Using corpus from another component doesn't seems to work in chromium and blocks webrtc roll into chromium
Bug: None
No-Try: True
Change-Id: I12c460bd1823e929fcdcb6a8feb90e647bb92c39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371661
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43585}
Lists of codecs have a lot of cross references (RTX/APT and the like).
We should introduce functionality to verify that those linkages are correct
before modifying the handling of these.
This CL introduces the CodecList class, which can be extended to do
that verification. It is used by pc/media_session.cc, but inter-module
APIs are not changed in this version (they will be later).
Bug: webrtc:360058654
Change-Id: Ifd6313d0289cfa090e51ac28bc775265d18fe6f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43582}
removing the last vestiges of the p2p/ dependencies and stop depending
on them for the "webrtc" static library.
BUG=webrtc:42226155
Change-Id: I0b6ac36c0a22054c229a94f55fa6690580b9d47f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371342
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43581}
There are many different clocks used for testing. One day there will
only be one but for now this function needs to support them all.
Bug: webrtc:381524905
Change-Id: I8e240167af2ada2494420c751722f8e0dc97f0d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43580}
It is not used so we don't need it.
Bug: webrtc:384483059
Change-Id: I99a4c3dca0881c56d5cd6eb41430505f2c9ccb03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43578}
In particular that avoids lifetime issues with the field trials passed into peerconnection, as now PC takes field trials object by unique_ptr and thus fully manages its lifetime.
Bug: webrtc:42220378
Change-Id: Ia863e9703b5c76ae1866d0ff995b83286c0b947e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43576}
This allows other tests using RTC stats to get pretty printing as well.
Bug: webrtc:381524905
Change-Id: Ib1eb9e1dad36b89e5b1c2ec687fcfeb308f82939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370761
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43575}
This is to fix build error when we set use_libcxx_modules=true in
chromium build.
Bug: chromium:40440396
Change-Id: Iad165a78a6920ccb858567d31fbe5e48d8a7b629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43574}
The negotiation of encrypted header extensions has already been enabled in Chromium, https://chromium-review.googlesource.com/c/chromium/src/+/5933829. Hence, it make sense to enable the encryption of header extensions by default also in webRTC environment so that all the tests run by taking this into considiration when new changes are made.
Bug: webrtc:358039777
Change-Id: I141fac01b0eb0f2ce5a0a365736f0dcf9f21ddcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43573}
When encoder selector is not enabled, currently we always fallback to
VP8 no matter how the codec preference is setup. Update to follow codec
preference order for the fallback.
Bug: chromium:378566918
Change-Id: Ia3fbfc9d407683ef7b3d6246af7e9ec58535dc89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370707
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43566}
It's supported at least on Mac, so gating it on Linux prevents
developers using Mac to run fuzzers easily.
The repository size increase is quite small
Bug: None
Change-Id: I06ce173356f1d7130acd720e70de806bf49f362b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371321
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43560}
This reverts commit 200fd82771ae29d23b2be40194be674b3437f0ab.
Reason for revert: breaks downstream
Original change's description:
> Validate frame consistency when writing DependencyDescriptor
>
> To write DependencyDescriptor frame properties should be consistent with
> the FrameDependencyStructure.
> Historically that was ensured by webrtc codec wrappers, but with with frame transform api interface there are now more ways to inject video frame for packetizing.
> Thus DependencyDescriptorWriter should be more protective to avoid crashes.
>
> Bug: chromium:379282549
> Change-Id: I98f226ff09c32154e18888c8e811e7981567ad45
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371301
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43551}
Bug: chromium:379282549
Change-Id: I7711756f774648cbb85c51b61424bb950c1d3775
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371420
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43556}
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.
Reason for revert: Revised codec matching to fix issue.
Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).
Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}
Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
BuiltinAudioProcessingBuilder should be used instead.
This would allow AudioProcessingImpl to have Environment construction parameter and thus use propagated rather than global field trials.
Bug: webrtc:369904700
Change-Id: I4fcc299bb9e65c109a3fe476c755a81c2aea551c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43553}
To write DependencyDescriptor frame properties should be consistent with
the FrameDependencyStructure.
Historically that was ensured by webrtc codec wrappers, but with with frame transform api interface there are now more ways to inject video frame for packetizing.
Thus DependencyDescriptorWriter should be more protective to avoid crashes.
Bug: chromium:379282549
Change-Id: I98f226ff09c32154e18888c8e811e7981567ad45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371301
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43551}
Previous CLs that disabled the rtc_enable_libevent build flag
did not reveal issues. Now continue to remove the source code for
the task queue.
Bug: webrtc:42224654
Change-Id: I0866b4b56f0a8d8b56a5b604c31a426d77ab8d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43550}
https://webrtc.googlesource.com/src/+/7738bc23ed7fee0d4856bdfe7b88985865829441
switched from using sizeof(uint32_t) to SRTP_SRCTP_INDEX_LEN.
It turned out that this is not always defined.
This patch defines it to 4.
BUG=webrtc:42222036
Change-Id: Ice3d24a6300d19bc2f573469aadd6474ace1b147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43548}
The `FrameToRender` method is deprecated and has been replaced by
`OnFrameToRender`.
Bug: webrtc:358039777
Change-Id: Ibe56bd43cf045d814137ba8c4374bc9b9ce8ef6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371302
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43547}
dtls_transport will when detecting a new fingerprint
(e.g by usage of pranswer) signal DtlsTransportState::kNew.
When this happen, the dtls crypto state is lost, and
sctp should reconnect, srtp does this automatically
in current code base.
The existing behavior in dcsctp is that it will detect
peer sending an init, and reconnect. But any messages sent
between the dtls restart and the message arriving from the
peer will be lost.
This patch changes so that this case is gracefully handled by
a) letting dcsctp_transport listen to dtls state
this is big part of patch and involves changing the type of
the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this
into a separate patch...
b) if a dtls restart happens, delete and restart socket.
Testcase that fails before patch and works after is attached.
Bonus: And include-what-you-use on patch
Bug: b/375327137
Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43546}
This reverts commit 7738bc23ed7fee0d4856bdfe7b88985865829441.
Reason for revert: Some downstream projects are still using an older version of libsrtp
Original change's description:
> srtp: use SRTP_SRCTP_INDEX_LEN define from libsrtp 2.6.0
>
> BUG=webrtc:42222036
>
> Change-Id: Ibf5c6b200501c114b9709b76685bb0ecd30bf9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359627
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43538}
Bug: webrtc:42222036
Change-Id: Icdac768bd4ccb6f1f4ada68637c0b979aefc39f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43544}
This is to fix build error when we set use_libcxx_modules=true in
chromium build.
Bug: chromium:40440396
Change-Id: I5ab1cfcc0d060021892aae0e5ff3f0b647ae4266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370860
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/main@{#43541}
removing the webrtc need for having sources in it.
BUG=webrtc:42226155
Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43540}
