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2021-06-22ModuleRtcRtcpImpl2: remove Module inheritance.upstream-masterMarkus Handell
This change achieves an Idle Wakeup savings of 200 Hz. ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is enabled in RtcpSender, which it never is. Hence the Module inheritance could be removed. The change removes all known dependencies of the module inheritance, and any related mentions of ProcessThread. Fixed: webrtc:11581 Change-Id: I440942f07187fdb9ac18186dab088633969b340e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34358}
2021-06-22Deprecating AbsoluteCaptureTimeReceiverMinyue Li
Bug: chromium:1056230, webrtc:10739 Change-Id: I42b6a6f1c61eaaa468898a09bb7add30f0a419fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223065 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34357}
2021-06-22RTCPSender: remove compatibility ctor & method.Markus Handell
This change removes compatibility APIs in RTCPSender now that downstream consumers updated. Bug: webrtc:11581, webrtc:6458 Change-Id: I82d70f1ab6b522b3884480b0b16cbdff9a1490c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222323 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34356}
2021-06-22TaskQueueStdlib: initialize the thread last.Markus Handell
TaskQueueStdlib initialized it's thread too early which permitted it to access uninitialized attributes. Also remove the |stopped_| event which isn't needed because of the platform thread being joinable. Fixed: webrtc:12876 Change-Id: Ibd27ce915e0e3ac92ebafca535c5a3fd72f9165e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223340 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34355}
2021-06-22Add RecursiveCriticalSection to the don't-use list of primitivesHarald Alvestrand
Bug: None Change-Id: If16da4582e1b4ae498982429d8a8eaeb81402099 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223341 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34354}
2021-06-22Replace use of RecursiveCriticalSection in VirtualSocketServerNiels Möller
Also change listen_queue_ member to use std::unique_ptr to manage ownership. Bug: webrtc:11567 Change-Id: I85171c9cd0253fdbcbce38b1cfebb1adb5bddd9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223063 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34353}
2021-06-22Revert "Fix echo return loss stats and add to RTCAudioSourceStats."Evan Shrubsole
This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c. Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing. Original change's description: > Fix echo return loss stats and add to RTCAudioSourceStats. > > This solves two problems: > * Echo return loss stats weren't being gathered in Chrome, because they > need to be taken from the audio processor attached to the track > rather than the audio send stream. > * The standardized location is in RTCAudioSourceStats, not > RTCMediaStreamTrackStats. For now, will populate the stats in both > locations. > > Bug: webrtc:12770 > Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182 > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34344} TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12770 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#34352}
2021-06-22Minor code cleanup of WebRtcVideoReceiveStream.Tommi
* Remove unnecessary decoder factory pointer. * Set video decoder factory in the ctor of the config class. * Prepare SetRecvParameters for not needing RecreateWebRtcVideoStream. Bug: none Change-Id: I48fbf2920c9fe50f3995ceab5667eb2f70618f25 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223067 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34351}
2021-06-22ModuleRtpRtcpImpl2: remove RTCP send polling.Markus Handell
This change migrates RTCP send polling happening in ModuleRtpRtcpImpl2::Process to task queues. ModuleRtpRtcpImpl2 would previously only cause RTCP sends while being registered with a ProcessThread. This is now relaxed so that RTCP will be sent regardless of ProcessThread registration status, and it seems no tests cared. Now there's only one piece of polling left in Process. Bug: webrtc:11581 Change-Id: Ibdcffefccef7363f2089c34a9c7d694d222445c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222603 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34350}
2021-06-22Update WebRTC code version (2021-06-22T04:05:30).webrtc-version-updater
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: I1c5330d75eab7ea018c302a433879f2926c44a61 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223302 Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34349}
2021-06-21ModuleRtpRtcpImpl2: update test code.Markus Handell
This change prepares for later CLs that partly replaces logic in the module that depends on the Module system for logic that depends on task queues. The change also changes SendTransport::SendRTCP to schedule packet reception with the simulated time controller. This fixes the problem that SendRTCP itself updates the simulated time which makes it hard to understand the tests. Finally, GlobalSimulatedTimeController was updated to support addition of custom SimulatedSequenceRunners like SendTransport. Bug: webrtc:11581 Change-Id: I0aa310ad0a10526479ad8c28affc38a413363ffd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222602 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34348}
2021-06-21Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from ↵Jake Bromberg
before either macOS or the iOS Simulator supported Metal rendering. Bug: webrtc:12638 Change-Id: Iced21bfac40192f609b65f5ea1dc3393ee1695fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222461 Commit-Queue: Jake Bromberg <jakebromberg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34347}
2021-06-21RTCPSender: migrate to Timestamp.Markus Handell
This change migrates RTCPSender to use webrtc::Timestamp, preparing for later improvements regarding bugs.webrtc.org/11581. Fixed: webrtc:12873 Change-Id: I1159701dc373883367d9b2c86823f8fb59904d55 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222324 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34346}
2021-06-21Reland "Port: migrate to TaskQueue."Markus Handell
This reverts commit a4aabb921353125f6d3a2caa2ceb9cda7e971f22. Reason for revert: downstream tests fixed. TBR=hta@webrtc.org Original change's description: > Revert "Port: migrate to TaskQueue." > > This reverts commit 06540166ca97028454adea48cec9bf109b771ddc. > > Reason for revert: breaks downstream test. > > Original change's description: > > Port: migrate to TaskQueue. > > > > Port uses legacy rtc::Thread message handling. In order > > to cancel callbacks it uses rtc::Thread::Clear() which uses locks and > > necessitates looping through all currently queued (unbounded) messages > > in the thread. In particular, these Clear calls are common during > > negotiation and the probability of having a lot of queued messages is > > high due to a long-running network thread function invoked on the > > network thread. > > > > Fix this by migrating Port to task queues. > > > > > > Bug: webrtc:12840, webrtc:9702 > > Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Markus Handell <handellm@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#34338} > > TBR=hta@webrtc.org,handellm@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I014ef9267d224c10595cfa1c12899eabe0093306 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12840, webrtc:9702 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223062 > Reviewed-by: Markus Handell <handellm@webrtc.org> > Commit-Queue: Markus Handell <handellm@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34339} # Not skipping CQ checks because this is a reland. Bug: webrtc:12840, webrtc:9702 Change-Id: I4d2e086b686da8d5272d67293406300a07edef81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223260 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34345}
2021-06-21Fix echo return loss stats and add to RTCAudioSourceStats.Taylor Brandstetter
This solves two problems: * Echo return loss stats weren't being gathered in Chrome, because they need to be taken from the audio processor attached to the track rather than the audio send stream. * The standardized location is in RTCAudioSourceStats, not RTCMediaStreamTrackStats. For now, will populate the stats in both locations. Bug: webrtc:12770 Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182 Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34344}
2021-06-21RTCPSender: migrate to own configuration struct.Markus Handell
The class depends on RtcRtcpInterface::Configuration which adds an unneeded dependency, and inhibits well-manored changes to the constructor interface. Fix this so that RTCPSender uses it's own configuration struct which can be extended in future CLs. Also add a legacy constructor while downstream dependencies are updated. Bug: webrtc:11581 Change-Id: I8d166ab8253b27c08fcbe6aa7c7adde92688b7dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222322 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34343}
2021-06-21Handle null return from ToI420 in encodersEvan Shrubsole
In cases where ToI420 fails it should be able to return null. Bug: webrtc:12877 Change-Id: Ia13859c104d978a29712ae10f8e15acada8406ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222613 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#34342}
2021-06-21Delete legacy RtpHeaderParser wrapperDanil Chapovalov
Bug: None Change-Id: I4deec4fab631488ef2d0706848cbbe4e085825bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221617 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34341}
2021-06-21Update VirtualSocketServer locking to match documentation.Niels Möller
Add GUARDED_BY annotation on members claimed to be protected by the lock, and add missing lock operations. Also mark a few members const. Bug: webrtc:11567, webrtc:2079 Change-Id: I8f12ca7627df0c24e07fa2ae24a387c6a0ed76cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208224 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34340}
2021-06-21Revert "Port: migrate to TaskQueue."Markus Handell
This reverts commit 06540166ca97028454adea48cec9bf109b771ddc. Reason for revert: breaks downstream test. Original change's description: > Port: migrate to TaskQueue. > > Port uses legacy rtc::Thread message handling. In order > to cancel callbacks it uses rtc::Thread::Clear() which uses locks and > necessitates looping through all currently queued (unbounded) messages > in the thread. In particular, these Clear calls are common during > negotiation and the probability of having a lot of queued messages is > high due to a long-running network thread function invoked on the > network thread. > > Fix this by migrating Port to task queues. > > > Bug: webrtc:12840, webrtc:9702 > Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Markus Handell <handellm@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34338} TBR=hta@webrtc.org,handellm@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: I014ef9267d224c10595cfa1c12899eabe0093306 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12840, webrtc:9702 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223062 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34339}
2021-06-21Port: migrate to TaskQueue.Markus Handell
Port uses legacy rtc::Thread message handling. In order to cancel callbacks it uses rtc::Thread::Clear() which uses locks and necessitates looping through all currently queued (unbounded) messages in the thread. In particular, these Clear calls are common during negotiation and the probability of having a lot of queued messages is high due to a long-running network thread function invoked on the network thread. Fix this by migrating Port to task queues. Bug: webrtc:12840, webrtc:9702 Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34338}
2021-06-21Update WebRTC code version (2021-06-21T04:05:45).webrtc-version-updater
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: I326f2fda91fe99cc2c77cb15c3868c075be15454 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223154 Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34337}
2021-06-20openssl_adapter: document SSL_CTX_set_verify_depth behaviourPhilipp Hancke
document the reason for the depth setting in the code. BUG=None Change-Id: Ia761833ff1cc6fb6cc2768d408e26fe87ded57ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222605 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34336}
2021-06-20Update WebRTC code version (2021-06-20T04:03:02).webrtc-version-updater
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: I6678db22a405627a8f681bb816c366eb22981119 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223111 Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34335}
2021-06-19Update WebRTC code version (2021-06-19T04:03:03).webrtc-version-updater
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: I69f95f118f2e656cfd32915bd5aa538033506807 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223140 Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34334}
2021-06-18Fixes off-by-one error in video capture moduleJohannes Kron
Fixed: webrtc:11290 Change-Id: I471b409c27d6ee577a0ed84e3a09d31fbbc16fcd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222609 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34333}
2021-06-18Delete unused class MockDelayableNiels Möller
Unused since cl https://webrtc-review.googlesource.com/c/src/+/218605 Bug: None Change-Id: Iea0641c3791867679d08a317a5a78c0e75436827 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223060 Commit-Queue: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34332}
2021-06-18Add jakobi to modules/audio_coding OWNERSIvo Creusen
Bug: None Change-Id: I299f38126dc1bb419448dcf6f61d3d0323e33885 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223040 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34331}
2021-06-18dcsctp: Add DcSctpSocketFactoryFlorent Castelli
The factory allows us to isolate the implementation from users who only need to depend directly on the public folder now. Bug: webrtc:12614 Change-Id: Ied09cf772ed427eaf17a7b5705f587da57405640 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220939 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34330}
2021-06-18dcsctp: Don't sent more packets before COOKIE ACKVictor Boivie
While in the COOKIE ECHO state, there is a TCB and there might be data in the send buffer, and RFC4960 allows the COOKIE ECHO chunk to bundle additional DATA chunks in the same packet, but there mustn't be more than one such packet sent, and that packet must have a COOKIE ECHO chunk as the first chunk in it. When the COOKIE ACK chunk has been received, the socket is allowed to send multiple packets. Previously, this was state managed by the socket and not the TCB, as the socket is responsible for moving between the different states. And when the COOKIE ECHO chunk was sent, the TCB was instructed to only send a single packet by the socket. However, if there were retransmissions or anything else that could result in calling TransmissionControlBlock::SendBufferedChunks, it would do as instructed and send those, even if the socket was in a state where that wasn't allowed. When the peer was dcSCTP, this didn't cause any issues as dcSCTP tries to be tolerant in what it receives (but strict in what it sends, except for when there are bugs). When the peer was usrsctp, it would send an ABORT for each received packet that didn't have a COOKIE ECHO as the first chunk, and then restart the handshake (sending an INIT). So this resulted in a longer handshake, but the connection would eventually be correctly established and any DATA chunks that resulted in the ABORTs would've been retransmitted. By making the TCB aware of that particular state, and to make it responsible for creating the SCTP packet with the COOKIE ECHO chunk first, and also to only send a single packet when it is in that state, there will not be any way to bypass this limitation. Also, while not explicitly mentioned in the RFC, the retransmission timer will not affect resending any outstanding DATA chunks that were bundled together with the COOKIE ECHO chunk, as then there would be two timers that both would drive resending COOKIE ECHO and DATA chunks. Bug: webrtc:12880 Change-Id: I76f215a03cceab5bafe9f16eb4775f3dc68a6f05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222645 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34329}
2021-06-18Update WebRTC code version (2021-06-18T04:03:27).webrtc-version-updater
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: Icd2faaa16bee1217af5c8e22db125aff1efb5f7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223000 Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34328}
2021-06-17AGC analog clipping predictor: integrate evaluatorAlessio Bazzica
Integrate ClippingPredictorEvaluator in AgcManagerDirect adding the possibility to run the predictor without affecting the analog gain adjustment process. The evaluator is used to compute precision, recall and F1 score. F1 score and the measured clipping prediction intervals are logged as `WebRTC.Audio.Agc.ClippingPredictor.F1Score` and `.PredictionInterval` histograms respectively. Bug: webrtc:12774 Change-Id: I708dcda9321f92d5bd17ec4c36ebce1165ead57f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221921 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34327}
2021-06-17Avoid assembling complicated but unused video rtp header extensionsDanil Chapovalov
Bug: chromium:1219407 Change-Id: I017de10813a1e80f4af0ba55d8d1aa73077dd131 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222615 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34326}
2021-06-17Catch possible `RuntimeException` from `getCameraCharacteristics`Xavier Lepaul
Also changed the logging of exceptions to give more details Bug: webrtc:10804 Change-Id: Ifba6dee3d1c8ba4ecab408ca7715c3b792d9c004 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222641 Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34325}
2021-06-17Refactoring: Move groups-by-mid into Bundle managerHarald Alvestrand
Bug: webrtc:12837 Change-Id: I2431dbbf8cc291b5f3848d81cf290fd3e97ec15d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222614 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34324}
2021-06-17Apply IWYU to jsep_transport_controller/collectionHarald Alvestrand
Bug: None Change-Id: I7b584099fcc26d7d74f2e08a0939a89a9196cd00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222612 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34323}
2021-06-17Mark VideoSendTiming flags as invalid by default.philipel
Bug: none Change-Id: I962df8a55c022193cb3ec036c3cf35f34f9b2412 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222611 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34322}
2021-06-17Reland "Deprecate microsecond timestamps in RTC event log."Björn Terelius
This is a reland of e6ee8fab7eac915b2b6abc9b71b6d33ad086f3d1 Original change's description: > Deprecate microsecond timestamps in RTC event log. > > (Microsecond timestamps are only used in the legacy wire-format, > and the clocks only have microsecond resolution on some platforms.) > > Also convert structs on the parsing side to use a Timestamp instead > of a uint64_t to represent the log time. > > Bug: webrtc:11933 > Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467 > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34097} Bug: webrtc:11933 Change-Id: I295be966ee96b50719ceb4690dad7e7ce958dbac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221361 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34321}
2021-06-17Change YuvConverter.convert to catch GLExceptions and return null.Fabian Bergmark
With https://webrtc-review.googlesource.com/c/src/+/222582, I420 conversion is allowed to fail. Bug: webrtc:12877 Change-Id: Iadae21ad889f084b8027206af4478223d7733d3e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222653 Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Commit-Queue: Fabian Bergmark <fabianbergmark@google.com> Cr-Commit-Position: refs/heads/master@{#34320}
2021-06-17Add timestamp to log message in generic_decoder.ccJohannes Kron
Bug: None Change-Id: Ib558247d887aff880853ef824f8d80d8e7e4feee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222610 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34319}
2021-06-17Remove unnused build configs for M1 builderChristoffer Jansson
Bug: webrtc:12882 Change-Id: I3aa95d2305bee28c3ea5333b641ac1657a87e0ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222643 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Christoffer Jansson <jansson@google.com> Cr-Commit-Position: refs/heads/master@{#34318}
2021-06-17Make WebRtcAudioReceiveStream::stream_ const.Tommi
This builds on a few other CLs that avoid recreating the audio receive streams on config changes and removes redundant config state in WebRtcAudioReceiveStream, constructs the embedded receive stream in the initializer list and keeps it const. Bug: webrtc:11993 Change-Id: Iad28e0170bee6bf1e08713a89af7c81435b4265e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222100 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34317}
2021-06-17Avoid using legacy rtp parser in neteq test::PacketDanil Chapovalov
Bug: None Change-Id: I9184954d9c99f0a34ae335d03843171864071e5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222648 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34316}
2021-06-17In RtcpTransceiver avoid extra PostTask during constructionDanil Chapovalov
it is not required because during construction members can be set on wrong thread, and in some corner cases it may even cause a crash. Bug: none Change-Id: I37d7f2a7772b6ab5e574077d3f53bca2529f9ae1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222651 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34315}
2021-06-17Revert "Avoid video stream allocation on configuration change after timeout."Jakob Ivarsson
This reverts commit 10814873c584df17e560462adcc2b72e488ada91. Reason for revert: Breaks down stream project Original change's description: > Avoid video stream allocation on configuration change after timeout. > > This is to prevent the video stream to get in a state where it is > allocated but there is no activity. > > Bug: b/189842675 > Change-Id: I0793bd4cbf2a4faed92cf811550437ae75742102 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221618 > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34276} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: b/189842675 Change-Id: If930360000f5ca290100920a4f215358b8c3e644 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222652 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34314}
2021-06-17Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap.Christoffer Rodbro
The behavior was changed on https://webrtc-review.googlesource.com/c/src/+/219696. The revert is due to unknown implications for a downstream project. As REMB caps are not used with send-side bandwidth estimation it should be a noop. Bug: webrtc:12306 Change-Id: Idecc49fda007f72512a8fc1e35d62e673b00df3d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222607 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34313}
2021-06-17Update WebRTC code version (2021-06-17T04:05:50).webrtc-version-updater
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: Id614e3f216e5bcd9870ec6bac74fc44264c9e1a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222843 Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34312}
2021-06-17Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293)chromium-webrtc-autoroll
Change log: https://chromium.googlesource.com/chromium/src/+log/6ade74989a..6f7025c98c Full diff: https://chromium.googlesource.com/chromium/src/+/6ade74989a..6f7025c98c Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/b400b802fd..39aab38bd4 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/9ccd2d2400..a6379d4f30 * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/be7dcbc361..466954eda3 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/701dca6ebd..941fd54fff * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/de894a1492..57d2a56d14 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/70bf08ecdb..96bc38d7d5 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/592d5ec077..74ef838a40 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2e51eecb00..680815db18 DEPS diff: https://chromium.googlesource.com/chromium/src/+/6ade74989a..6f7025c98c/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: Ib1e008619c60a83a30f70e19bafcbf9f96630bcd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222842 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34311}
2021-06-16Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176)chromium-webrtc-autoroll
Change log: https://chromium.googlesource.com/chromium/src/+log/19c2bebe7d..6ade74989a Full diff: https://chromium.googlesource.com/chromium/src/+/19c2bebe7d..6ade74989a Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/46146bca12..b400b802fd * src/build: https://chromium.googlesource.com/chromium/src/build/+log/94441aacf6..9ccd2d2400 * src/buildtools/linux64: git_revision:72d5a6e15d868abc8451fe0a3b6596e86a2ffc40..git_revision:d2dce7523036ed7c55fbb8d2f272ab3720d5cf34 * src/buildtools/mac: git_revision:72d5a6e15d868abc8451fe0a3b6596e86a2ffc40..git_revision:d2dce7523036ed7c55fbb8d2f272ab3720d5cf34 * src/buildtools/win: git_revision:72d5a6e15d868abc8451fe0a3b6596e86a2ffc40..git_revision:d2dce7523036ed7c55fbb8d2f272ab3720d5cf34 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b9f393bf29..9e4ba8b69f * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/1c1a1626e9..701dca6ebd * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/3bdcf36fcf..de894a1492 * src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/8b07d9bbd0..d57b60b2a9 * src/third_party/r8: o1uegxayAMktc600LZ1gX5ZzkC_qhU-frNcWJfmBg98C..gXyBDv_fM87KnLcxvF5AGV5lwnm-JXIALYH8zrzdoaMC * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/dd9e6218fa..2e51eecb00 DEPS diff: https://chromium.googlesource.com/chromium/src/+/19c2bebe7d..6ade74989a/DEPS Clang version changed llvmorg-13-init-12576-g643b6407:llvmorg-13-init-12881-g4017d033 Details: https://chromium.googlesource.com/chromium/src/+/19c2bebe7d..6ade74989a/tools/clang/scripts/update.py TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I58a03ad0dcfa7215ea9ef6c3d745d20c48418804 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222860 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#34310}
2021-06-16Reformat pc/g3doc/rtp.mdArtem Titov
Bug: None Change-Id: I9d528ea414b8214b4f7e193b56ad399f0b8f562c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222649 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34309}
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-02 02:24:23 +0000