Release 1.4.0 #12

mininny · 2021-10-01T06:12:51Z

No description provided.

UpdateActiveSimulcastLayers has been blocking WebRtcVideoChannel::SetSend which may be called quite frequently during negotiations. This CL changes UpdateActiveSimulcastLayers to not synchronize with the transport's task queue to wait for the changes to get applied. This synchronization is quite costly, but so too are other remaining things in VideoSendStream, so we should aim to get rid of the `thread_sync_event_` in VideoSendStream. Bug: webrtc:12840, webrtc:12854 Change-Id: Idb48d29b6b8382881c7c1e6f1d0f5e708dbca30f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221203 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34228}

Change log: https://chromium.googlesource.com/chromium/src/+log/b54a8c30e7..44fa1f9723 Full diff: https://chromium.googlesource.com/chromium/src/+/b54a8c30e7..44fa1f9723 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/b006ae39d1..ebd72f5c3a * src/build: https://chromium.googlesource.com/chromium/src/build/+log/bf4ecd926c..5f88cb3c2c * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/5bd5c4f9b4..3321bf9f5d * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/10cb761b19..f39abc2692 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/d030bb4cd0..3261a3665e * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/9705687c25..8891092106 * src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/ae2171b89c..5874d918cf * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/5617cf30e7..a761ddb116 * src/tools/luci-go: git_revision:c9957ed0ce0fd363aac127056344eba1b873bad0..git_revision:2cc9805d5ad186367461ef1c4f0c59098b450418 * src/tools/luci-go: git_revision:c9957ed0ce0fd363aac127056344eba1b873bad0..git_revision:2cc9805d5ad186367461ef1c4f0c59098b450418 * src/tools/luci-go: git_revision:c9957ed0ce0fd363aac127056344eba1b873bad0..git_revision:2cc9805d5ad186367461ef1c4f0c59098b450418 DEPS diff: https://chromium.googlesource.com/chromium/src/+/b54a8c30e7..44fa1f9723/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: Ic4beb4c1f5fdf4d036dae8e0b2fd2bc79d9c2a53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221342 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@{#34229}

AV1X->AV1 mapping added to SdpVideoFormatToVideoCodecInfo in https://webrtc-review.googlesource.com/c/src/+/215586 results in discrepancy of codec name between SDP and VideoCodecInfo. That violates VideoCodecInfo design and breaks downstream projects. This CL moves the mapping from VideoCodecInfoToSdpVideoFormat and SdpVideoFormatToVideoCodecInfo to VideoCodecTypeMime. Bug: b/181690054 Change-Id: I2a76524c29b082241f2ec72a60a209ce9b0c7c5f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221205 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34230}

Bug: None Change-Id: I4c0ab1ba7730bdcdd826aa41b67b80a96d92c8f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221204 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34231}

This is to address a test failure seen on the msan bot(s). Tbr: handellm@webrtc.org Bug: webrtc:12857 Change-Id: I77cbe158e3d0aae62d4a4c0783d5ee6d74edcc22 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221362 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34233}

TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: Iebd999259a109a0776f36b1cd49b4d3debd64260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221400 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@{#34235}

Bug: webrtc:12840 Change-Id: I57e5373ae33060bd3743cea8ada21c845cbbd944 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221365 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34237}

Removes dependence on rtc::Thread APIs from BPAS, which removes the need to remove callbacks via rtc::Thread::Clear(). Bug: webrtc:12840 Change-Id: I0bcc1828c5ab38f521b583f52707174961f28e8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221366 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34239}

Bug: webrtc:12840 Change-Id: Ic25ceb9a487b28575ab7530c54b13781ed404f7a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221367 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34240}

AllocationSequence 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 AllocationSequence to task queues. Bug: webrtc:12840, webrtc:9702 Change-Id: I42bbdb59fb2c88b50e866326ba15134dcc6ce691 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221369 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34241}

TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: Idddcfd5a877bebe57ab9dfa56bddfb9c7acc31a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221583 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@{#34243}

CandidateStats didn't use an initializer list which caused the `candidate` member variable to be constructed with a random id (calling an expensive rng method), only to be overwritten directly thereafter. Bug: webrtc:12840 Change-Id: I0366f674281d236896cb9539812dc2d88c1b37ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221600 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34244}

This is step 1: Encapsulating the data. Bug: webrtc:12837 Change-Id: I15df30dc294c90136a90b072608ed4c2e8925dcb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221602 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34245}

Bug: webrtc:12837 Change-Id: I886ec89427207e1dc291c9959f1b6113c97cbca3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221616 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34246}

Bilinear is faster but lesser quality, box is best quality. Our code base has disagreed about which filter to use for quite some time, causing aliasing bug reports. In an effort to avoid aliasing artifacts and make our scaling filters more predictable, we're updating all uses to kFilterBox. WebRTC already uses kFilterBox everywhere except for these three places. The main discrepency was between Chromium and WebRTC but that has already been fixed. This CL fixes the last remaining bilinears. This brings the WebRTC kFilterBox use count up from 11 to 14 and the kFilterBilinear use count down from 3 to 0. Bug: chromium:1212630 Change-Id: I5fe4aa92b9275d65b91ea97925533055d190d317 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221372 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34248}

We want to turn off PT based demux because SSRC-based endpoints that send media prematurely (which is a popular non-standard behavior still heavily in use) can otherwise get incorrect mappings and unsignalled ssrc issues because of the PT demux path. This CL disables PT based demuxing when the MID header extension is present on all m= sections in the SDP for that kind (audio/video), not caring if it was in the offer or answer. However if PT demuxing has been used in the past then it is always allowed. This ensures PT is off by default but that either offer or answer can enable PT and once it has been on it is also possible to get early media with PT. - Want PT-based demux? The MID header extension has to be removed in either the offer or the answer. Follow-up O/As allow PT demuxing if possible. - Want to use MID or SSRC demuxing? Great, you don't need PT-based demux and won't mind that we turned it off for you. The reason for disabling PT demux at offer time (if MID is present) instead of waiting for the SDP answer is because by the time the SDP answer arrives, early media could have triggered PT demux and caused incorrect mappings. The safe thing is to assume a spec-compliant endpoint until proven otherwise. However if PT demux is ever enabled, then from that point on we always allow PT-based demux in follow-up O/A exchanges. This ensures we don't drop packets in follow-up exchanges. The fact that PT-based demux is disabled during the initial offer should not matter because before the initial O/A exchange we don't have fingerprints. This change only affects Unified Plan and bundled groups. Existing test coverage ensuring we do not break legacy endpoints: [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/peer_connection_integrationtest.cc;l=1156 [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/rtp-demuxing.html;l=59 UnsignaledStreamTest is also updated to test the interesting setups. A kill-switch is added in case we want to disable this change. Bug: webrtc:12814 Change-Id: I807a82a543325753633aaef698e06cb4c9dfebaa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221101 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34251}

Instead, adding specific setters that are needed at runtime: * SetDepacketizerToDecoderFrameTransformer * SetDecoderMap * SetUseTransportCcAndNackHistory The whole config struct is big and much of the state it holds, needs to be considered const. For that reason the Reconfigure() method is too broad of an interface since it overwrites the whole config struct and doesn't actually handle all the potential config changes that might occur when the config changes. Bug: webrtc:11993 Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34252}

This reverts commit e2561e1. Reason for revert: Speculative revert: breaks an downstream project Original change's description: > Remove AudioReceiveStream::Reconfigure() method. > > Instead, adding specific setters that are needed at runtime: > * SetDepacketizerToDecoderFrameTransformer > * SetDecoderMap > * SetUseTransportCcAndNackHistory > > The whole config struct is big and much of the state it holds, needs to > be considered const. For that reason the Reconfigure() method is too > broad of an interface since it overwrites the whole config struct > and doesn't actually handle all the potential config changes that might > occur when the config changes. > > Bug: webrtc:11993 > Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363 > Reviewed-by: Niels Moller <nisse@webrtc.org> > Commit-Queue: Tommi <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34252} TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11993 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34253}

As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}

This reverts commit 8a18e5b. Reason for revert: Removing the problematic DCHECK. Original change's description: > Revert "Remove AudioReceiveStream::Reconfigure() method." > > This reverts commit e2561e1. > > Reason for revert: Speculative revert: breaks an downstream project > > Original change's description: > > Remove AudioReceiveStream::Reconfigure() method. > > > > Instead, adding specific setters that are needed at runtime: > > * SetDepacketizerToDecoderFrameTransformer > > * SetDecoderMap > > * SetUseTransportCcAndNackHistory > > > > The whole config struct is big and much of the state it holds, needs to > > be considered const. For that reason the Reconfigure() method is too > > broad of an interface since it overwrites the whole config struct > > and doesn't actually handle all the potential config changes that might > > occur when the config changes. > > > > Bug: webrtc:11993 > > Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Commit-Queue: Tommi <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#34252} > > TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11993 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34253} # Not skipping CQ checks because this is a reland. Bug: webrtc:11993 Change-Id: I0d3bf9abdcdc8d3f9259d014e6074a5e6b6cc73c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221747 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34255}

This was found when fuzzing. If the specified number of parameter count was larger than std::numeric_limits<size_t>::max()/2, the comparison would overflow and read out-of-bounds. This would only apply to 32-bit platforms and it would lead to a crash as it would access all of the virtual memory range, and more. Fixed: chromium:1216758 Change-Id: I2193d3ed078120b6c3e4645c0b16b9f230055e8d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221742 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34256}

* Removes playing_lock_, sync_info_lock_ and video_sync_lock_. * Also remove video_capture_thread_race_checker_ which was redundant. Only video_sync_lock_ was actually needed. The other two aren't needed anymore because of changes made to RtpStreamsSynchronizer class last year (see webrtc:11489). In the one case where we had a lock, we post a task to the thread where the state is maintained. This task is for capturing histograms which I'm not sure we should have been capturing on the audio thread anyway. Also making ChannelReceiveFrameTransformerDelegate compatible with more tests by using TaskQueueBase instead of rtc::Thread. A number of tests that instantiate ChannelReceive (and thereby CRFTD) set the worker thread as a TQ and not actually an rtc::Thread instance. In those cases CRFTD would previously have gotten a nullptr for the worker thread. Bug: webrtc:11993 Change-Id: I59f4b2afbfedb06f241d9a613f8538adc19cd6d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221364 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34257}

TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: Ia0291826304e065fdd6facc8389c9b1620390b59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225782 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@{#34455}

The previous logic to abandon chunks when partial reliability was used was a bit too eager and trigger happy. * Chunks with limited retransmissions should only be abandoned when a chunk is really considered lost. It should follow the same rules as for retransmitting chunks - that it must be nacked three times or due to a T3-RTX expiration. Before this change, a single SACK not referencing it would be enough to abandon it. This resulted in a lot of unnecessary sent FORWARD-TSN and undelivered messages - especially if running with zero retransmissions. The logic to expire chunks by limited retransmissions will now only be applied when a chunk is actually nacked. * The second partial reliability trigger - expiration time - wasn't evaluated when producing a middle chunk of a larger message. A number of test cases were added and updated as chunks will now be abandoned immediately instead of first scheduled for retransmission and later abandoned. Bug: webrtc:12961 Change-Id: I0ae17b2672568bdbdc32073a99d4c24b09ff5fe9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225548 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34458}

Change log: https://chromium.googlesource.com/chromium/src/+log/2945be3cf4..2bb5b1a1b1 Full diff: https://chromium.googlesource.com/chromium/src/+/2945be3cf4..2bb5b1a1b1 Changed dependencies * src/build: https://chromium.googlesource.com/chromium/src/build/+log/f080e7fe97..9c63d2e3b0 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/97a7aa55f8..5ec64dc385 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/4f683cd3ca..bc1396c8b8 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/8783a04a0d..bc78b5f00d * src/third_party/androidx: gzeKboANx6mGXwrAAyzTn8vZ21URSzhIsHXcKEfxIr0C..jiRYYawcc0irI4AKkkbC0H2fB0mv4vUpWlsqcDKOdMoC * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/59b2e89091..4a3ec1268e * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/10829a05e3..4b3261e487 * src/tools/luci-go: git_revision:2f836b4882d2fa8c7a44c8ac8881c3a17fad6a86..git_revision:01aa19ce019f7bf94712f3dd2538cf72a2a3451b * src/tools/luci-go: git_revision:2f836b4882d2fa8c7a44c8ac8881c3a17fad6a86..git_revision:01aa19ce019f7bf94712f3dd2538cf72a2a3451b * src/tools/luci-go: git_revision:2f836b4882d2fa8c7a44c8ac8881c3a17fad6a86..git_revision:01aa19ce019f7bf94712f3dd2538cf72a2a3451b DEPS diff: https://chromium.googlesource.com/chromium/src/+/2945be3cf4..2bb5b1a1b1/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: Ie5a00b5364a999215ca6c0edfdad294047c486e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225841 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@{#34460}

TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org Bug: None Change-Id: Iba87a93109cd0e715fae5d989a2da5c80325ede1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225862 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@{#34461}

This reverts commit e2ab77b. See bugs, this CL seems to be the culprit of crashes in cricket::TurnPort::OnMessage and jingle_glue::JingleThreadWrapper::Dispatch. TBR=handellm@webrtc.org, hta@webrtc.org Bug: chromium:1227839, chromium:1228462 Change-Id: I7521210970fe543a01682bb08de31ac025e79981 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225880 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34462}

Change log: https://chromium.googlesource.com/chromium/src/+log/2bb5b1a1b1..6d8828f6a6 Full diff: https://chromium.googlesource.com/chromium/src/+/2bb5b1a1b1..6d8828f6a6 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/d51b8338d3..e1acc6a309 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/9c63d2e3b0..8269260083 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/5ec64dc385..695a354117 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/bc1396c8b8..d749d1b98b * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/bc78b5f00d..c1d40d8b39 * src/third_party/androidx: jiRYYawcc0irI4AKkkbC0H2fB0mv4vUpWlsqcDKOdMoC..-umIXLPTAdxRy2iaK4QFSeOf4t7PAKglJP7ggvWhfRwC * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4a3ec1268e..3345f09ed6 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/4b3261e487..1a00526b21 * src/tools/luci-go: git_revision:01aa19ce019f7bf94712f3dd2538cf72a2a3451b..git_revision:2ac8bd9cbc20824bb04a39b0f1b77178ace930b3 * src/tools/luci-go: git_revision:01aa19ce019f7bf94712f3dd2538cf72a2a3451b..git_revision:2ac8bd9cbc20824bb04a39b0f1b77178ace930b3 * src/tools/luci-go: git_revision:01aa19ce019f7bf94712f3dd2538cf72a2a3451b..git_revision:2ac8bd9cbc20824bb04a39b0f1b77178ace930b3 DEPS diff: https://chromium.googlesource.com/chromium/src/+/2bb5b1a1b1..6d8828f6a6/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I6dcb26d533a3cf7d9f1c5492f2849707a2edcdd8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225900 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@{#34463}

When layers are activated/deactivated via UpdateActiveSimulcastLayers, the flag wasn't being updated. This resulted in calls to Stop() getting ignored after an implicit start via activating layers. (cherry picked from commit 35b1cb4) Bug: chromium:1234779 Change-Id: I4a72e624874526d27d3e97d6903112367c5e77fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227700 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34654} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227965 Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/branch-heads/4577@{#1} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

…transport at add-track" This reverts commit 7a2db8a. After this commit, the PeerConnection would assume that any new m= sections will be added to the first existing BUNDLE group. This is true of JSEP endpoints (if they don't do SDP munging), but is not necessarily true for non-JSEP endpoints. This breaks the following scenarios: * Remote offer adding a new m= section that's not part of any BUNDLE group. * Remote offer adding a m= section to the second BUNDLE group. The latter is specifically problematic for any application that wants to bundle all audio streams in one group and all video streams in another group when using Unified Plan SDP, to replicate the behavior of using now-deprecated Plan B without bundling. TBR=hta@webrtc.org Bug: webrtc:12837, webrtc:12906, chromium:1236202 Change-Id: I97a348c96443dee95e2b42792b73ab7b101fd64c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227681 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/branch-heads/4577@{#2} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

Schedule the frames to be decoded based on the pacing delay from the last decode scheduled time. In the current implementation, multiple threads and different functions in same thread can call MaxWaitingTime(), thereby increasing the wait time each time the function is called. Instead of returning the wait time for a future frame based on the number of times the function is called, return the wait time only for the next frame to be decoded. Threads can call the function repeatedly to check the waiting time for next frame and wake up and then go back to waiting if an encoded frame is not available. (cherry picked from commit 82c2248) Change-Id: I00886c1619599f94bde5d5eb87405572e435bd73 Bug: chromium:1237402 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226502 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34660} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228532 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/branch-heads/4577@{#3} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

When pacing is enabled for the low latency rendering path, frames are sent to the decoder in regular intervals. In case of a jitter, these frames intervals could add up to create a large latency. Hence, disable frame pacing if the pre-decode queue grows beyond the threshold. The threshold for when to disable frame pacing is set through a field trial. The default value is high enough so that the behavior is not changed unless the field trial is specified. (cherry picked from commit 2ddc39e) Bug: chromium:1237402 Change-Id: I901fd579f68da286eca3d654118f60d3c55e21ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228241 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34705} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228540 Cr-Commit-Position: refs/branch-heads/4577@{#4} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

Software fallback wrapper now reports least common multiple of requirements for two encoders. SimulcastEncoderAdapter queries actual encoder before InitEncode call and requests alignment for all layers if simulcast is not supported by any of the encoders. (cherry picked from commit c98aebb) Bug: chromium:1084702 Change-Id: Iaed8190737125d447036b6c664b863be72556a5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225881 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34466} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228561 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/branch-heads/4577@{#5} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

…pdating layers. When implicit start/stop happens via activation/deactivation of layers occurs, don't change the state of the 'alive' flag since further activations following full de-activation need to be allowed to happen when Stop() has not been called. (cherry picked from commit 264cf54) Bug: chromium:1234779 Change-Id: Ic3cae387990122eaa2f48de096ff9dafa7c34414 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228242 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34698} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228900 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/branch-heads/4577@{#6} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

As a first step we only want to enable frame pacing for the case where min playout delay == 0 and max playout delay > 0. (cherry picked from commit b2745ba) Bug: chromium:1237402, chromium:1239469 Change-Id: Icf9641db7566083d0279135efa8618e435d881eb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228640 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34752} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229021 Cr-Commit-Position: refs/branch-heads/4577@{#7} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

…manently. After a send stream is stopped, it can still be re-used and implicitly restarted by activating layers. This change removes marking the flag we use for async operations as 'not alive' inside Stop() and only doing so when the send stream is stopped permanently. The effect this has is that an implicit start via UpdateActiveSimulcastLayers() will run and correctly update the states. Before, if a stream had been stopped, the safety flag would prevent the async operation from running. (cherry picked from commit 51238e6) Bug: chromium:1241213 Change-Id: Iebdfabba3e1955aafa364760eebd4f66281bcc60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229304 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34809} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229582 Cr-Commit-Position: refs/branch-heads/4577@{#8} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

…ature/update-to-m93 # Conflicts: # audio/channel_send.cc # call/rtp_video_sender.cc # call/rtp_video_sender_unittest.cc # modules/audio_processing/aec3/transparent_mode.cc # modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc # modules/video_coding/svc/scalability_structure_full_svc.cc # modules/video_coding/svc/scalability_structure_full_svc_unittest.cc # modules/video_coding/svc/scalability_structure_key_svc.cc # modules/video_coding/svc/scalability_structure_key_svc_unittest.cc # sdk/objc/native/api/audio_device_module.mm # sdk/objc/native/src/audio/audio_device_module_ios.mm # tools_webrtc/ios/build_ios_libs.py

There are two problems with setLocalDescription / setRemoteDescription in ObjC SDK. First, RTCSessionDescription.nativeDescription returns a raw nullableSessionDescriptionInterface pointer, where sLD/sRD are calling Clone() method unconditionally, so it might crash. Second, unnecessary sLD/sRD calls Clone() of the raw pointer and does not delete it, so this pointer will leak. To solve these problems, I changed the return type of nativeDescription to std::unique_ptr and removed the call to Clone() method. Bug: webrtc:13022, webrtc:13035 Change-Id: Icbb87dda62d3a11af47ec74621cf64b8a6c05228 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227380 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/master@{#34647}

RTCIceCandidate.nativeCandidate returns a unique_ptr that can be null. As with the previous CL, this is used without checking whether it is null or not, so it should be fixed. Bug: None Change-Id: I70a84f7a2a9a9d47b8cefa198204f9b6d63a7c7f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227620 Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34649} (cherry picked from commit 8c48757)

…-m93 Fix crash on addIceCandidate, setRemote/LocalDescription

Update WebRTC Version to M93

This reverts commit c92fc76. Reason for revert: Gerritt submitted patch set #10 not #12 - this broke Original change's description: > Assign supported codecs per section, not globally. > > This uses the PayloadTypeSuggester for PT disambiguation, > ensuring that conficts are managed across the transport. > > Bug: webrtc:360058654 > Change-Id: Ic8f0569d0d96d97c13765169903929ec8a110104 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383400 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Evan Shrubsole <eshr@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#44284} Bug: webrtc:360058654 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I16d0d92fb9a1ecc65eda060f81d41faefcca1e4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/384000 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44285}

This reverts commit 374b892. Reason for revert: Racing CLs sorted out Original change's description: > Revert "Assign supported codecs per section, not globally." > > This reverts commit c92fc76. > > Reason for revert: Gerritt submitted patch set #10 not #12 - this broke > > Original change's description: > > Assign supported codecs per section, not globally. > > > > This uses the PayloadTypeSuggester for PT disambiguation, > > ensuring that conficts are managed across the transport. > > > > Bug: webrtc:360058654 > > Change-Id: Ic8f0569d0d96d97c13765169903929ec8a110104 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383400 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Evan Shrubsole <eshr@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#44284} > > Bug: webrtc:360058654 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Change-Id: I16d0d92fb9a1ecc65eda060f81d41faefcca1e4c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/384000 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#44285} Bug: webrtc:360058654 Change-Id: I80b59aab396abbc0afa60dab3516f2a22e822a5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383981 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44288}

Tommi and others added 30 commits June 4, 2021 12:32

webrtc-version-updater and others added 23 commits July 12, 2021 05:48

Merge pull request #11 from sendbird/feature/mininny/modifications-to…

3155be2

…-m93 Fix crash on addIceCandidate, setRemote/LocalDescription

Merge pull request #9 from sendbird/feature/update-to-m93

1ac5df9

Update WebRTC Version to M93

Update version and build script

9a3fd03

mininny requested review from intoxicated and x-0o0 October 1, 2021 06:12

intoxicated approved these changes Oct 1, 2021

View reviewed changes

mininny merged commit 94ab479 into master Oct 1, 2021

mininny deleted the release/1.4.0 branch October 1, 2021 07:11

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Release 1.4.0 #12

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