Release notes for WebRTC in Chromium M151

[Fredrik Solenberg]

WebRTC Changelog M151 (7871..7922)

Full list of changes Chromium Release Schedule

This release contains 171 commits by 25 authors.

Summary (AI-generated)

• API & Architecture Modernization: Implemented initial SFrame encryption/decryption infrastructure and RTP depacketizer. Created dedicated api/video/timing module. Deprecated and removed legacy libjingle_peerconnection and libjingle_peerconnection_api targets. Merged rtc_stats and rtc_stats_api targets. Migrated SSLFingerprint creation away from raw-pointer factories.
• Transport & Congestion Control: Enabled ALR probing for RFC 8888 Congestion Control Feedback. Paced CCFB considering actual wire overhead and bandwidth-limited status. Fixed video recovery freeze on RTP sequence number wrap-around. Made ICE tiebreaker a construction-time argument for Port.
• Audio & Video Engine: Introduced DefaultVideoJitterTiming, RtcpRttCalculator, and RttSimulator to modernize video timing and jitter estimation. Added zero-hertz feature flag for static non-screenshare video. Added native thread checking for audio recording APIs.
• PeerConnection & Threading: Constructed Call on the signaling thread and simplified RtpTransceiver/BaseChannel threading. Fixed potential deadlocks and race conditions during PeerConnection teardown. Associated RTX with preceding codecs in payload type handling.
• Security & Robustness: Fixed OOB write vulnerability in RotateDesktopFrame and DXGI size mismatch. Hardened dcSCTP forward TSN processing and throttled RTCP decryption error logging.

Categories

Category	Changes
API	6
Audio	5
General	113
Infrastructure	8
Peerconnection	9
Security	2
Stats	2
Transport	16
Video	10

Detailed List of Changes (newest first)

Change Description	Category	Links	Bug
Forward fix google-internal golden test	General	📝 🔍	webrtc:400630582
Remove payload type 102 reservation	Audio	📝 🔍	webrtc:400630582
Refactor media constants to inline constexpr absl::string_view	General	📝 🔍	webrtc:42223790
Remove special handling of -1 in bandwidth modifiers	General	📝 🔍	webrtc:502249086
Generate keyframes only for requested SSRCs	General	📝 🔍	webrtc:527515375, b/527798266
Update how complete frames are reported to DefaultVideoJitterTiming.	General	📝 🔍	b/493549134
Add libprotobuf-mutator to DEPS	Infrastructure	📝 🔍	None
Add a xvfb_capture_tests rtc_test_suite for the tests that use xvfb.	General	📝 🔍	webrtc:498394143
Update network route on SRTP activation	Transport	📝 🔍	webrtc:461532446
Remove deprecated methods from StunMessage	General	📝 🔍	webrtc:42224821
Add todo guide for outlawing SDP munging	General	📝 🔍	webrtc:414284082
video_timing_simulator: Replace home-made `VideoTimingFactory` with official `VideoJitterTimingFactory`	General	📝 🔍	b/423646186
SFrameDescriptor should follow google style guide	General	📝 🔍	webrtc:479862368
Make VideoJitterTimingFactory injectable through PeerConnectionFactoryDependencies.	General	📝 🔍	b/493549134
agents: add api-compat-reviewer skill	General	📝 🔍	webrtc:465491622
Fix data channel ID reuse race condition	General	📝 🔍	chromium:480916451
Reduce max time between repeat frame to avoid flakiness.	Video	📝 🔍	b/519058486
Relax feedback NTP timestamp reordering threshold to 5 seconds	Transport	📝 🔍	webrtc:515090595
Prevent potential nullptr dereference if metrics are not enabled.	General	📝 🔍	None
Add support for external frame drop signal in EncoderBitrateAdjuster.	General	📝 🔍	webrtc:42225768
Make sure rid has a matching ssrc when requesting keyframe.	General	📝 🔍	b/510386551
Split desktop_capture_unittests rtc_library.	General	📝 🔍	webrtc:498394143
Adding thread checker for audio record on native APIs	Audio	📝 🔍	webrtc:524395191
Add RtpDepacketizerSframe (SFrame RTP depacketizer)	Transport	📝 🔍	webrtc:479862368
Update VCMTiming to use VideoJitterTimingInterface.	General	📝 🔍	b/493549134
Sframe encryptor implementation	API	📝 🔍	webrtc:479862368
Move the below build targets to api/video:	API	📝 🔍	webrtc:42225070
Fix test window leaks in FullScreenWinApplicationHandlerTest	General	📝 🔍	chromium:409473386
snap: fix race condition	Transport	📝 🔍	webrtc:526609852
NetEq Smart Flushing	General	📝 🔍	webrtc:12201
Gracefully handle exceptions during Android AudioTrack and AudioRecord initialization	General	📝 🔍	webrtc:524395191
[infra] Add WebRTC CI builders to Gardener rotation	General	📝 🔍	None
Create api/video/timing directory.	API	📝 🔍	b/493549134
Android: Expose VideoCapturer.isCapturing() and add tests	General	📝 🔍	webrtc:512514764
Fix video recovery freeze on RTP seq number wrap	Transport	📝 🔍	webrtc:516639936
Negotiate receiver reference time (RFC 3611 rtcp-xr:rcvr-rtt) in SDP	General	📝 🔍	webrtc:516205747
Reassign ownership of WebRTC documentation	Infrastructure	📝 🔍	None
Allow SDP munging in peerconnection_unittests	General	📝 🔍	webrtc:414284082
Rename StopDecodeTimer to UpdateDecodeTimeEstimate in VCMTiming.	General	📝 🔍	b/493549134
snap: cache early dtls packets in race condition	Transport	📝 🔍	webrtc:426480601
Remove all untagged Subscribe methods	General	📝 🔍	webrtc:465197113
Fix Multiple SSL clocks issue when running PeerConnectionIntegrationTestWithFakeClock.	General	📝 🔍	None
Validate RTP header extension IDs in ParseExtmap	Transport	📝 🔍	b/522284100
Don't initialize jvm in PeerConnectionFactory if it is already initialized.	General	📝 🔍	None
[SCReAM] Self-contain Application-Limited detection via fixed-window acked rate	General	📝 🔍	webrtc:447037083
Cleanup dead code.	General	📝 🔍	webrtc:42225000
Exclude rate_tracker_rust_tests from ASan builds	General	📝 🔍	chromium:513478482
Remove deprecated symbols from rtc_base and p2p	Transport	📝 🔍	webrtc:465197113, webrtc:42225170, webrtc:42224904
Use accessors to reach into ASN1_STRING	General	📝 🔍	chromium:443769299
Change force-test-environment flag to true	General	📝 🔍	webrtc:521534686
Update RFC8888 feedback_fraction from 0.05 to 0.1 in l4s_test.cc	General	📝 🔍	webrtc:519655843
SetLocalDescription: Avoid redundant offer/answer generation	General	📝 🔍	chromium:40120757
Make RtpHeaderExtensionId(int) explicit	General	📝 🔍	webrtc:514817938
Improve threading annotations in `RenderingTracker`.	General	📝 🔍	b/423646186
Fix double zlep in android/OWNERS	Infrastructure	📝 🔍	None
Centralize shard_timeout=4500 android config in rtc_test definition.	General	📝 🔍	webrtc:524165465
Extract CreateH264ConstrainedBaselineProfile	Video	📝 🔍	webrtc:521921838
Fix extmap URI validation in SDP parsing and sanitise logging	General	📝 🔍	webrtc:509982955
Cleanup unused code in test audio Channel	Audio	📝 🔍	webrtc:42223992
Fix CheckTagAddedToStringInDefaultOnLogMessageAndroid.	General	📝 🔍	webrtc:524165465
Fix webrtc_perf_tests for force-test-env	General	📝 🔍	webrtc:521534686
Remove unused environment_factory dependencies	API	📝 🔍	webrtc:521534686
Consolidate temporal layer limit to kMaxTemporalStreams	Transport	📝 🔍	chromium:456855150
[RFC8888] Enable ALR probing when Congestion Control Feedback is enabled.	Transport	📝 🔍	webrtc:522844909, chromium:523643492
Add Environment-based constructor to DefaultVideoJitterTiming.	General	📝 🔍	b/493549134
Bugfix: Fix declaration order in `ParsedRtcEventLogBuilder`	Infrastructure	📝 🔍	b/423646186
Fix modules_unittests for force-test-env flag	General	📝 🔍	webrtc:521534686
Bugfix: Exclude packet with zero SSRC in GetRtxOsnLoggingStatus	General	📝 🔍	b/423646186
Bugfix: Check RT_X_ header size, not RT_P_ header size.	General	📝 🔍	b/423646186
Run tests on Mac-26 arm64 machines instead of Mac-15.	General	📝 🔍	None
Fix video_tests for force-test-env	General	📝 🔍	webrtc:521534686
Fix many more test binaries for force-test-env flag	General	📝 🔍	webrtc:521534686
Fix audio_engine_tests for force-test-env flag	General	📝 🔍	webrtc:521534686
Restrict max resolution in Vp9FrameBufferPool	General	📝 🔍	b/509981960
Enforce simulcast RID limits and ignore duplicates	General	📝 🔍	b/509982146
Fix modules_tests for force-test-env flag	Video	📝 🔍	webrtc:521534686
Clean up android build targets	Infrastructure	📝 🔍	webrtc:152419689
Adding dct@ to WebRTC owners.	Infrastructure	📝 🔍	None
Add pre-reviewer skill containing code review guidelines+best practices	General	📝 🔍	None
stats: use correct IANA registry for SRTP cipher	Stats	📝 🔍	webrtc:518266748
Use injected clock in FakePacketTransport	General	📝 🔍	webrtc:42223992
Remove overly verbose DLOG in the ZeroHertzAdapterMode.	General	📝 🔍	b/522251236
Avoid global time function in network_tester tool	General	📝 🔍	webrtc:42223992
[WebRTC] Migrate from Legacy Natives to Proxy Natives	General	📝 🔍	chromium:40253109
Fix rtc_p2p_unittests for force-test-env flag	Transport	📝 🔍	webrtc:521534686
Fix rtc_media_unittests for force-test-env flag	General	📝 🔍	webrtc:521534686
Fix peerconnection_unittests for force-test-env	General	📝 🔍	webrtc:521534686
Sframe: rename Encrypter/Decrypter to Encryptor/Decryptor	Infrastructure	📝 🔍	webrtc:479862368
Update deps of videocodec_test_mediacodec	General	📝 🔍	None
Propagate Environment into fake ice transport	Transport	📝 🔍	webrtc:42223992
p2p: remove deprecated SetIceTiebreaker	Transport	📝 🔍	webrtc:42224914
Move jitter estimation logic from VideoStreamBufferController.	General	📝 🔍	b/493549134
Limit SSRCs in ssrc-group to 32	General	📝 🔍	b/509982617
Fix rtc_pc_unittests for force-test-env flag	Peerconnection	📝 🔍	webrtc:521534686
Make videocodec_test_mediacodec a rtc_test	General	📝 🔍	None
fix SVC tests flakiness on Android	General	📝 🔍	None
Add flag to allow enabling the zero hertz feature for non-screenshare content that also contains periods of static video.	Video	📝 🔍	b/522251236
Fix rtc_unittests for force-test-env flag	General	📝 🔍	webrtc:521534686
Integrate `RttSimulator` into video timing simulator	Video	📝 🔍	b/423646186
Add documentation on SCReAM v2 implementation differences	General	📝 🔍	webrtc:447037083
Add jitter estimation logic to `DefaultVideoJitterTiming`.	General	📝 🔍	b/493549134
Bugfix: include `rtx_ssrc` in `all_known_ssrcs_`	General	📝 🔍	b/423646186
Add `RttSimulator`.	General	📝 🔍	b/423646186
Throttle log frequency for RTCP decryption errors.	Transport	📝 🔍	b/367372987
Pace CCFB using actual wire overhead and is_bandwidth_limited status	General	📝 🔍	webrtc:519655843
Delete ThreadProcessingFakeClock as unused	General	📝 🔍	webrtc:42223992
Add force-test-environment flag and checks	General	📝 🔍	webrtc:521534686
Use ScopedTaskSafety in WebRtcSessionDescriptionFactory	Peerconnection	📝 🔍	None
Use CreateTestEnvironment in media tests	General	📝 🔍	None
RtcpRttCalculator bugfix: do NOT reject reports whose dlsr or dlrr ==0.	General	📝 🔍	b/423646186
Fix OOB write in RotateDesktopFrame and DXGI size mismatch	Security	📝 🔍	chromium:520199394
Fix possible uaf in WebRtcSessionDescriptionFactory::Post	General	📝 🔍	b/428836165, b/515407473
Use std::span in ToRtpCapabilities	Audio	📝 🔍	webrtc:514817938
Fix incorrect OwnedFactoryAndThreads teardown sequence	Peerconnection	📝 🔍	b/515407473
Handle possible deadlocks and hangs during PeerConnection teardown	General	📝 🔍	b/515407473
Remove 6 deprecated symbols	General	📝 🔍	webrtc:42224904
Remove the deprecated libjingle_peerconnection target	General	📝 🔍	webrtc:463418937
Integrate DefaultVideoJitterTiming into VCMTiming.	General	📝 🔍	b/493549134
Use new RtpHeaderExtensionCapability constructors	General	📝 🔍	webrtc:514817938
Simplify WaitUntil assertions in tests	General	📝 🔍	None
Don't include private header	General	📝 🔍	None
Add an allowlist for expand_directory	General	📝 🔍	None
Add `RtcpRttCalculator`.	Video	📝 🔍	b/423646186
[dcsctp] Treat forward tsn as payload when saving it for later processing	General	📝 🔍	b/510385580
Initialize TaskQueuePacedSender with configuration at construction	General	📝 🔍	None
Modernize and simplify AudioState	General	📝 🔍	None
Construct Call on the signaling thread	Peerconnection	📝 🔍	webrtc:42222117
Associate RTX with preceding codec in redesign	Peerconnection	📝 🔍	webrtc:360058654
Remove TaskQueueBase::Current() from Call construction	Security	📝 🔍	None
Hdrext: Replace ABSL_DEPRECATE_AND_INLINE with [[deprecated]]	API	📝 🔍	webrtc:514817938
Audio: Apply ABWENoTWCC trial to ReconfigureBitrateObserver	Audio	📝 🔍	None
Migrate from deprecated SSLFingerprint::CreateUnique* factories	General	📝 🔍	None
Simplify RtpTransceiver and BaseChannel construction threading	Peerconnection	📝 🔍	webrtc:42224170
Make ICE tiebreaker a construction time argument of Port	Transport	📝 🔍	webrtc:42224914
Merge ScopedFakeClock and ScopedBaseFakeClock	General	📝 🔍	webrtc:42223992
fix(test): Handle worktrees in repo root detection scripts	General	📝 🔍	None
Remove libjingle_peerconnection_api	General	📝 🔍	webrtc:463418937
Add TargetTransferRate.is_bwe_limited and support in googcc and scream	General	📝 🔍	webrtc:519655843
Add settling delay to CroppingWindowCapturer	General	📝 🔍	chromium:518242046
[WGC] Fix frame size synchronization	General	📝 🔍	chromium:517727318
Migrate test_support_unittests to rtc_test_suite	General	📝 🔍	webrtc:498394143
Delete SSLFingerprint factories returning raw pointer with ownership	General	📝 🔍	None
Move api/stats/OWNERS constraints in the parent folder.	API	📝 🔍	None
Add some sequence checkers to simulator streams.	General	📝 🔍	b/423646186
RtpHeaderExtensionId: Clean up usage in pc/rtp_transceiver_unittest.cc	Peerconnection	📝 🔍	webrtc:42232595
Merge rtc_stats and rtc_stats_api build targets.	Stats	📝 🔍	webrtc:42226139
Add initial DefaultVideoJitterTiming class.	Video	📝 🔍	b/493549134
Delete deprecated test::DirectTransport constructor	General	📝 🔍	webrtc:42223992
Reenable markdown autoformat	General	📝 🔍	chromium:510238003
[visualizer] Add RTCP bitrate time series and packet overhead to total graphs	Transport	📝 🔍	None
PT: Fix test failures and PT mapping conflicts under WebRTC-PayloadTypesInTransport	Peerconnection	📝 🔍	webrtc:360058654
Remove SetOptions and other unused methods from channel interfaces	General	📝 🔍	None
Cleanup non-deprecated test::DirectTransport constructor	General	📝 🔍	webrtc:42225410
Relax BasicPortAllocator to take TaskQueue instead of Thread	General	📝 🔍	webrtc:42225410
Stop relying on the generic libjingle_peerconnection_api dependency.	General	📝 🔍	webrtc:463418937
Configure render delay constant via constructor of VCMTiming.	General	📝 🔍	b/493549134
Store const audio/video options inside RtpTransceiver on construction	Video	📝 🔍	webrtc:42224170
Define spaceship <=> operator for StrongAlias	General	📝 🔍	None
Deprecate implicit conversion from RtpHeaderExtensionId to int	General	📝 🔍	webrtc:514817938
Perform `{Assembled,Rendered}Margin` calculations in ms-level precision.	General	📝 🔍	b/423646186
RtpHeaderExtensionId: Clean up usage in logging, pc, and video	Video	📝 🔍	webrtc:42232595
RtpHeaderExtensionId: Clean up usage in pc/media_session_unittest.cc	Peerconnection	📝 🔍	webrtc:42232595
RtpHeaderExtensionId: Clean up usage in various tests	General	📝 🔍	webrtc:42232595
RtpHeaderExtensionId: Clean up usage in tests and engines	General	📝 🔍	webrtc:42232595
PT: Replicate legacy answer negotiation for directionality	General	📝 🔍	webrtc:360058654
RtpHeaderExtensionId: Clean up usage in remaining tests and build files	Infrastructure	📝 🔍	webrtc:42232595
Make Video Encoder V2 config structs classes with getter/setters.	Video	📝 🔍	webrtc:496266459
cleanup: Remove several deprecated symbols	General	📝 🔍	webrtc:465197113
Fix tests to respect command line field trials	General	📝 🔍	None
Namespace: Fix tests that didn't wrap properly	General	📝 🔍	webrtc:42232595
Deprecate VideoFrameMetadata::GetFrameDependencies	General	📝 🔍	webrtc:515776877