There is a crash report from the Windows OS API
where it repro only win10, not a Win11.
Unfortunately, Microsoft can't access the dump file or
hasn't repro internally so we decided to disable the WGC
fallback use in the OS other than Win11 now.
Once the change (support WGC fallback) reaches to Microsoft Edge
and it produce crash report, Edge team will take the
dump file to the Windows OS API owner for Win10 level fix (or
bring the Win11 fix to Win10).
Bug: chromium:1312937
Change-Id: I5335e2c57076d4fab08e9c74ade599259cff10d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258821
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36558}
XDG desktop portal utils implicitly rely on few portal interfaces.
This change makes those interfaces explicit.
Bug: chromium:1291247
Change-Id: I3c300f33d04bfa8857ac9f1f9d634dbfc077e591
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258720
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36557}
Versions of Windows before Win11 will crash when `CreateForMonitor` is
called, but the system has no attached displays. This can be avoided by
adding a check to ensure at least one display is found before we return
true in `IsMonitorValid`. Previously we would early return `true` if the
"monitor" we were checking was the `kFullDesktopScreenId`.
Bug: chromium:1316478
Change-Id: I2562fe3834db574cf3706ee1d604472ac03f9ff3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258920
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36555}
This CL adds flag "max_queue_time", e.g:
WebRTC-SlackedTaskQueuePacedSender/Enabled:true,max_queue_time:75ms
If the PacingController's ExpectedQueueTime() is greater than or equal
to "max_queue_time" then we will use high precision delayed tasks to
ensure a smoother (less bursty) pacing of packets.
This should only get triggered in ultra high definition scenarios or
temporarily during key frames being sent. I have confirmed manually that
with the flag set to 75 ms I get low precision when not sending key
frames and temporarily get high precision for 5-15 delayed tasks during
the sending of a key frame.
Bug: webrtc:13957
Change-Id: I3c8aeba5fe18812ed0187610cd3f92a375bc6f18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258623
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36538}
Adds two metrics for stereo detection:
- An enum indicating whether the last 10 seconds contained persistent stereo content or not, logged every 10 seconds.
- An enum indicating whether any persistent stereo content at all has been detected, logged at the end of the AEC lifetime.
These metrics allow us to assess:
- What proportion of all audio is treated as stereo.
- What proportion of sessions encounter any significant stereo content. If this is unexpectedly high, the stereo detection code may need fine tuning.
Metrics are only logged for component lifetimes exceeding 5 seconds: This is to avoid brief AEC lifetimes due to internal resets etc within APM.
Corresponding Chrome CL for XML histogram declarations:
https://crrev.com/c/3579317
Bug: chromium:1295710
Change-Id: I93e2bf74588cf4bb2a8922dbfad079bccab01456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258760
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36537}
Remote desktop wrapper needs to create a barrier on when the screencast
portal is done (either succeeded or failed). This change adds an initial
enum to differentiate it from other values. CL also generalizes the
helper `PortalFailed` to `OnPortalDone` so as to account for both
failure/success scenarios.
Bug: chromium:1291247
Change-Id: I188f72533e75a88c9b30ce2bb093dae548cef7b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258540
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36526}
The Windows 2019 Enterprise SKU is based on RS5 but is missing support
for WGC. This changes adds a function IsWgcSupported which will check
that the API is actually present, which should be more robust than just
performing a version check.
Bug: webrtc:13932
Change-Id: Ib6a2278f316b9b86671df77fd37468c79564d655
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258163
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36506}
During temporary stereo content when the AEC3 uses a mono reference signal, the signal is downmixed by averaging instead of using only the left channel.
Additionally, temporary stereo content is flagged as an echo path change.
Tested: Modified local build: Verified stereo mode entered / left in accordance with hysteresis and timeout thresholds. Verified temporary stereo detected during temporary stereo playout. Made an aecdump and inspected content.
Bug: chromium:1295710
Change-Id: I6bd53e615dfb3ec39bc1c73275b7d6d599ac7c57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36504}
Even if playout audio is only very briefly stereo, the AEC will enter stereo processing mode. To save CPU and improve AEC performance, this CL adds a hysteresis period before treating playout as stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I29116ab2e7823e25a02aa3b66a1c619f1d966d9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258479
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36503}
If playout audio is temporarily stereo, the AEC will currently enter stereo processing mode indefinitely. To save CPU and improve AEC performance, this CL adds support for falling back to mono after a period of no stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I690b5b22f8407f950bf41f3bcaa9ca0138452157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258421
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36502}
This CL adds a component in the TS implementation to return a delayed
version of the voice probability values observed when `Suppress()` is
called. That is needed in order to temporally align the voice
probability values to the processed audio since TS adds algorithmic
delay.
Bug: webrtc:13663
Change-Id: I5041ace3939d2ce7ba084ae703428e66f1aa06be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255860
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36496}
The values returned by `TransientSuppressor::Initialize()` and by
`TransientSuppressor::Suppress()` are never used.
Bug: webrtc:13663
Change-Id: I20b8afb5a66f49e5ebaf132acf8bcd1c4292a5f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255822
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36492}
More robust API option that allows to fully initialize TS when created.
Bug: webrtc:13663
Change-Id: I42c38612ef772eb6d0bbde49d04ea39332a0e3c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255821
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36490}
Adding a delay unit to be used in the APM Transient Suppressor (TS)
sub-module through which the observerd voice probabilities are
temporally aligned to the audio processed by TS, which introduces
algorithmic delay.
Bug: webrtc:13663
Change-Id: I2136c303914580851c742d8db89478a13b06dacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255680
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36487}
It is now required to specify which VAD is used to compute the speech
probability passed when `TransientSuppressor::Suppress()` is called.
In this way, it is possible to adapt parameters and/or logic of a
`TransientSuppressor` implementation to the behavior of the used
VAD. This CL also adds a "no VAD" mode option, which ignores the speech
probability argument passed when `Suppress()` and always applies mild
suppression to preserve transparency.
Finally, this CL adds a field trial to choose which VAD is used by
APM for transient suppression. Wiring the RNN VAD to TS will be done
in a follow-up CL.
Bug: webrtc:13663
Change-Id: I21ed49f91875a4ee0f04db97ea87c0dbc3db7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250962
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36485}
This is a slight change in behavior that fixes a bug where all expansions are not counted due to more than 10ms available in the sync buffer, which can happen after repeated expansions.
The counter should also be updated when in muted mode.
Bug: webrtc:13322
Change-Id: I067689ee251d3d1ae990a27cdd271f718b0d6f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257360
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36483}
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch
Both features are enabled by default in the AEC3 config.
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
Apart from making the construction more straightforward, this change allows recreating the BlockProcessor at runtime. This is used to change parameterization at runtime in an upcoming CL [1].
[1] https://webrtc-review.googlesource.com/c/src/+/258129
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I2e0275c5c97044cb4370042633266b193c06b960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258100
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36473}
This CL sets speed 9 for all resolutions when two or less cores are
available, as a heuristic for a "slow" machine.
This gives a large speed bost at a relatively small quality loss.
A field-trial kill-switch is available to override this behavior.
Bug: webrtc:13888
Change-Id: I24278a45de000ad7984d0525c47d9eb6b9ab6b60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257421
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36466}
to support clients (e.g. RTCPReceiver) that collect and report RTT per sender ssrc.
Bug: webrtc:8239, webrtc:13853
Change-Id: I907fb35277b0f23bbe9f2cd2ef979ce0fb1f9338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257440
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36439}
In r36379 a change to per-resolution setting of denoising was introduced
that unintentionally enabled denoising on lower resolutions in the case
that VideoCodec::VP9()->denoising was false.
The CL makes sure the per-resolution setting are only allowed to
disable denoising, not enable it.
Bug: webrtc:13888
Change-Id: Ice07a5a7d27798dc2182a40af0ec521bde6210b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257303
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36412}
Change adapts the `base_capturer_pipewire` so that a portal can be
injected in the capturer. This allows the remoting to inject its
own portal for the purpose of capturing desktop stream as long
as the injected portal provides implementation of the new interface
that is added as part of this change.
Additionally, a method has been exposed on the capturer to get
details about the portal session so that the remoting
implementation can use the same underlying session for controlling
inputs on the remote host.
Finally, desktop capturer interface is extended with a generic
method `GetMetadata` that is used to retrieve session related
information by CRD and relay it over to its input injector. Clients
provide override for the method and it eventually invokes the
underlying `GetSessionDetails` method on the portal instance.
Bug: chromium:1291247
Change-Id: I0dbd154eb16d4149f967c4a818eea51e7e6eb9a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36399}
This cl adds a new field trial parameter to WebRTC-Bwe-EstimateBoundedIncrease that allows the delay based estimate to immediately increase to the upper link capacity estimate.
Bug: none
Change-Id: I875121a41f65cc8e76bb87bbf421731ba6b4551b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257142
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36392}
This is used until the first RTT measurement becomes available.
100ms is a reasonable default and used in other places.
Bug: webrtc:10178
Change-Id: I14f530504a4866fbe75f025dfe184fd6e296b75e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36384}
Recent Clang versions have enhanced -Wunused-but-set-variable which now
warns about this.
Bug: chromium:1309955
Change-Id: Ie70df85f5a6d2cbabb4e10960bfd926ff7bf32cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257162
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36381}
This CL changes the default speed settings for TL0/TL[1-2] from
7/8 to 9/9 at 1080p resolutions and up. We also disable the denoiser
at these resolutions.
Settings can be overriden using existing WebRTC-VP9-PerformanceFlags
field trial.
Bug: webrtc:13888
Change-Id: I70f19efdace88d70bbb90bc6dd5149653eb079c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257141
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36379}
As stashed frames are retried their `tl0_pic_idx` are again unwrapped which can lead to the `tl0_unwrapper_` to unwrap the `tl0_pic_idx` of newer frames backwards. Instead unwrap the `tl0_pid_idx` only once and save it with the frame if necessary.
Related VP9 CL: https://webrtc-review.googlesource.com/c/src/+/253844
Bug: none
Change-Id: I8265dc5f36ee257db92d79cec719f56b165d3855
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256966
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36356}
Since 12.2, the orange/red indicator at the top right of the screen shows up as a window in the Chrome getDisplayMedia() picker, as it's not filtered out by the existing filters. Screenshots in the bug.
Bug: chromium:1297731, b/218211225
Change-Id: I0f87e8d2af42a5a2e3d84f69fe73596e9cf35622
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251841
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#36350}
