and clean up methods that are now detected as unused.
BUG=None
Change-Id: If5dac3d43d4df6c7c108504c202c2383fe4a3f27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321580
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40823}
Speculative fix. Writing the test for it is more complex.
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-Commit-Position: refs/heads/main@{#40820}
making the UsedId generator the source of truth.
BUG=webrtc:12197
Change-Id: I4318a1366f8b2e20ea5ae264232437a9006c5103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321120
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40802}
which can happen if the remote end reuses a mid.
BUG=webrtc:15471
Change-Id: I38da7dced712400002bc61d616e481a1255aa896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319460
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@{#40776}
It's possible that a peer can signal the same payload with multiple
packetization options. As such, we shouldn't try to fall back to default
packetization until we have considered all the alternatives.
Bug: webrtc:15473
Change-Id: I21772b4d8c53819d1c3105988551ebdbea0df045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320241
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40775}
allowing for a lot of de-templating
BUG=webrtc:15214
Change-Id: Ibe1a5f5d704564566f24c496822a4308ba23c4dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319160
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40774}
which allows exposing more granular errors from CreateOffer/CreateAnswer
BUG=webrtc:15499
Change-Id: If72a84515e220d1e7ca739318bf0b6e8a662f60e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40763}
which is not a generic SDP serializer but only deals with the
simulcast SDP.
BUG=None
Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40761}
If there happens to be an asymmetry between local and remote codecs we
shouldn't validate that there's a 1:1 packetization mapping for every
codec. It's sufficient to check that there's at least one matching
packetization per codec.
Bug: webrtc:15473
Change-Id: Ib4fc8fdd54bb4dccf96f0c802746c848e2deed83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320440
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40760}
and the associated UMA metrics after rollout in M116 stable.
BUG=webrtc:14782
Change-Id: Ib2e0f96e8aa0c1ffbf48aea30f93195aa8b44bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40754}
which is forbidden by
https://tools.ietf.org/html/rfc5761#section-4
BUG=webrtc:12197
Change-Id: I6227f01e7dcbca3f5871a2e4a8cea3c4db0b16cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319120
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40752}
after the local offer stopped the only transceiver
BUG=None
Change-Id: I563207a26b6f0d8f41e5853521f05215b6a0eb09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319520
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40722}
This reverts commit 2d71807fe09aad67efcd660fe286044ff10982ba.
Reason for revert: With the new AsyncAudioProcessing API, the issue that was introduced can now be worked around.
Original change's description:
> Revert "ConnectionContext: remove media engine without blocking."
>
> This reverts commit 2ba941e6bc1d20acb9cfda4b87ba53c80640bbcb.
>
> Reason for revert: Temporarily reverting due to b/269628432.
>
> Original change's description:
> > ConnectionContext: remove media engine without blocking.
> >
> > Bug: webrtc:14449
> > Change-Id: I445114c14f4d440a5a8cac003266047fe4588dab
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288580
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38928}
>
> Bug: webrtc:14449
> Change-Id: If2f23662e486a1c1f85c318fc98c441aab9ace31
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295862
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39454}
Bug: webrtc:14449
Change-Id: I43bb7a3b366eb60b3dc4b88dd9d47d570bb99bc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311941
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40705}
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}
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}
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}
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}
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}
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}
This time, hit the BUILD files too (where possible).
Bug: webrtc:11943
Change-Id: Ic8f2d77e1ba66f740efe0ef73b1ea6051356dedc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40654}
for the known standard semantics FID (used by rtx) and
FEC-FR (used byFlexFEC) they should match the expected two SSRCs.
For the nonstandard SIM group this should be limited by the maximum
number of simulcast layers supported.
BUG=chromium:1459124
Change-Id: I7cc2417a3ab207658ec80e8d7e9984c1ae631f53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315323
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40652}
since their content typically is not processed further.
BUG=webrtc:142258
Change-Id: I5bcfb6c3a6f3a301acb497b83f8a4dbc3023c5db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317603
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40649}
This is mainly to remove them from the list of sigslot blockers.
Bug: webrtc:11943
Change-Id: I7908b953d7b2e3e1f7fd6c4da52412f70f1666c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317901
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40641}
Following https://webrtc-review.googlesource.com/c/src/+/262666, use_rtx() was checked in PeerConnectionFactory::GetRtpReceiverCapabilities but was missed in GetRtpSenderCapabilities. Therefore clients that hardcode use_rtx = false end up in an inconsistent state where RTX is not fully disabled.
Bug: None
Change-Id: Ice5f29a77c59e9081f9dd72c13c819024a34a7dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40625}
Instead, use BlockingCall to match with how unregistration is done.
This is needed because the ThreadWrapper implementation in Chromium, overriding the Thread implementation in WebRTC, does not order sent (blocking) tasks along with posted tasks.
That makes the functional difference that Thread1 posting and sending
tasks to Thread2, can not assume that the tasks run in the order they
were posted and sent. I.e. in this case:
// Running on Thread1.
thread2->PostTask([](){ Foo(); });
thread2->BlockingCall([](){ Bar(); });
Thread2 may actually execute Bar() first, and then Foo().
Bug: chromium:1470992
Change-Id: I1f83f12ce39c09279c0f2b3bc71c3a33e2cb16c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317700
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40624}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
since that transport is unset most of the time when rtcp-mux is used.
BUG=None
Change-Id: Ic1d732369c5544059112173af767488aed7ec8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316926
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40598}
It used input frame resolution before this change which caused unnecessary resolution adaptations when resolution scaling is used.
Found that initial frame dropping was always enabled for AV1 SVC. After fixing DropDueToSize the AV1 SVC tests [1] started to fail ("number of encoded temporal layers is less than expected") on bots. The tests encode 1850x1110 in L3T3 for 5s using the default 300kbps start bitrate. Before the fix the initial frame dropping kicked in and reduced the resolution to a level that let encoder to generate all temporal layers. After the fix the resolution stayed at 1850x1110 and encoder dropped all T1 and T2 layer frames. Mitigated this by increasing test duration from 5 to 10s. This gives enough time for BWE to ramp up and for encoder to generate (stop dropping) all temporal layers.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/test/svc_e2e_tests.cc;l=460;bpv=1
Bug: chromium:1466809
Change-Id: I16802689e234f8fc16f891f024d5f644985de01c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315142
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40536}
as part of the overall motion to remove subtypes of cricket::Codec.
Also update surrounding code to use LOG_AND_RETURN_ERROR.
BUG=webrtc:15214
Change-Id: I7e4a416be662e2e10e351e11d20442ce562d7428
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315080
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40523}
which is present if a fec mechanism like FlexFEC is negotiated
spec change:
https://github.com/w3c/webrtc-stats/pull/765
BUG=webrtc:15250
Change-Id: I7d71d49fab0153d734f22831e6684d2acfc647fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314981
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40514}
remove some of the templating around the Codec-derived types and
use more modern C++ loops.
BUG=webrtc:15214
Change-Id: I2710628741deca647e7ae88f5966ec7c7f12669a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311260
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40475}
