See go/deprecating-media-constraints for motivation.
Setting this min bitrate is necessary for BWE to work properly when
sending screencast in low BW scenarios. The value 100 kbps appears to be
a sensible default in practise (this is the value used by Google Meet).
In order for apps not to have to rely on non-standard APIs
(googScreencastMinBitrate) for BWE to work properly, we change the
default to 100 kbps. This will unblock deprecating and removing legacy
mediaConstraints.
A kill switch is added in case this causes any unforeseen issues, but if
all goes well we can remove the kill switch in the next milestone.
Bug: chromium:1315155
Change-Id: I02b4eb0dfb26f934e678760313a0423f412512c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36523}
Modify cl/ a bit and add fieldtrialsstring on observer
not to break downstream projects.
---
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 14/inf
This cl/ passes field trials all the way from c++
to the android NetworkMonitorAutoDetect.java
Bug: webrtc:10335
Change-Id: Ic6842612eed36b684340f0f78f4087bee249cc50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36498}
---
Bug: webrtc:10335
Change-Id: Ied43770977465a0042541a61d29a9015c0b9cdc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258622
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36520}
Now that FieldTrialsView is (almost) a supported object,
let downstream tests use the ScopedKeyValueConfig for
testing.
Bug: webrtc:10335
Change-Id: I7bf7577387f1123b473275990ca32b80a8ddf9bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36519}
This makes moving from sigslot to CallbackList slightly simpler in some
situations.
Bug: webrtc:11943
Change-Id: I5c6dafb8ac597a119b90b64f369fa9e6316e38da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36515}
A managed device might have camera access restricted, which results in
the following crash:
Caused by android.os.ServiceSpecificException: validateClientPermissionsLocked:1044: Callers from device user 0 are not currently allowed to connect to camera "1"
at android.os.Parcel.createException(Parcel.java:2085)
at android.os.Parcel.readException(Parcel.java:2039)
at android.os.Parcel.readException(Parcel.java:1987)
at android.hardware.ICameraService$Stub$Proxy.connectDevice(ICameraService.java:624)
at android.hardware.camera2.CameraManager.openCameraDeviceUserAsync(CameraManager.java:389)
at android.hardware.camera2.CameraManager.openCameraForUid(CameraManager.java:588)
at android.hardware.camera2.CameraManager.openCamera(CameraManager.java:516)
at org.webrtc.Camera2Session.openCamera(Camera2Session.java:359)
at org.webrtc.Camera2Session.start(Camera2Session.java:322)
at org.webrtc.Camera2Session.<init>(Camera2Session.java:298)
at org.webrtc.Camera2Session.create(Camera2Session.java:276)
at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:214)
at android.os.HandlerThread.run(HandlerThread.java:67)
Change-Id: I5e7b8d238e9381d1f8a4fe9858e8eb480d578fa0
Bug: webrtc:13950
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258363
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36513}
There doesn't seem to be any tests currently running with
VideoToolbox H.264 codec.
Add a minimal test to make sure there are no regressions.
Bug: webrtc:13934
Change-Id: I664833d06e9b4f919b0662781891f5448e6057c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257922
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36511}
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/ passes field trials all the way from c++
to the android NetworkMonitorAutoDetect.java
Bug: webrtc:10335
Change-Id: Ic6842612eed36b684340f0f78f4087bee249cc50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36498}
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}
The test may spawn threads that reference the global clock, so the
code that resets the fake clock, needs to run when those threads
have been stopped.
Bug: webrtc:13947
Change-Id: I1f7dc921b7a8ff8e7e89e03e4397e635b67d77be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258361
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36489}
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}
An assert that the same tests are running before and after the migration is failing.
The new config was adding "video_capture_test".
Also add bot 'Android64 Builder arm64' instead of 'Android32 Builder arm64'.
Bug: webrtc:13899
Change-Id: Ic87576ab9740c0dc138f0b087e97ee0ae4be3376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258138
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36484}
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}
Move global state into a singleton class. Eliminates use of
GlobalMutex. Also use std::atomic rather than volatile, for improved
thread safety.
Bug: webrtc:11567
Change-Id: I305d16ed2f4a9a6497df119e66d9bba08337339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258120
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36475}
