Upstream commit: https://webrtc.googlesource.com/src/+/d8f2b0380b3ec980af35ce4b92ba6a211ec8c76d
[M118] Fire MaybeSignalReadyToSend in a PostTask when recursive
Speculative fix. Writing the test for it is more complex.
(cherry picked from commit 83894d384763c613e548e6352838406e6e0c2fc1)
Bug: chromium:1483874
Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#40820}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322124
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5993@{#3}
Cr-Branched-From: 5afcec093c1403fe9e3872706d04671cbc6d2983-refs/heads/main@{#40703}
Upstream commit: https://webrtc.googlesource.com/src/+/7349579c4272d22b88c7dbb497b9bc2acfc5a26f
[M118] FrameCadenceAdapter: stop delayed refresh frame calls on dtor.
The FrameCadenceAdapter starts a delayed task to request a
new refresh frame on receiving frame drop. However, the
resulting RepeatingTaskHandle was not Stop()ed on destruction,
leading to UAF.
(cherry picked from commit fb98b01061e7eec51a800b53d4346827f89336a5)
Fixed: chromium:1478944
Change-Id: Iba441420953e989cfc7fcfd2f358b5b30f375786
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#40747}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320420
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5993@{#1}
Cr-Branched-From: 5afcec093c1403fe9e3872706d04671cbc6d2983-refs/heads/main@{#40703}
Upstream commit: https://webrtc.googlesource.com/src/+/48df56e9ac503bea9efea269656932f2d00de7cd
Remove SignalSSLHandshakeError signal from SSLStreamAdapter.
Also removing has_slots depdency from OpenSSLStreamAdapter and moving
it to the OpenSSLStreamAdapter subclass where it's still needed.
Bug: webrtc:11943
Change-Id: Ibcae5ea1efff146d78b32bb0eca63d7f44ed08c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318885
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40702}
Upstream commit: https://webrtc.googlesource.com/src/+/7b6faa12435d5f3e9143158c520450ee579f87c3
Move assignment of a streams random-msid
move this a bit later in the process since the current handling will consider two ssrc-lines with a cname in the same RTX FID ssrc-group to be part of separate streams due to the different randomly assigned msids. This leads to a misdetection as plan-b SDP.
BUG=None
Change-Id: Ie8acce9c2c7fb9eabda479b90e8cc7406dcb1696
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40701}
Upstream commit: https://webrtc.googlesource.com/src/+/85c05a8a1702d2e94daf6436043523998ba413b0
Update RemoteBitreateEstimatorAbsSendTime to use BitrateTracker
BitrateTracker uses same implementation as RateStatistics, but provides api using Timestamp and DataRate types instead of plain numbers
Bug: webrtc:13756
Change-Id: Ie37fa58ede7590f870ec4376a64e7cf2c94431d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318841
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40697}
Upstream commit: https://webrtc.googlesource.com/src/+/59574ca6d36f4769ecc734a5a52cdbb23f04d971
Add absl::AnyInvocable to SSLStreamAdapter::Create
Remove internal use of SignalSSLHandshakeError and prepare removal of
sigslot dependency from SSLStreamAdapter.
Bug: webrtc:11943
Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40695}
Upstream commit: https://webrtc.googlesource.com/src/+/7cc1ca26c8350b27ffc7252387b60e80b3e1b56f
Improve ssrc-group validation
disallowing more than one ssrc-group with the same semantic
and primary ssrc.
BUG=chromium:1477075
Change-Id: I4bce0555cd49834725d9b97693d26c971bc5d5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318822
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40694}
Upstream commit: https://webrtc.googlesource.com/src/+/f2827c4b1a7eb574fd83f208001a75b880205c01
FrameCadenceAdapter: schedule repeats before issuing decodes.
The code currently issues frames for encode before scheduling
a new repeat. Swap this order to account for time taken by for
slow encodes.
Bug: webrtc:15456
Change-Id: I74177069e30c1bf65268231ffba033411a0f7b9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40690}
Upstream commit: https://webrtc.googlesource.com/src/+/4c556219e59f2ff3ea62da44e1ff81c6cd1f8021
Cleanup RTPSenderAudio::SendAudio
Combine all parameters into single struct so that it is easier to add and remove optional parameters
Use Timestamp type instad of plain int to represent capture time
Use rtc::ArrayView instead of pointer+size to represent payload
Merge passing audio level into send function.
Bug: webrtc:13757, webrtc:14870
Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40688}
Upstream commit: https://webrtc.googlesource.com/src/+/6e586e1ad29039d51c66c1eedce3d25a5dd49b81
[Fuchsia] Remove fuchsia_target_api_level from .gn
We get this automatically from the //build checkout now
Bug: chromium:1432399
Change-Id: I223d7c5448244ed62821207068f979555617da57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318560
Auto-Submit: Chong Gu <chonggu@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40686}
Upstream commit: https://webrtc.googlesource.com/src/+/fd7b27ef67755a02d8f35ac661601b0dc2c9bba9
Validate SIM ssrc-group parameters
similar to what is done for FID and FEC-FR but SIM can have more than
one secondary SSRC.
BUG=chromium:1477075
Change-Id: I4c9b4feaa421f53e424fc17bfc9ee2c185c68fb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318520
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40679}
Upstream commit: https://webrtc.googlesource.com/src/+/3756e29b150a13c89c4028cb7f4d4e82d70e971f
Remove another ctor from BasicPortAllocator
This constructor isn't used in production. Removing it further
made the construction state of the class simpler, allowed for removal
of the separate Init() method and making more members const.
Bug: none
Change-Id: Ibc8516a01ce7e385207251d841d21bb7b72c9d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318281
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40678}
Upstream commit: https://webrtc.googlesource.com/src/+/5866e1a0edf22e2ba80d8c928e7d1ea58b6043f2
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
following the previous change to rename the classes derived from
cricket::RtpParameters
Also rename ChangedRecvParameters to ChangedReceiveParameters.
BUG=webrtc:13931
Change-Id: Ia51dd39905a5cbb98162c3948930e43ccaf3786d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314500
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40677}
Upstream commit: https://webrtc.googlesource.com/src/+/8fa8619d7e1d81564c476a276588ee6307141c27
FrameCadenceAdapter: account for encode sequence contention.
The synthetic delay added in ZeroHzAdapterMode::OnFrame does not
account for delay with respect to the initial frame post from
FrameCadenceAdapter::OnFrame. Fix this to account for time spent
in contention on the encode sequence.
Bug: webrtc:15456
Change-Id: I63446e8dfe8f62b09d972434a705e912f8a73d69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318420
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40675}
Upstream commit: https://webrtc.googlesource.com/src/+/fe57cecdfe110254c9418c70f040172b966ee707
Remove BasicPortAllocator ctor that's not used in production
The additional constructor unnecessarily increases the complexity
of the class and other downstream classes.
Bug: none
Change-Id: Ied797feb3c982a50b7b47e65018cfc90ca90bf6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40671}
Upstream commit: https://webrtc.googlesource.com/src/+/df3683e9a71e456cd83e3bc112b667a7407749ef
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
since the spec and implementation took a different route
BUG=chromium:1354101
Change-Id: I6beda0db89b9e771ad2a7b51ba739bc46e18a331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40665}
Upstream commit: https://webrtc.googlesource.com/src/+/55b593fb6bf4e13e64c7dec41e07ce24a88bdf3f
Remove EncodedFrame::MissingFrame and start removing Decode() param
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.
Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
Byron Campen