So the behavior is consistent across all these chips. Applying the fix in [1] universally resolves this issue on tegra30-hda. Tested on the Ouya game console and the tf201 tablet. [1] commit 60019d8c ("ALSA: hda/tegra: workaround playback failure on Tegra194") Reviewed-by:Jon Hunter <jonathanh@nvidia.com> Tested-by:
Ion Agorria <ion@agorria.com> Reviewed-by:
Sameer Pujar <spujar@nvidia.com> Acked-by:
Peter Geis <pgwipeout@gmail.com> Link: https://lore.kernel.org/r/20210108135913.2421585-3-pgwipeout@gmail.comSigned-off-by:
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Peter Geis authored
Current implementation defaults the hda clocks to clk_m. This causes hda to run too slow to operate correctly. Fix this by defaulting to pll_p and setting the frequency to the correct rate. This matches upstream t124 and downstream t30. Acked-by:
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Jonathan Neuschäfer authored
MIXART.txt has been converted to ReST and renamed. Fix the reference in alsa-configuration.rst. Fixes: 3d8e8186 ("ALSA: doc: ReSTize MIXART.txt") Signed-off-by:
Jonathan Neuschäfer <j.neuschaefer@gmx.net> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210101221942.1068388-1-j.neuschaefer@gmx.netSigned-off-by:
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- 08 Jan, 2021 6 commits
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Takashi Iwai authored
Pioneer devices have both playback and capture streams sharing the same iface/altsetting, and those need to be paired as implicit feedback. Instead of a half-baked (and broken) static quirk entry, set up more generically for those devices by checking the number of endpoints and the attribute of the secondary EP. Fixes: bf6313a0 ("ALSA: usb-audio: Refactor endpoint management") Reported-by:
František Kučera <konference@frantovo.cz> Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
There are devices that have multiple endpoints sharing the same iface/altset not only for sync but also for the actual streams, and the audioformat for such an endpoint needs to be handled with the proper endpoint index; otherwise it confuses the endpoint management. This patch extends the audioformat to annotate the endpoint index, and put the proper ep_idx=1 to Pioneer device quirk entries accordingly. Fixes: bf6313a0 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
The current endpoint handling assumed (more or less) a unique 1:1 relation between the endpoint and the iface/altset. The exception was the sync EP without the implicit feedback which has usually the secondary EP of the same altset. This works fine for most devices, but it turned out that some unusual devices like Pinoeer's ones have both playback and capture endpoints in the same iface/altsetting and use both for the implicit feedback mode. For handling such a case, we need to extend the endpoint management to take the shared interface into account. This patch does that: it adds a new object snd_usb_iface_ref for managing the reference counts of the each USB interface that is used by each endpoint. The interface setup is performed only once for the (sharing) endpoints, and the doubly initialization is avoided. Along with this, the resource release of endpoints and interface refcounts are put into a single function, snd_usb_endpoint_free_all() instead of looping in the caller side. Fixes: bf6313a0 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
The implicit feedback mode needs to handle two endpoints and the choice of the audioformat object for the sync EP is important since this determines the compatibility of the hw_params. The current code uses the same audioformat object if both the main EP and the sync EP point to the same iface/altsetting. This was done in consideration of the non-implicit-fb sync EP handling, and it doesn't match well with the cases where actually to endpoints are defined in the sameiface / altsetting like a few Pioneer devices. Modify snd_usb_find_implicit_fb_sync_format() to pick up the audioformat that is assigned in the counter-part substreams primarily, so that the actual capture stream can be opened properly. We keep the same audioformat object only as a fallback in case nothing found, though. Fixes: 9fddc15e ("ALSA: usb-audio: Factor out the implicit feedback quirk code") Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.deSigned-off-by:
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Takashi Iwai authored
The recent change in the endpoint management moved the endpoint object creation from the stream open time to the parser of the audio descriptor. It works fine for the standard audio, but it overlooked the other places that create audio streams via quirks (QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices; those call snd_usb_add_audio_stream() manually, hence they miss the endpoints, eventually resulting in the error at opening streams. Moreover, now the sync EP setup was moved to the explicit call of snd_usb_audioformat_set_sync_ep(), and this needs to be added for those places, too. This patch addresses those regressions for quirks. It adds a local helper function add_audio_stream_from_fixed_fmt(), which does the all needed tasks, and replaces the calls of snd_usb_add_audio_stream() with this new function. Fixes: 54cb3190 ("ALSA: usb-audio: Create endpoint objects at parsing phase") Reported-by:
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Joe Perches authored
strlcpy is deprecated. see: Documentation/process/deprecated.rst Change the calls that do not use the strlcpy return value to the preferred strscpy. Done with cocci script: @@ expression e1, e2, e3; @@ - strlcpy( + strscpy( e1, e2, e3); This cocci script leaves the instances where the return value is used unchanged. After this patch, sound/ has 3 uses of strlcpy() that need to be manually inspected for conversion and changed one day. $ git grep -w strlcpy sound/ sound/usb/card.c: len = strlcpy(card->longname, s, sizeof(card->longname)); sound/usb/mixer.c: return strlcpy(buf, p->name, buflen); sound/usb/mixer.c: return strlcpy(buf, p->names[index], buflen); Miscellenea: o Remove trailing whitespace in conversion of sound/core/hwdep.c Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcprmknw@mail.gmail.com/Signed-off-by:
Joe Perches <joe@perches.com> Acked-by:
Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/22b393d1790bb268769d0bab7bacf0866dcb0c14.camel@perches.comSigned-off-by:
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- 07 Jan, 2021 2 commits
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Takashi Iwai authored
Back-merge of 5.11-devel branch for syncing the result changes. Signed-off-by: -
Takashi Iwai authored
Merge tag 'asoc-fix-v5.11-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.11 A collection of mostly driver specific fixes, plus a maintainership update for TI and a fix for DAPM driver removal paths.
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