Release notes for WebRTC in Chromium M150

[Harald Alvestrand]

WebRTC Changelog M150 (7827..7871)

Full list of changes Chromium Release Schedule

This release contains 174 commits by 31 authors.

Summary (AI-generated)

• Security: Fixed several vulnerabilities including an OOB write in RotateDesktopFrame (chromium:522134569), and Wayland capture issues (integer overflow in cursor validation chromium:513054275, and buffer geometry validation chromium:509294495).
• Network Thread Migration: Continued efforts to move packet handling to the network thread, including audio/video packet demuxing, Call::OnSentPacket enforcement, and Call::ReceiveStats association.
• Payload Type Redesign: Made significant progress on the payload type allocation redesign, implementing late allocation for video, unifying audio RED linking, and introducing PayloadTypeSuggester.
• SCReAMv2 Congestion Control: Landed numerous improvements to SCReAMv2, including asymmetric step filter for loss rate, reference window backoff inline calculations, and improved handling of reordered packets.
• Cryptex & SFrame: Implemented cryptex header extension negotiation and added SFrame packet buffer support for RTP-level frame assembly.
• API & Modernization: Introduced StrongAlias for RTP header extension IDs and exposed options to prefer SRTP GCM cipher suites in the iOS SDK.

Categories

Category	Changes
API	3
Audio	10
General	115
Infrastructure	3
Peerconnection	6
Security	4
Stats	1
Transport	17
Video	15

Detailed List of Changes (newest first)

Change Description	Category	Links	Bug
[M150] Fix OOB write in RotateDesktopFrame and DXGI size mismatch	Security	📝 🔍	chromium:522134569, chromium:520199394
[M150] [WGC] Fix frame size synchronization	General	📝 🔍	chromium:520652713, chromium:517727318
[SCReAMv2] Inline reference window backoff calculation due to delay in ScreamV2	General	📝 🔍	webrtc:447037083
Ignore internal transport packets in DatagramConnection	Transport	📝 🔍	None
SCReAM v2: Parse TransportPacketsFeedback once into ScreamFeedback	General	📝 🔍	webrtc:447037083
PT: Fix video and audio RED codec handling in TypedCodecVendor and CodecVendor	Audio	📝 🔍	webrtc:360058654
Refactor state caching for SctpDataChannel observers	Peerconnection	📝 🔍	webrtc:510487699
Fix race condition in unsignaled stream creation when worker!=network	Transport	📝 🔍	webrtc:42222117
Add SFrame packet buffer for RTP-level frame assembly	Transport	📝 🔍	webrtc:479862368
Rename LOG_ERROR to RTC_LOG_ERROR	General	📝 🔍	webrtc:517208673
Make video_options_ const in SdpOfferAnswerHandler	General	📝 🔍	None
Fix GetSingleActiveLayerPixels	General	📝 🔍	b/517029078, b/507191231, webrtc:510393737
Fix race condition in CroppingWindowCapturer	General	📝 🔍	chromium:517207235
Make audio_options_ const in SdpOfferAnswerHandler	General	📝 🔍	None
Fix inference of scalability mode	General	📝 🔍	b/517029078, b/507191231, webrtc:510393737
Delete legacy_delay_estimator dead code	General	📝 🔍	None
Clean up the finch experiment kUseHeuristicForFindingEditor	General	📝 🔍	chromium:409473386
Mark ios_force_software_aec_HACK as deprecated	General	📝 🔍	webrtc:42233827
Apply global audio options at the engine level	Audio	📝 🔍	webrtc:42224170
Cut lower than threshold audio at the end (not only at the beginning).	Audio	📝 🔍	None
Remove unnecessary using webrtc:: directives	General	📝 🔍	webrtc:42232595
HeaderExtensionId: Deprecate header extension functions taking int	API	📝 🔍	webrtc:514817938
SCReAMv2: Replace EWMA loss rate filter with asymmetric step filter	General	📝 🔍	webrtc:447037083
Polish deprecated section in the style guide	General	📝 🔍	None
Fix potential buffer underflow in handling of STUN_ATTR_GOOG_MISC_INFO	General	📝 🔍	b/513584780
rtc_event_log_visualizer: Fix crash due to infinite queue delay	General	📝 🔍	webrtc:436707095
Include Sframe library in libwebrtc	Infrastructure	📝 🔍	webrtc:479862368
Remove VirtualSocketServer dependency on FakeClock as unused	General	📝 🔍	webrtc:42223992
Use StrongAlias for RTP header extension identification.	API	📝 🔍	webrtc:514817938
SCReAMv2: Visualize newly lost, recovered, and CE marked packet events	General	📝 🔍	webrtc:436707095
pc: ignore DTLS-decrypted packets in RtpTransport::OnReadPacket	Transport	📝 🔍	webrtc:517079993
Single threaded RtpTransceiver construction	Transport	📝 🔍	webrtc:42224170
MediaEngine: pass parameters-changed callback at construction	General	📝 🔍	webrtc:42222804
PT redesign: handle raw, change allocation and update golden tests	Video	📝 🔍	webrtc:360058654, webrtc:412904801
Bind default sink to existing unsignaled receive stream	Transport	📝 🔍	None
Allow media channel creation from the signaling thread	Peerconnection	📝 🔍	webrtc:42222804
NetEq: Align correlation buffer size in Merge with DspHelper expectations	General	📝 🔍	None
Enable dynamic speed controller by default, with new AV1 defaults.	Video	📝 🔍	webrtc:443906251
Add testing.md to project GEMINI.md	Infrastructure	📝 🔍	None
refactor(pc): unify audio RED linking in payload type redesign	Audio	📝 🔍	webrtc:360058654
Restrict number of actions in VP9 fuzzer	Video	📝 🔍	webrtc:516663678
Iterate over all VP9 GoF `pid_diff`s to determine frame references.	Video	📝 🔍	None
Remove obsolete import of //build/config/chromeos/ui_mode.gni	General	📝 🔍	b:354842935
Implement cryptex header extension negotiation	General	📝 🔍	webrtc:455813732
Update channel constructors to support off-thread creation	Audio	📝 🔍	webrtc:42222804
Validate the length of `num_ref_pics` with `kMaxVp9RefPics` instead of `EncodedFrame::kMaxFrameReferences`.	General	📝 🔍	chromium:454367825
Initialize more RtpSender parameters via the constructor	General	📝 🔍	None
Add fake clock to VideoCodecTextFixtureImpl in non-relatime mode.	General	📝 🔍	webrtc:443906251
Refactor ChannelReceive to accept PacketRouter in constructor	General	📝 🔍	None
Use fake clock for codecs in VideoCodecTester.	General	📝 🔍	webrtc:443906251
Make dependencies optional in VideoFrameMetadata	General	📝 🔍	webrtc:515776877
Remove unused RtpRtcpInterface::SetLocalSsrc method	General	📝 🔍	webrtc:41480926, webrtc:42221687
CryptoOptions: expose srtpPreferGcmCryptoSuites in iOS SDK	General	📝 🔍	webrtc:42221827
Add a "deep parse" mode to Av1QpParser returning the true average QP.	Video	📝 🔍	webrtc:496266459
In ApmTest use SimulatedTimeController to freeze the time	General	📝 🔍	webrtc:42223992
Refactor vp9 low tier logic when dynamic speed is enabled.	Video	📝 🔍	webrtc:443906251
AudioProcessing: Add integration tests for NeuralResidualEchoEstimator	General	📝 🔍	webrtc:442444736
Fix signed integer overflow in video codec tester	Video	📝 🔍	webrtc:14852
AV1 encoder: Avoid holding reference to environment.	Video	📝 🔍	None
Remove deprecated OnDroppedFrame and make OnFrameDropped pure virtual.	General	📝 🔍	webrtc:467444018
Fuzz WebRTC-LibvpxVp9Encoder-PostEncodeFrameDrop field trial in vp9_encoder_references_fuzzer	General	📝 🔍	chromium:515855651, webrtc:500517546
CryptoOptions: add boolean to prefer SRTP GCM cipher suites	General	📝 🔍	webrtc:42221827
Style cleanup Thread and its tests	General	📝 🔍	None
Payload Type Redesign: Progress and fixes	Peerconnection	📝 🔍	webrtc:360058654
Fix potential UAF in StunDictionaryWriter::CreateDelta + heap exhaustion in StunDictionaryView::ApplyDelta	Transport	📝 🔍	None
Add SSRC tracking and receive sink clearing scaffolding	General	📝 🔍	webrtc:42222117
Add better checks for temporal/spatial bounds in EncoderBitrateAdjuster.	Security	📝 🔍	webrtc:514671098
Change WebRtcVideoReceiveChannel decoder_factory annotation	General	📝 🔍	None
Unify worker and network threads in full stack tests	General	📝 🔍	webrtc:514760674
Instruct agent to read agents/README.md on startup	General	📝 🔍	None
Fix RFC 8888 feedback reordering timeline reset	General	📝 🔍	webrtc:515090595
Remove redundant smoothed RTT test	General	📝 🔍	webrtc:447037083
Unrestrict visibility of video_codec_tester	General	📝 🔍	webrtc:14852
logging: Fix RFC 8888 congestion control feedback reordering parsing	General	📝 🔍	webrtc:515090595
Add g3doc file for v2 video encoder api	Video	📝 🔍	webrtc:496266459
dtls-in-stun: Remove the period retransmit	General	📝 🔍	webrtc:367395350
Remove obsolete ReconfigureForTesting from AudioReceiveStreamImpl	General	📝 🔍	None
Migrate unit tests to use CreateTestEnvironment	General	📝 🔍	webrtc:515090586
Update metrics in video codec tester	Video	📝 🔍	webrtc:14852
Move libgav1-based Av1QpParser to modules/video_coding/utility.	General	📝 🔍	webrtc:496266459
Reorder remote stream updates in BaseChannel::UpdateRemoteStreams_w	General	📝 🔍	webrtc:42224170
Explicitly disable DTLS in STUN if not using BoringSSL	Transport	📝 🔍	webrtc:367395350
Record DTLS handshake errors without referencing the PeerConnection object	Transport	📝 🔍	webrtc:514547040
Use injected rather than global clock in test::VideoProcessor	General	📝 🔍	webrtc:42223992
Reject stale BUNDLE MIDs missing from descriptions	General	📝 🔍	webrtc:514442582
Reinitialize mono push resampler	General	📝 🔍	webrtc:514442562
RTP Header Extension: record redefinitions and support error return	Transport	📝 🔍	webrtc:504685269
Return InitEncode errors from simulcast adapter	General	📝 🔍	webrtc:514425860
Redesign: Implement RTX PT convention and enhance video codec support	Video	📝 🔍	webrtc:360058654
Add nullability annotations to Call receive classes	General	📝 🔍	None
Deprecate old QueryCodecSupport interfaces in MediaCapabilities	General	📝 🔍	chromium:505034803
sdp: fix sdp munging detection edge case	General	📝 🔍	chromium:512953564
Payload type allocation: improve stability and test compatibility	General	📝 🔍	webrtc:360058654
Initialize FrameInfo rotation in GenericDecoderTest	General	📝 🔍	webrtc:514423244
Move audio and video packet demuxing to the network thread	Transport	📝 🔍	webrtc:42222117
Handle reordered packets and random loss in ScreamV2	General	📝 🔍	webrtc:447037083
Support packet reordering in CcFeedbackGenerator	General	📝 🔍	webrtc:436707095
Add GetStats to MockAudioDevice.	General	📝 🔍	b/496439120
Add RtpDemuxer::RemoveAllSinks and tests.	General	📝 🔍	webrtc:42222117
Refresh style guide	Infrastructure	📝 🔍	b/509430854
In video_frame.cc, add a null check for the result of VideoFrameBuffer::ToI420() within NativeToJavaVideoFrame to prevent crashes if the conversion fails.	General	📝 🔍	webrtc:503507701
Cap RTCConfiguration certificates and enforce 32-bit overflow checks	Transport	📝 🔍	chromium:513154132, chromium:513154132
Wayland capture: Fix integer overflow in cursor bitmap validation	Security	📝 🔍	chromium:513054275
Validate CGImage dimensions in MouseCursorMonitorMac	General	📝 🔍	chromium:513268100
Fix race conditions in RTCPeerConnectionFactoryTests	Peerconnection	📝 🔍	None
Synchronize capture queue before verifying mock expectations	General	📝 🔍	None
Guard RestoreTokenManager add/reads with Mutex	General	📝 🔍	chromium:513049286
[PT Redesign] Implement late payload type allocation for video.	Audio	📝 🔍	webrtc:360058654
srtp: add UseCryptex API to SrtpSession and SrtpTransport	General	📝 🔍	webrtc:455813732
Improve SrtpSession thread safety and modernize sequence checking	General	📝 🔍	webrtc:361372443
Minor updates to WebRTC Video Engine	Video	📝 🔍	None
Use real rather than simulated task queues in rtp replayer fuzzers	Transport	📝 🔍	chromium:510952673
Simplify WebRTC Voice Engine, remove `friend`.	Audio	📝 🔍	None
Remove RtpPacketSinkInterface inheritance from ReceiveStatisticsImpl	General	📝 🔍	None
Make some VideoReceiveStream2 and RtpVideoStreamReceiver2 members const	General	📝 🔍	webrtc:42222117
Fix UB when comparing two empty webrtc::Buffer objects	General	📝 🔍	webrtc:42224551
Replace UsedPayloadTypes with PayloadTypeSuggester in CodecVendor.	Peerconnection	📝 🔍	webrtc:360058654
Add documentation for testing best practice in WebRTC	General	📝 🔍	None
Add Reported lost time series to ECN feedback graph in event log visualizer	General	📝 🔍	webrtc:436707095
Configure RTCP mode during RTP/RTCP module construction	General	📝 🔍	webrtc:42222117
Implement support machinery for payload type allocation redesign.	Video	📝 🔍	webrtc:42225436
Consolidate remote SSRC representation in audio receive components	Audio	📝 🔍	webrtc:42222117
Add IsEmpty to RtpStreamReceiverController and RtpDemuxer	General	📝 🔍	webrtc:42222117
Move integration test helper functions from .h to .cc	General	📝 🔍	None
Disallow RTP header extension ID of 0	Transport	📝 🔍	chromium:506682780
Rename target_delay to stats_target_delay in VideoDelayTimings.	General	📝 🔍	b/493549134
Delete workaround Thread implementation that do not set self as TaskQueue	General	📝 🔍	webrtc:42221679
Check validity of RTP header extenision ID at construction	Transport	📝 🔍	chromium:506682780
Detect codec collisions between audio and video sections	Audio	📝 🔍	webrtc:42224689
Adds rust version of webrtc::RateTracker	General	📝 🔍	webrtc:416446214
Rely on TaskQueueBase interface in modules/rtp_rtcp	General	📝 🔍	webrtc:42225410
Move MaxWaitingTime and associated state to FrameDecodeTiming.	General	📝 🔍	b/493549134
Update field-trials.md for clarity and freshness	General	📝 🔍	b/499941267
Update Call::ReceiveStats to be associated with the network thread	Video	📝 🔍	webrtc:42222117
sdp: introduce MCD::AttributeLevel for session/media-level attrs	General	📝 🔍	webrtc:455813732
Add a missing include on android	General	📝 🔍	chromium:503250165
Add OnFrameDropped override to vp9 encoder fuzzer.	Video	📝 🔍	webrtc:467444018
Move signaling safety flag into SctpDataChannel and clarify its purpose	Peerconnection	📝 🔍	webrtc:510487699
Prevent wrong scalability mode from being used when base layer inactive.	General	📝 🔍	webrtc:510393737
Use TimeController instead of FakeClock in fuzzers/RtpReplayer	General	📝 🔍	webrtc:42223992
Refactor pc/media_session_unittest.cc and introduce Yoda-test swapping tool.	API	📝 🔍	None
Remove rusty base64 implementation	General	📝 🔍	webrtc:416446214
LNA: Return after unexpected permission callback	General	📝 🔍	chromium:421223919
ScreamV2, add application limited state	General	📝 🔍	webrtc:447037083
snap: add RTCConfiguration for enabling SNAP	General	📝 🔍	webrtc:426480601
In SctpDataChannel use plain bool as safety flag.	General	📝 🔍	chromium:504716948
Remove arm32 bots and redundant arm64 bots	General	📝 🔍	webrtc:427152624
Remove redundant android_compile_arm64_rel bot from CQ	General	📝 🔍	webrtc:427152624
Fix misformated tables in style guide	General	📝 🔍	chromium:510238003
Deprecate ArrayView alias	General	📝 🔍	webrtc:439801349
Add rust versions of Timestamp and TimeDelta	General	📝 🔍	webrtc:416446214
Activate corruption detection tests.	General	📝 🔍	webrtc:358039777
sdp munging: detect modification of msid stream/track	General	📝 🔍	webrtc:414284082
Wayland capture: validate buffer geometry before pixel copy	Security	📝 🔍	chromium:509294495
Remove virtual specifier from RenderTime and MaxWaitingTime in VCMTiming	General	📝 🔍	b/493549134
Migrate android bots from Pixel2 to Pixel7	General	📝 🔍	webrtc:427152624
Cleanup: Iterative removal of matured deprecated symbols (Batch 3)	General	📝 🔍	webrtc:465197113
Fix deps for ssl_header target and add missing frameworks for sdk targets.	General	📝 🔍	webrtc:251890128
pc: move PeerConnectionInterface implementation to the right file	General	📝 🔍	None
Switch android more config from arm32 to arm64	General	📝 🔍	webrtc:427152624
Introduce CodecConfiguration and ResiliencyInfo in pc/	Audio	📝 🔍	webrtc:360058654
Add changelog generator skill	General	📝 🔍	webrtc:465491622
Export GetLoopbackIP to fix WebRTC roll into Chromium	General	📝 🔍	webrtc:251890128
Add GetStats to AudioInput in Android ADM	Stats	📝 🔍	b/384830998
Use Str instead of quotes in DEPS	General	📝 🔍	None
Require HasChannel calls to be called on the signaling thread	Transport	📝 🔍	webrtc:475126742
CHECK on adding STUN attributes after signature is applied.	General	📝 🔍	chromium:504567957
Migrate test_support_unittests to rtc_test_suite	General	📝 🔍	webrtc:498394143
Support std::string_view in RTC_LOG macros	General	📝 🔍	webrtc:42234461
Restrict rtc_test_suite to only allow tests and forward shard_timeout in rtc_cc_test	General	📝 🔍	webrtc:498394143
Use AbslStringify in StringBuilder for custom types	General	📝 🔍	None
Use Pixel 7 phones to run WebRTC Android tests.	General	📝 🔍	webrtc:427152624
Use ip_address helpers to get loopback IP.	General	📝 🔍	webrtc:251890128
Enforce consistent network thread usage in Call::OnSentPacket	Transport	📝 🔍	webrtc:42222117