Release notes for WebRTC in Chromium M149

[Harald Alvestrand]

WebRTC Changelog M149 (7778..7827)

Full list of changes Chromium Release Schedule

This release contains 207 commits by 35 authors.

Summary (AI-generated)

• Security: Fixed an Out-of-Bounds (OOB) write in RotateDesktopFrame / DXGI size mismatch (chromium:522134600) and added temporal/spatial bounds checks in EncoderBitrateAdjuster (webrtc:514671098).
• Performance & Threading: Significant reduction of blocking calls on the signaling and network threads by caching ICE credentials, caching DTLS roles, and batching media channel operations using ScopedOperationsBatcher.
• Network & Transport: DTLS 1.3 is now enabled by default. Capped RTCConfiguration certificates and enforced 32-bit overflow checks (chromium:513154132). Added initial support for Network Slices in ICE candidates and statistics.
• API & Modernization: Continued modernization of P2P and STUN code to use std::span. Deprecated AutoThread and AutoSocketServerThread.
• Video: Enabled post-encode frame dropping in libvpx VP9 (webrtc:500517546) and added corruption probability to video receive stream stats.

Categories

Category	Changes
API	3
Audio	6
General	163
Infrastructure	5
Peerconnection	8
Security	2
Transport	12
Video	8

Detailed List of Changes (newest first)

Change Description	Category	Links	Bug
[M149] Fix OOB write in RotateDesktopFrame and DXGI size mismatch	Security	📝 🔍	chromium:522134600, chromium:520199394
[M149] [WGC] Fix frame size synchronization	General	📝 🔍	chromium:520653469, chromium:517727318
[M149] Add better checks for temporal/spatial bounds in EncoderBitrateAdjuster.	Security	📝 🔍	chromium:516658686, webrtc:514671098
[M149] Cap RTCConfiguration certificates and enforce 32-bit overflow checks	Transport	📝 🔍	chromium:513154132, chromium:513154132, chromium:514928807
[M149] In SctpDataChannel use plain bool as safety flag.	General	📝 🔍	chromium:514928086, chromium:504716948
[M149] Guard RestoreTokenManager add/reads with Mutex	General	📝 🔍	chromium:514929759, chromium:513049286
[M149] Validate CGImage dimensions in MouseCursorMonitorMac	General	📝 🔍	chromium:514928856, chromium:513268100
Remove dead method NeedsIceRestart_n from JsepTransportController	General	📝 🔍	None
Add corruption probability to video receive stream stats log.	Video	📝 🔍	webrtc:358039777
Add cursor size validation to Wayland capture	General	📝 🔍	chromium:507743672
Add resolution query support for MediaCapabilities	General	📝 🔍	chromium:505034803
Fix data race on channel_ in RtpTransceiver	General	📝 🔍	webrtc:42224170
[v4l2] Ensure deviceUniqueIdUTF8 is initialized	General	📝 🔍	chromium:506921095
Fix some header includes issues	General	📝 🔍	webrtc:251890128
Move network route change handling to RtpTransceiver	Transport	📝 🔍	webrtc:42224170
Add an ice-option for goog-sped-v1	General	📝 🔍	webrtc:367395350
Update plan for revising PT allocation strategy	General	📝 🔍	webrtc:360058654
Fix ASan null pointer in StartAecDump	General	📝 🔍	webrtc:505372808
Smaller updates to VCMTiming.	General	📝 🔍	b/493549134
add a missing include and a couple missing deps	General	📝 🔍	webrtc:251890128
Remove stringstream fallback from MakeVal in logging.h	General	📝 🔍	webrtc:42234461
[desktop capture] Fix TOCTOU race condition in CroppingWindowCapturer.	General	📝 🔍	chromium:504557432
Add option for asynchronous corruption evaluation.	General	📝 🔍	webrtc:358039777
Always restart Scream on route change	Transport	📝 🔍	webrtc:506000414
[P2P] Modernize STUN and P2P code to use std::span	API	📝 🔍	webrtc:42225170
Add helpers for converting between std::span and	General	📝 🔍	webrtc:42225170
Enable CallbackList constructor subscription and adopt in NetworkManager	General	📝 🔍	webrtc:42222117
Prevent SetParameters from overwriting initial encodings	Peerconnection	📝 🔍	webrtc:500993975
[dcsctp] Ignore nacks for already acked packets	General	📝 🔍	chromium:502356094
Remove input parameters from JitterEstimator::GetJitterEstimate.	General	📝 🔍	b/493549134
Cleanup: Remove matured deprecated symbols with zero external usage	General	📝 🔍	webrtc:41480926
Add missing peerconnection_unittests back	General	📝 🔍	webrtc:498394143
Use injected clock when injecting clock for testing into SSL adapter	General	📝 🔍	webrtc:42223992
Fix double mapoffset application in PipeWire video capture	Video	📝 🔍	chromium:505647674
Fix signal subscription leak in xdg-desktop-portal callbacks	General	📝 🔍	chromium:505371989
Cleanup: Remove 4 matured deprecated symbols with zero external usage	General	📝 🔍	webrtc:9725, webrtc:467444018
Fix uninitialized LogSink::min_severity_ and improve InitializeLogging	General	📝 🔍	None
Add an alias to jni_zero::AdoptRef and CreateLeaky in webrtc	General	📝 🔍	chromium:481689330
snap: fix behavior on subsequent offer/answer	General	📝 🔍	chromium:505771664, webrtc:426480601
Split video_tests out from video_engine_tests	General	📝 🔍	webrtc:498394143
Make FrameInstrumentation movable and prefer move to pass by reference.	General	📝 🔍	webrtc:358039777
Add a compile-time test of subclassing TurnPort	General	📝 🔍	None
Refactor: Move ssrc, media_channel, SetFrameDecryptor, and SetFrameTransformer to RtpReceiverBase	General	📝 🔍	webrtc:479862368
clobber ios bots	General	📝 🔍	None
Split resource_adaptation_tests and video_adaptation_tests	General	📝 🔍	webrtc:498394143
Remove WebRTC-Bwe-NetworkRouteConstraints and WebRTC-Bwe-NoFeedbackReset	General	📝 🔍	webrtc:42221535, webrtc:42234928
Some OnDroppedFrame / OnFrameDropped cleanup.	Video	📝 🔍	webrtc:467444018
Inject Environment instead of just field trials to OpenSSLAdapter	General	📝 🔍	webrtc:42223992
Remove SimpleStringBuilder and move StringBuilder impl to cc	Infrastructure	📝 🔍	None
Remove VideoChannel and VoiceChannel classes	General	📝 🔍	webrtc:42224170
Remove more external auth related code	General	📝 🔍	webrtc:487743907
Split call_tests into its own rtc_test_suite	General	📝 🔍	webrtc:498394143
Consolidate media content setting in BaseChannel	Audio	📝 🔍	webrtc:42224170
Update TaskQueueStdLib to use SystemTime instead of TimeMicros	General	📝 🔍	webrtc:42223992
Update RTCError factory functions and a few usages.	General	📝 🔍	None
Wire Sframe enable callback through sender and receiver constructors.	General	📝 🔍	webrtc:479862368
Update organizational contributions documentation and sitemap	General	📝 🔍	b/499941987
Use injected clock in Fuchsia Capturer	General	📝 🔍	webrtc:42223992
Remove the usage of the field-trial WebRTC-Aec3DeactivateInitialStateResetKillSwitch	General	📝 🔍	webrtc:42233815
Discard unauthenticated TURN responses to unauthenticated requests	General	📝 🔍	chromium:504572664
Prevent transceiver crash during SDP local rejection	General	📝 🔍	chromium:504194151
Move EncodedFrame's receive time to Timestamp	General	📝 🔍	webrtc:42223979
Remove the usage of the field-trial WebRTC-Aec3CoarseFilterResetHangoverKillSwitch	General	📝 🔍	webrtc:42233815
Remove the usage of the field-trial WebRTC-Aec3TransparentModeKillSwitch	General	📝 🔍	webrtc:42233815
Remove the unused RealFFT functionality	General	📝 🔍	webrtc:505072868
Remove obsolete @CalledByNative annotation values	General	📝 🔍	chromium:496331306
AEC3: Refactor SubbandNearendDetector constructor to call SetConfig	Audio	📝 🔍	webrtc:442444736
Remove the usage of the field-trial WebRTC-Aec3AecStateFullResetKillSwitch	General	📝 🔍	webrtc:42233815
Refactor waiting for timeout in PhysicalSocketServer	General	📝 🔍	webrtc:42223992
Change tests to not add attributes after STUN signatures.	General	📝 🔍	chromium:504567957
Use injected clock in Linux Capturers	General	📝 🔍	webrtc:42223992
Split test_support_unittests into smaller targets	General	📝 🔍	webrtc:498394143
Add support for y4m and ivf files in VideoCodecTestFixture.	General	📝 🔍	webrtc:496266459
Fix short-circuit evaluation in RollbackTransports	General	📝 🔍	chromium:504597736
Unify SetReceive and SetSend in media channels	Video	📝 🔍	None
Use injected clock in Windows Capturers	General	📝 🔍	webrtc:42223992
Remove WebRTC-Video-Vp9FlexibleMode	General	📝 🔍	chromium:329396373
Introduce SetSsrcTask and batch SetSsrc in ApplyLocalDescription	Peerconnection	📝 🔍	webrtc:42222117
Discard STUN attributes that follow MESSAGE-INTEGRITY	General	📝 🔍	chromium:504567957
Remove audio stream from old sync group on update.	Audio	📝 🔍	chromium:504599749
pc: fix use-after-free in AllocateSctpSids	General	📝 🔍	chromium:503422316
[Pipewire] Fix mouse cursor data race.	General	📝 🔍	chromium:504551032
Send packets as ECT(1) after all route changes	General	📝 🔍	webrtc:447037083
Resolve dcheck due to wrong threading check in VideoCaptureV4L2	General	📝 🔍	webrtc:504490094
Split peerconnection_unittests into smaller targets	General	📝 🔍	webrtc:498394143
Call Release in SimulcastEncoderAdapter destructor if inited.	General	📝 🔍	chromium:504620824
Always release VideoEncoders before destruction.	General	📝 🔍	chromium:504620824
Ensure ECN can be read from posix socket without OPT_SEND_ECN=1	General	📝 🔍	webrtc:504789166
Fix Data Channel encryption downgrade via BUNDLE desync in kPrAnswer.	General	📝 🔍	webrtc:504579798
Refactor RTP header extension ID allocation to be persistent and shared.	Transport	📝 🔍	webrtc:503013383
Batch up blocking calls when creating media channels in RtpTransceiver	Peerconnection	📝 🔍	webrtc:42222117
Deprecate AutoThread and AutoSocketServerThread	Infrastructure	📝 🔍	webrtc:469327588
Use injected clock in DesktopCaptureOptions	General	📝 🔍	webrtc:42223992
Avoid blocking calls during simulcast transceiver creation	General	📝 🔍	webrtc:42222804
Add IvfFrameReader utility class.	Video	📝 🔍	webrtc:496266459
Update valid effort range and scaling factors for LibaomAv1EncoderV2	General	📝 🔍	webrtc:496266459
Use test::RunLoop in sdk/objc/unittests/main.mm	General	📝 🔍	webrtc:469327588
Use test::RunLoop in rtc_base/test_client_unittest.cc	General	📝 🔍	webrtc:469327588
Use TimeController in p2p/dtls/dtls_ice_integration_fixture.h	General	📝 🔍	webrtc:469327588, webrtc:42223992
Use test::RunLoop in pc/rtp_transceiver_unittest.cc and fix shadowing	Peerconnection	📝 🔍	webrtc:469327588
Introduce StopStandardAsync in RtpTransceiver	General	📝 🔍	webrtc:42222804
Asynchronously clear media channels in RtpTransceiver	General	📝 🔍	webrtc:42222804
Relax RTP header extension ID reuse checks in RtpTransport	Transport	📝 🔍	webrtc:503013383
Fix MSVC C2362 errors in audio_device_core_win	General	📝 🔍	None
Use RunLoop in sdk/ tests	General	📝 🔍	webrtc:469327588
Modernize WebRTC integration tests to support simulated time	General	📝 🔍	webrtc:42223992
Fix JNI Zero's Chromium into Google3 roll	General	📝 🔍	chromium:503034881
Clarify 'issue' terminology in skills doc and emphasize use of `-m`	General	📝 🔍	None
Reject bundles with codec collisions.	General	📝 🔍	webrtc:42224689
Fix UAF in AndroidVideoTrackSource::SetState	Peerconnection	📝 🔍	b/403168866
Add hta-reviewer skill	General	📝 🔍	None
Refactor RtpTransceiver channel initialization to use ScopedOperationsBatcher	General	📝 🔍	webrtc:42222804
Check vector sizes when crossfading from CNG/expand to normal.	General	📝 🔍	chromium:502661101
Allow signaled SSRCs to override learned bindings in RtpDemuxer	Peerconnection	📝 🔍	webrtc:502130956
Eliminate blocking calls in NeedsIceRestart	General	📝 🔍	webrtc:42222117
Fix use-after-free in ScreenCast and Camera portal D-Bus callbacks	General	📝 🔍	chromium:491979284, chromium:499587071
Cache ICE credentials and remove signaling thread blocking calls	Transport	📝 🔍	webrtc:42222117
DTLS role caching in JsepTransportController	Transport	📝 🔍	webrtc:42222117
Cleanup audio nack logic.	Audio	📝 🔍	b/462023185
Fix deadlock in TaskQueueGcd::Delete() when called from the queue.	General	📝 🔍	b/492945633
Introduce explicit configuration for logging initialization	General	📝 🔍	webrtc:42234107
Extract SDP bandwidth histogram reporting to a helper function	General	📝 🔍	chromium:501883592
Add 'negative' category to SDP bandwidth UMA histogram.	General	📝 🔍	chromium:501883592
Handle scalabilityMode in default VideoEncoderFactory::QueryCodecSupport	General	📝 🔍	webrtc:496700735
Stop using ScopedFakeClock in wrapping_active_ice_controller_unittest	General	📝 🔍	webrtc:42223992
Fix missing asm/unistd_64.h error in local Android ARM64 builds	Infrastructure	📝 🔍	b/502437863
Extend IP normalization to handle various IPv4-embedding formats.	General	📝 🔍	webrtc:497635018
[AEC3] Remove unused GetConfiguration from NeuralResidualEchoEstimator	Audio	📝 🔍	webrtc:442444736
Increase max # of datachannels to 65535 per spec	General	📝 🔍	webrtc:42228011
Remove blocking extension validity check from BaseChannel.	Peerconnection	📝 🔍	webrtc:42222117, webrtc:360058654
Update how the DTLS role is queried and reported.	Transport	📝 🔍	None
Deprecate EncodedImageCallback::OnDroppedFrame.	Video	📝 🔍	webrtc:467444018
Add network slice to RTCIceCandidateStats.	General	📝 🔍	webrtc:494142581
Move .agents to agents/ and add PRESUBMIT check	General	📝 🔍	webrtc:465491622
Remove network thread blocking call from CreateChannel	Transport	📝 🔍	webrtc:42222804
Add UMA histogram for SDP bandwidth values.	General	📝 🔍	chromium:501883592
Generalize RTCP-FB wildcard generation in SDP serialization.	General	📝 🔍	webrtc:500638469
Update summarize-external-contribs to skip bot commits	General	📝 🔍	None
Implement queryCodecSupport:scalabilityMode: on RTCDefaultVideoEncoderFactory	General	📝 🔍	webrtc:496700735
sdp munging: fix sctp-port uma id collision	General	📝 🔍	webrtc:414284082
Handle IPv4-mapped addresses in IP classification functions.	General	📝 🔍	webrtc:479635018
Make sure SetSsrc clears non-empty encodings and degradation_preference	General	📝 🔍	webrtc:500993975
Add git-cl skill doc with specific instructions for upload.	General	📝 🔍	webrtc:465491622
sdp: harden sctp max-message-size and sctp-port parsing	Transport	📝 🔍	chromium:498185618
Add unit tests and smaller updates for DecodeTimePercentileFilter.	General	📝 🔍	b/493549134
Add AbslStringify to IPAddress	General	📝 🔍	None
Sframe SDP negotiation support.	General	📝 🔍	webrtc:479862368
Apply FieldTrial to NetEq Config within AudioIngress	General	📝 🔍	webrtc:500324590
Remove legacy callback setters from ChannelInterface	General	📝 🔍	webrtc:42222804
Add tools_webrtc/summarize_external_commits.py	General	📝 🔍	None
Enable post-encode frame dropping in libvpx VP9	Video	📝 🔍	webrtc:500517546, b/500062571
Rename SetChannel to SetChannelForTest and inline logic	Transport	📝 🔍	webrtc:42222804
Refactor RtpTransceiver::SetChannel and remove PushNewMediaChannel	General	📝 🔍	webrtc:42222804
Add logging support for task queue names (including Thread) and prefix.	General	📝 🔍	webrtc:500769400
Fix TOCTOU race in RtpVideoSender::OnVideoLayersAllocationUpdated	Transport	📝 🔍	chromium:500767595
Fix the Chromium into WebRTC autoroller	General	📝 🔍	chromium:501094363
Remove unused is_dtx variable in NetEqImpl::InsertPacket	General	📝 🔍	webrtc:501115607
RtpTransceiver: Inject packet callbacks at channel construction time.	General	📝 🔍	webrtc:42222804
Use LibvpxInterface to get PSNR	General	📝 🔍	webrtc:388070060
Delete WebRTC-LibaomAv1Encoder-PostEncodeFrameDrop	General	📝 🔍	webrtc:351644568
Use scoped_refptr for RtpSenderInterface::SetEncoderSelector	General	📝 🔍	webrtc:42224373, b/419314207
Stop using ScopedFakeClock in dtls_transport_unittest	General	📝 🔍	webrtc:42223992
Modernize test style	General	📝 🔍	None
Replace use of SimpleStringBuilder with StringBuilder	Audio	📝 🔍	webrtc:500767594
datachannel: propagate max-message-size changes	General	📝 🔍	chromium:490588131
Add queue_name() to the TaskQueueBase interface	General	📝 🔍	webrtc:500769400
Update estimated_max_decode_time to hold the current estimate.	General	📝 🔍	b/493549134
Stop using ScopedFakeClock in audio_coding tests	General	📝 🔍	webrtc:42223992
Stop using ScopedFakeClock in dtls_srtp_transport_integrationtest	General	📝 🔍	webrtc:42223992
dcsctp: Resend SHUTDOWN ACK if receiving SHUTDOWN	General	📝 🔍	webrtc:500087337
dcsctp: Ignore shutdown API while shutting down	General	📝 🔍	webrtc:500087337
Move the NullVideoDecoder into a separate file and target.	Video	📝 🔍	chromium:500960863
Add a network slice property on Candidate.	General	📝 🔍	webrtc:494142581
Enable git cl format for Markdown files	General	📝 🔍	None
Stop using ScopedFakeClock in screenshare_layers_unittest	General	📝 🔍	webrtc:42223992
build: cleanup api/ deps	API	📝 🔍	webrtc:467294026
Use RunLoop in rtc_tools/ tests	General	📝 🔍	webrtc:469327588
Stop using ScopedFakeClock in rtc_event_log_unittest	General	📝 🔍	webrtc:42223992
Try to fix ('isolate tests') step failures.	General	📝 🔍	webrtc:498394143
Use EncodingOptions to control wildcard usage in SDP.	General	📝 🔍	webrtc:500638469
[webrtc] Remove use_afl GN arg	Infrastructure	📝 🔍	chromium:492241998
Use RunLoop in modules/ tests	General	📝 🔍	webrtc:469327588
Introduce rtc_test_suite and rtc_cc_test	General	📝 🔍	webrtc:498394143
Fix getParams return type.	General	📝 🔍	None
In Scream, ensure start bitrate is set correct if first feedback delayed	General	📝 🔍	webrtc:447037083
Move HarfBuzz from harfbuzz-ng to harfbuzz	General	📝 🔍	None
Remove unused FakeClock from data_channel_integrationtest	General	📝 🔍	webrtc:42223992
Rename two callback based methods to *Task in MediaChannel	General	📝 🔍	None
Refactor PeerConnection transceiver teardown to use batching	General	📝 🔍	webrtc:42222804
Fix network_tester compile issue.	General	📝 🔍	None
Notify on change of a Network's network slice.	General	📝 🔍	webrtc:494142581
[ObjC SDK]: Fix scalability mode not found error.	General	📝 🔍	webrtc:496700735
Allow specifying GPU adapter LUID for WGC capturer	General	📝 🔍	chromium:40929600
build: clean up rtc_base deps	Infrastructure	📝 🔍	webrtc:467294026
Batch resetting of unsignaled receive streams	General	📝 🔍	webrtc:42222804
dcsctp: Resend SHUTDOWN ACK on timer expiration	General	📝 🔍	webrtc:500087337
Stabilize RenegotiateManyVideoTransceiversAndWatchAudioDelay test	General	📝 🔍	webrtc:42225724
[ObjC SDK]: Added scalabilityMode to RTCRtpEncodingParameters	General	📝 🔍	webrtc:496700735
Replace task vectors with ScopedOperationsBatcher	Peerconnection	📝 🔍	webrtc:42222804
Update thread blocking call metrics and batcher logging	General	📝 🔍	webrtc:42222804
Merge logic from RenderTimeInternal into public RenderTime.	General	📝 🔍	b/493549134
Move NetworkSlice enum to network_constants.	API	📝 🔍	webrtc:494142581
Fix inlining of ArrayView	General	📝 🔍	b/499946633, webrtc:439801349
Default enable DTLS1.3	General	📝 🔍	None
Refresh h-cc-pairs style doc	General	📝 🔍	b/496861998
Reword 'how to depend on Abseil' section	General	📝 🔍	b/496863306
Deprecate ArrayView alias	General	📝 🔍	webrtc:439801349