The tests require H264 to be enabled using the proprietary_codecs
GN args.gn option.
Bug: webrtc:11607, webrtc:13961
Change-Id: I22dc3d94c844873ac12b9dce8e88a97f4fcf7657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276046
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38133}
This reverts commit 6326c9c201c7331d68c9beb0a93f6f6e21063cd2.
Reason for revert: breaks upstream project
Original change's description:
> Add an active ICE controller that wraps a legacy controller (#7/n)
>
> The wrapping ICE controller will allow existing ICE controller implementations to migrate to the active interface, and eventually deprecate the legacy interface.
>
> Follow-up CL has unit tests for P2PTransportChannel using the new wrapping controller.
>
> Bug: webrtc:14367, webrtc:14131
> Change-Id: I6c517449ff1e503e8268a7ef91afda793723fdeb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275302
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Sameer Vijaykar <samvi@google.com>
> Cr-Commit-Position: refs/heads/main@{#38130}
Bug: webrtc:14367, webrtc:14131
Change-Id: I61dd98de62657852068c7566b55f19f662df9ff4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276043
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Sameer Vijaykar <samvi@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38131}
The wrapping ICE controller will allow existing ICE controller implementations to migrate to the active interface, and eventually deprecate the legacy interface.
Follow-up CL has unit tests for P2PTransportChannel using the new wrapping controller.
Bug: webrtc:14367, webrtc:14131
Change-Id: I6c517449ff1e503e8268a7ef91afda793723fdeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275302
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38130}
- Propagating `RtpPacketInfo::local_capture_clock_offset`, an
existing field that is related to the abs-capture-timestamp
header extension field `estimated_capture_clock_offset`
- Propagated through `SourceTracker::SourceEntry`
Bug: webrtc:10739, b/246753278
Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38129}
This add field trial string "skip_if_est_larger_than_fraction_of_max"
Dont send a probe if min(estimate, network state estimate) is larger than this
fraction of the set max bitrate.
Bug: webrtc:14392
Change-Id: I7333f6ef45ab0c019f21b9e4c604352219e1d025
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275940
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38123}
Controlled by a field trial, P2PTransportChannel can now use an active ICE controller instead of a legacy ICE controller.
P2PTransportChannel unit tests need non-trivial changes to exercise the refactored code path, so the testing changes are added in a follow-up CL.
Bug: webrtc:14367, webrtc:14131
Change-Id: I00d4930a5692c7d6d331ea9d6c2a2199304e363c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274701
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38114}
"ccmni" is the cellular interface name in MTK platform.
The accurate name may be ccmni0, ccmni1 ...etc. We would like
to add it in order to provide correct adapter type when user
invokes webrtc function GetAdapterTypeFromName.
Bug: b/246680543, webrtc:14438
Change-Id: I30074a442b3c02b93bc72a044675718955bd96f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275860
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38113}
Instead protect pending tasks with a shared object.
Some tests destroy VirtualSocket on a different thread than it is used on,
Some tests destroy VirtualSocket together with VirtualSocketServer after
associated thread is deleted, thus complicated check is used to ensure
VirtualSockets are safe to use.
Bug: webrtc:9702
Change-Id: I1a19cd24ac6a598a1cde64434104cad0b750096e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274460
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38103}
The reason for rejecting the congested frames in the first place, is
that they do not obey a Gaussian distribution around the line from the
Kalman filter. It therefore also does not make sense to include them
in the noise (*) estimation.
(*) noise = variation around the line from the Kalman filter.
Bug: webrtc:14151
Change-Id: Id8a44ba5f13bf9787ab54848109430ef7657f67a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275762
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38100}
The `TimestampExtrapolator` is only used by the `VCMTiming`
class, despite there being references to it from both
`modules/rtp_rtcp/BUILD.gn` and `modules/video_coding/BUILD.gn`.
Bug: webrtc:14111
Change-Id: If1a02a56a0c83b13d619ca08dc76c884fa829369
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38093}
The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.
This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.
Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
This functionality should have been added as part of the original CLs,
but was missed. The purpose of the validation is avoid catastrophic
failures due to misconfiguration (such as RTC_CHECK crashes).
The purpose is not to always provide practically reasonable values.
Bug: webrtc:14151
Change-Id: Icbddade865bd6a868f467a1df7055026935f36f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275560
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38090}
The value is today set to 200 which is too low for an audio packet to trigger sending probes.
For the initial probing, it would be good if audio packets, that may arrive before the first video frame can trigger sending a probe.
Also fix field trial parsing of required number of probes.
Bug: webrc:14392
Change-Id: I1f3cebcda38b71446e3602eef9cfa76de61a1ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38089}
Tests currently rely on the sorted order of connections held within the ICE controller, which sorts the connections by usability. The internal ordering is not part of the ICE controller contract.
Tests use the ordering as a proxy for certain expectations, so changed the tests to explicitly test the expectations.
Bug: webrtc:14367, webrtc:1413
Change-Id: Iaf33c61f6eb968c2c93a0265b6c48ad6218e23a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275304
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38088}
https://crrev.com/c/3885576 removed the last downstream consumer of the
constructor which took a ScreenCastPortal::CaptureSourceType. Now that
it has been removed, that constructor definition can also be removed and
the CaptureSourceType enum can be made private. There's still benefit in
storing and using this internally as the enum, since it's values match
that of the underlying system API.
The previously anonymous-namespaced function |ToCaptureSourceType| had
to be converted to a private static method as part of this change, since
it would be unable to access the type otherwise.
Fixed: chromium:1359411
Change-Id: I81ff24fbdddf9db02c9c5152d007dd82c194865a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274680
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38084}
