This is similar to what the old backend does when stopping. Given that we do not
fully understand the failure mode yet, this is a speculative fix.
Differential Revision: https://phabricator.services.mozilla.com/D169448
In m-c de8c14e4972f (Bug 1654112 - fix timestamp issues with RTP sources) we
added a call to packet_info.receive_time_ms() which is now deprecated.
receive_time_ms() is actually calling receive_time_.ms()
Differential Revision: https://phabricator.services.mozilla.com/D168292
This updates the version wpf-gpu-raster which adds support for
GPUs/drivers that use truncation instead of rounding when converting
vertices to fixed point.
It also adds the GL vendor to InitContextResult so that we can detect
AMD on macOS and tell wpf-gpu-raster that truncation is going to happen.
Differential Revision: https://phabricator.services.mozilla.com/D167503
Upstream commit: https://webrtc.googlesource.com/src/+/b82c45815a7c9fc4722e006219e8fdb58066ddb0
[TurnPort] Check if turn entry was found when deleting a connection.
[Merge to 108]
This is a simple way to avoid crashing, but the underlying issue
of why the entry has been removed, is a bit more complex to fix
and will be fixed in a follow-up CL.
(cherry picked from commit 9c606497d155d6304933517576f051f84d50e907)
Bug: chromium:1374310
Change-Id: I9dc0cf9e1acdcc3b3a205104346cc835b3f79c1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279283
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#38405}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281520
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5304@{#6}
Cr-Branched-From: 024bd84ca1cf7d3650c27912a3b5bfbb54da152a-refs/heads/main@{#38083}
Upstream commit: https://webrtc.googlesource.com/src/+/4005adc6d639035fdd5323861d9a2fc6bd946e99
[M107] ice: include tiebreaker in computation of foundation attribute
the foundation attribute is currently calculated as
CRC32(baseaddress, protocol, relayprotocol)
which is a way to satisfy the requirements from
https://www.rfc-editor.org/rfc/rfc5245#section-4.1.1.3
However, this leaks the base address which defeats the
MDNS obfuscation described in
https://datatracker.ietf.org/doc/draft-ietf-mmusic-mdns-ice-candidates/
since the CRC32 can be reversed using a table lookup as shown in
https://github.com/niespodd/webrtc-local-ip-leak/
To defeat that lookup, "seed" the CRC32 with the ICE tie-breaker which is a randomly picked unsigned 64 bit integer described in
https://www.rfc-editor.org/rfc/rfc5245#section-5.2
The tie-breaker is not known to Javascript and adding it scopes the foundation within the peer connection as described in section 4.1.1.3
To manually test (preferably with a DCHECK for IceTiebreaker() in ComputeFoundation)
- gather candidates twice on https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/ and observe that the foundations are not the same after this change
- create two RTCPeerConnections with {iceCandidatePoolSize: 1}, create a datachannel, call setLocalDescription, inspect the candidates and observe that the foundations are not the same after this change.
Unit test changes have been split into a separate CL for easier integration.
BUG=webrtc:14605,chromium:1378916
(cherry picked from commit 08b882d762edadb9797334859d915c5c1e34896b)
Change-Id: I6bbad1635b48997b00ae74d251ae357bf8afd12f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#38485}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281421
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/branch-heads/5304@{#5}
Cr-Branched-From: 024bd84ca1cf7d3650c27912a3b5bfbb54da152a-refs/heads/main@{#38083}
Upstream commit: https://webrtc.googlesource.com/src/+/eef098d1c7d50613d8bff2467d674525a9d0c57c
[Merge-107] [Stats] Avoid DCHECK crashing if SSRCs are not unique.
To properly handle SSRC collisions in non-BUNDLE we need to change how
RTP stats IDs are generated, but that is a riskier change to be dealt
with in a separate CL.
For now, we just make sure that crashing is not a possibility during
SSRC collisions as a mitigation for https://crbug.com/1361612. This is
achieved by adding a TryAddStats() method to RTCStatsReport returning
whether successful.
(cherry picked from commit da6297dc53cb2eaae7b1c5381652de9d707a7d48)
Bug: chromium:1361612
Change-Id: I8577ae4c84a7c1eb3c7527e9efd8d1b0254269a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275766
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#38197}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277900
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5304@{#4}
Cr-Branched-From: 024bd84ca1cf7d3650c27912a3b5bfbb54da152a-refs/heads/main@{#38083}
We already cherry-picked this when we vendored a22c2a0c58.
Upstream commit: https://webrtc.googlesource.com/src/+/ec86e953c4b1eb0efd5b0a06834eaa8f73d3660d
[merge to M107] Revert "rtp sender: don't send BYE on deactivating streams"
This reverts commit a22c2a0c581cbe3f612f7a7d9fb9840186cc1e06.
Reason for revert: breaks upstream project
Original change's description:
> rtp sender: don't send BYE on deactivating streams
>
> as this breaks RTCP assumptions about SSRCs being no longer
> active as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.6
>
> This should not be sent in reaction to temporarily disabling
> a stream via RTCRtpParameters.active as this does not mean that
> the participant is leaving the session as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
> and does not indicate end of participation as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.1
> which stipulates BYE should be the last packet sent from this SSRC.
>
> BUG=webrtc:11082
>
> Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38059}
(cherry picked from commit 03e6cccc28d76f28c926ce7cadaeba2c6a6cacf4)
Bug: webrtc:11082
Change-Id: Iaaff0c0d7bb857fe9ce78ebcc716f3c6f1bc5c4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275640
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Original-Commit-Position: refs/heads/main@{#38097}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276045
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5304@{#2}
Cr-Branched-From: 024bd84ca1cf7d3650c27912a3b5bfbb54da152a-refs/heads/main@{#38083}
Upstream commit: https://webrtc.googlesource.com/src/+/eb485a627145e2bd65e0b1428ec6301325ff5697
[107] VideoStreamEncoder: set at target quality based on codec.
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.
(cherry picked from commit e1a198b41dceab7818592b32288975489b3a9d12)
Bug: b/245029833, chromium:1364573
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-Original-Commit-Position: refs/heads/main@{#38091}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275775
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5304@{#1}
Cr-Branched-From: 024bd84ca1cf7d3650c27912a3b5bfbb54da152a-refs/heads/main@{#38083}
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