This field trial was enabled by default for a long while.
Bug: webrtc:42234783
Change-Id: If050f88a3649c43d895110f4f68160f020f854e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376421
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43885}
Note that this needs to be done with a work directory that supports
fuzzer builds, otherwise IWYU will bail out with complaints about
find-bad-constructs and raw-ptr-plugin
Some manual work was required to resolve the TaskQueueFactory which
is forward-declared by environment which required a manual include
of the header file.
The DcSctp packet fuzzer was also updated use the
disable_checksum_verification option which was moved to the
DcSctpOptions struct.
vp9_encoder_references_fuzzer was trying to include libvpx includes
which had to be reverted.
BUG=webrtc:42226242
Change-Id: I9fdcf979e73fdee77106c4583faff21ca7abf19f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43873}
To minimize direct construction of BasicPortAllocator, network emulation manager api is changed to push toward injecting network dependencies to PeerConnectionFactory and let it create PortAllocator
Bug: webrtc:42232556
Change-Id: I0c86d797a97d543c2f033286281dc1145d4ef51b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376880
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43872}
That accessor forces test helpers to create BasicPortAllocator themself
rather than deligate such task to PeerConnectionFactory
Bug: webrtc:42232556
Change-Id: I262e032da110222198e6308f57a5e5f2d7ba4601
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376741
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43870}
This way that emulated network may be injected into PeerConnectionFactoryDependencies
and thus would allow test code not to manually create BasicPortAllocator
Bug: webrtc:42232556
Change-Id: Ifac29e584e66d7e585e8c8d74959cba288c6ffe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376500
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43864}
This was generated by
Running
$ for i in test/network/*.cc; do ./tools_webrtc/iwyu/apply-include-cleaner $i; done
$ for i in test/network/*.h; do ./tools_webrtc/iwyu/apply-include-cleaner $i; done
$ python3 ./tools_webrtc/gn_check_autofix.py -C out/Default
manually removing <sys/socket.h> include as suspicious.
manually modifying test/DEPS file.
Bug: webrtc:42226242
Change-Id: Ifda037e1385996ac3b68190c7e30e5309356ebb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43857}
NetworkStateEstimator is not used by WebRTC from receive side.
ReceiveSidesCongestionController::SetTransportOverhead is not needed either since NetworkStateEstimator is removed.
Note, CongestionControlFeedbackGenerator is used with RFC 8888 only and feedback frequency will be refactored in later cl.
Bug: webrtc:42220808
Change-Id: I08980aa19117e1de7a9b7896d05d07715dd9f962
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375460
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43821}
This reverts commit a97304ca03c2aeb4267dc1bd794c50aa8bdb9a69.
Reason for revert: performance tests still rely in on global field trials to configure PC created by this test fixture
Original change's description:
> Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper
>
> Bug: webrtc:42220378
> Change-Id: I3dc1a71c043ef506b6d592673b04e49f4a022d17
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374901
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43803}
Bug: webrtc:42220378, webrtc:392672060
Change-Id: Ide265c1284f9d53c0b652ed5e144dfb0a532f87a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375621
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43812}
Needed for testing in a follow-up CL.
Using ToString rather than absl::StrCat because I want the name of the
enum (e.g. "INVALID_MODIFICATION") as opposed to the enum value (int).
Bug: none
Change-Id: I45a925fad65395d1e6a886a9f787c2f360fb8604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374343
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43777}
Before this CL VP8 and AV1 used the same max QP=56. Tests show that at this QP AV1 delivers a worse PSNR than VP8. We want AV1 min quality to be not worse than VP8. This CL reduces the default max QP for AV1 to 52. With this value libaom AV1 encoder delivers PSNR close to libvpx VP8 at QP 56.
Bug: webrtc:351644568, b/369540380
Change-Id: I2e27ddab562f9c9710b11dc09076b03d7b308bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374041
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43751}
With this CL, the decision if an RTP packet should be sent as ect(1) is made in RtpControllerSend depending on if RFC 8888 has been negotiated and if CCFB is received with ECN enabled.
Since webrtc does not yet adapt to ECN feedback, packets are sent as ECT(1) until the first feedback is received.
Change-Id: Iddf63849328afbe54a7c8f921f2e8db134aeff6a
Bug: webrtc:42225697
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367388
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43609}
reland of
https://webrtc-review.googlesource.com/c/src/+/371661
with an absolute BoringSSL path instead of a relative one
BUG=None
Change-Id: I0f2aef4646b8e7c25ea8e0944889d05baa06bd58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371940
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43598}
- avoid trying to log requested resolution when it is nullopt
- avoid scaling when required resolution happens to be empty. Frame may still arrive in such scenario either because of bugs test tries to catch, or simly because of asynchronous nature of the system under test.
Bug: b/227581196
Change-Id: If1f210c7e372285be38b3f30482827afcb80ede0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43590}
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}
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}
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}
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}
Fix todo to ensure TransportSequence numbers are generated if CCFB according to RFC 8888 is used. Transport sequence numbers are used in BWE algorithms regardless of feedback format.
Bug: webrtc:42225697
Change-Id: I6eab95c0241d590f6e7a90d19c82d13ab8692f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370341
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43515}
The purpose is to be able to add more tests that verify that BWE still work and verify ECN behaviour e2e.
Bug: webrtc:42225697
Change-Id: Ie178d29d7870bfa3211d10925d00c621617ddf48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370561
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43511}
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
This replaces the WaitUntilCondition function that was used in the
peer_connection_encodings_integrationtest previously. Along with that it
adds tests and improved error message printing.
As a drive-by, matchers were added for RTCError as these are the return
type of this utility function.
Bug: webrtc:381524905
Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43494}
Also CHECK in OutputPathWithRandomDirectory. This function is used in tests that need a unique folder to avoid interaction with other tests that may run in parallel. Continuing with a non-unique folder if the creation fails, is likely to cause surprising errors later on.
Bug: webrtc:379973428
Change-Id: I6a30ef9034be8132e2362eff5e46e3b99b30acd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368542
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43431}
This is a reland of commit 775639e930f14a619974944594b40c633cc574a3
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43412}
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
Cleanup some of the TODOs for H.265. They are either invalid or their handling should be merged with other codec types.
Bug: chromium:41480904
Change-Id: I76263354b1b87035e240d77283b21a9a26dcb45b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366044
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43359}
To stress there is no intention to use each instance more than once.
Bug: webrtc:369904700
Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43324}
It works in the same way as the first packet received callback and can be used for latency measurements.
One important detail is that RTCP and probe packets are excluded from triggering the callback.
Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
This would simplify migrating from PeerConnectionFactoryDependencies::audio_processing
for users who use own implementation of the AudioProcessing
Bug: webrtc:369904700
Change-Id: Id05f7280fd01a3e8fd4953f1b24b2467335ab065
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43273}
