Release notes for WebRTC in Chromium M148

[Harald Alvestrand]

Attached is the AI-generated summary of changes in Chromium M148.

This is a test run for AI-generated summaries - comment if you find them useful!

[Harald Alvestrand]

A format that might be slightly easier to read.

WebRTC Changelog M148 (7727..7778)

Full list of changes Chromium Release Schedule

This release contains 159 commits by 29 authors.

Summary (AI-generated)

• Infrastructure Modernization: Extensive migration from webrtc::ArrayView to std::span across the codebase, including api/, modules/, and common_video/.
• Test Reliability: Significant refactoring of P2P and connectivity tests to use GlobalSimulatedTimeController and RunLoop, improving test deterministic behavior and removing legacy SIMULATED_WAIT.
• Security & Robustness: Addressed several potential out-of-bounds access and buffer overflow issues in audio channel handling, dcSCTP, and RTP packet parsing. Hardened SimpleStringBuilder and added safety checks for CSRC lists and RIDs.
• API Refinement: Transitioned core signaling interfaces (BaseChannel, ChannelInterface) to use RTCError for better error propagation, replacing legacy boolean return values.
• Media & Video Improvements: Added support for GPU textures in desktop capture and improved H.264 decoder compatibility through subset profile matching. Refactored RTP transport to centralize payload type and header extension management.
• Audio Processing: AEC3 now supports dynamic on-the-fly configuration updates for suppressor gain computation.

Categories

Category	Changes
API	5
Audio	13
General	91
Infrastructure	16
Peerconnection	6
Security	4
Stats	5
Transport	12
Video	7

Detailed List of Changes (newest first)

Change Description	Category	Links	Bug
[M148] Always release VideoEncoders before destruction.	General	📝 🔍	chromium:505251595, chromium:504620824
[M148] [Pipewire] Fix mouse cursor data race.	General	📝 🔍	chromium:505447260, chromium:504551032
[M148] Fix TOCTOU race in RtpVideoSender::OnVideoLayersAllocationUpdated	Transport	📝 🔍	chromium:500767595
[M148] Relax RTP header extension ID reuse checks in RtpTransport	Transport	📝 🔍	chromium:504980502, webrtc:503013383
[M148] Check vector sizes when crossfading from CNG/expand to normal.	General	📝 🔍	chromium:504500274, chromium:502661101
[M148] Allow signaled SSRCs to override learned bindings in RtpDemuxer	Peerconnection	📝 🔍	chromium:503666505, webrtc:502130956
[Merge-148] Cherry pick "Move the NullVideoDecoder into a separate file and target."	Video	📝 🔍	chromium:500960863
Fix rebase issue of two colliding CLs	Video	📝 🔍	None
Move payload type demuxing management to RtpTransport	Transport	📝 🔍	webrtc:42222117
Capture channel pointers on signaling thread to fix data race	Transport	📝 🔍	webrtc:481443652
Use RunLoop in audio/ tests	General	📝 🔍	webrtc:469327588
Replace ArrayView with std::span in api/	API	📝 🔍	webrtc:439801349
Allow EventLogAnalyzer to be created without a valid parsed log	General	📝 🔍	None
Use RunLoop in call/ tests	General	📝 🔍	webrtc:469327588
Use RunLoop in p2p/ tests	General	📝 🔍	webrtc:469327588
Use real Thread in examples/ instead of AutoThread	General	📝 🔍	webrtc:469327588
Use RunLoop in media/ tests	General	📝 🔍	webrtc:469327588
Use RunLoop in video/ tests	General	📝 🔍	webrtc:469327588
Use RunLoop in rtc_base/ tests	General	📝 🔍	webrtc:469327588
Group members in VCMTiming into a VideoDelayTimings struct.	General	📝 🔍	b/493549134
Use RunLoop in test/ tests	General	📝 🔍	webrtc:469327588
Use RunLoop in pc/ tests	Peerconnection	📝 🔍	webrtc:469327588
Audio: Enable dynamic on-the-fly AEC3 configuration updates for the suppressor gain computation	Audio	📝 🔍	webrtc:442444736
Delete deprecated reinterpret_array_view	General	📝 🔍	webrtc:439801349
Use ScopedOperationsBatcher for media description pushdown	Audio	📝 🔍	webrtc:42222804, webrtc:42223005
Add gn-check-autofix skill	General	📝 🔍	webrtc:465491622
Replace ArrayView with std::span in modules/video_coding/av1	Video	📝 🔍	webrtc:439801349
Move the LibaomAv1Encoder V2 encoder implementation to separate file.	General	📝 🔍	webrtc:496266459
Scream, ensure feedback contains at least one received packet	General	📝 🔍	webrtc:447037083
Synchronize final stats collection in E2E quality tests	Stats	📝 🔍	webrtc:481443652
Replace ArrayView with std::span everywhere except api	Infrastructure	📝 🔍	webrtc:439801349
PipeWire capture: fix camera capability enumeration race and duplicates	General	📝 🔍	webrtc:478386505
Update ScopedOperationsBatcher tasks with finalizers to propagate errors	General	📝 🔍	webrtc:42222804
Support custom socket server in test::RunLoop	General	📝 🔍	webrtc:469327588
Use the PayloadType class rather than uint8_t in Codec class	General	📝 🔍	webrtc:360058654
Replace ArrayView with std::span in modules/	Infrastructure	📝 🔍	webrtc:439801349
Audio: Reduce max channels to 16 to prevent buffer overflow	Audio	📝 🔍	webrtc:495018167
Fix potential dcSCTP NackBetweenAckBlocks OOB access	Security	📝 🔍	webrtc:495700476
p2ptc: clean up unused legacy API	General	📝 🔍	None
Convert kMax constants in video_codec_constants to size_t	General	📝 🔍	None
Remove SIMULATED_WAIT	General	📝 🔍	webrtc:381524905
More spring cleaning	Infrastructure	📝 🔍	None
Rename methods in ScopedOperationsBatcher	General	📝 🔍	webrtc:42222804
Create include-cleaner skill	General	📝 🔍	webrtc:465491622
Add .gemini to .gitignore	General	📝 🔍	webrtc:465491622
Refactor RTP Header Extension Management into RtpTransport	Transport	📝 🔍	webrtc:42222117
Replace ArrayView with std::span in common_video/	Infrastructure	📝 🔍	webrtc:439801349
Use first_ssrc() instead of ssrcs[0] in channel.cc	General	📝 🔍	None
Use GlobalSimulatedTimeController in PortTest	General	📝 🔍	webrtc:381524905, webrtc:42223992, webrtc:469327588
Replace ArrayView with std::span in modules/audio_processing	Audio	📝 🔍	webrtc:439801349
Spring cleanup of WATCHLIST and OWNER files	Infrastructure	📝 🔍	None
Delete the AEC3 json config fuzzer	Audio	📝 🔍	chromium:440374794, chromium:441805811, chromium:441805815
Truncate CSRC list in RtpPacket::SetCsrcs if too large	Transport	📝 🔍	chromium:486317116
Change how TimingFrameInfo is passed to ReceiveStatisticsProxy.	General	📝 🔍	b/493549134
build: remove unnecessary targets in video_coding	Video	📝 🔍	webrtc:467294026
build: clean up modules audio build files	Audio	📝 🔍	webrtc:467294026
Initialize NetEqNetworkStatistics fields.	General	📝 🔍	None
Add gtest-parallel skill	General	📝 🔍	webrtc:465491622
Refactor stun_port_unittest to use GlobalSimulatedTimeController	General	📝 🔍	webrtc:381524905, webrtc:42223992, webrtc:469327588
pc: rename data_channel_unittest	General	📝 🔍	None
Correct setting Scream padding end time	General	📝 🔍	webrtc:447037083
Use GlobalSimulatedTimeController in P2PTransportChannelTest	General	📝 🔍	webrtc:469327588, webrtc:42223992, webrtc:381524905
Use GPU texture in desktop capture	General	📝 🔍	chromium:40929600
Use subset profile matching for H.264 decoder support	General	📝 🔍	webrtc:347724928
Harden SimpleStringBuilder safety checks	Security	📝 🔍	chromium:486536241
build: clean up api/BUILD.gn libjingle_peerconnection_api target	API	📝 🔍	webrtc:467294026
Simplify thread yielding by removing support for high-priority tasks	General	📝 🔍	webrtc:42222804
Follow-up auto -> explicit type	General	📝 🔍	None
Refactor AddCertificateReports to prevent crash	Stats	📝 🔍	chromium:486495143
Replace LOG_AND_RETURN_ERROR with "return LOG_ERROR"	General	📝 🔍	None
Use GlobalSimulatedTimeController in BasicPortAllocatorTest	General	📝 🔍	webrtc:469327588, webrtc:381524905, webrtc:42223992
snap: assert behavior when sctp-init changes due to pranswer	General	📝 🔍	webrtc:426480601
Ignore BWE expectations in ScreamTest	General	📝 🔍	webrtc:447037083
Refactor BaseChannel+ChannelInterface to return RTCError instead of bool	Peerconnection	📝 🔍	None
Fix out-of-bounds write in SetupCodec due to excessive RIDs	Security	📝 🔍	chromium:486536241
Move VCMTiming::kDefaultRenderDelay to anonymous namespace in .cc file.	General	📝 🔍	b/493549134
Fix unsafe buffer usage error in apm_data_dumper.h	General	📝 🔍	None
build: remove unnecessary targets in video_coding	Video	📝 🔍	webrtc:467294026
Run simulcast adapter test ConcurrentEncodeAndOnEncodedImage on TSAN.	General	📝 🔍	webrtc:467444018
Sframe API support for sender and receiver	API	📝 🔍	webrtc:479862368
Prevent RTCStatsCollector crash for stopped transceivers	Stats	📝 🔍	b/494115682
Allow unspecified max allocatable bitrate in VideoSendStreamImpl	Audio	📝 🔍	webrtc:494350649
Use GlobalSimulatedTimeController in dtmf_sender_unittest	General	📝 🔍	webrtc:42223992
Remove unused method VCMTiming::UpdateCurrentDelay.	General	📝 🔍	b/493549134
FrameEncodeMetadataWriter: Don't make callbacks while holding lock.	General	📝 🔍	webrtc:467444018
build: clean up modules/congestion_controller targets	Infrastructure	📝 🔍	webrtc:467294026
gn_check_autofix: increase error limit for gn check	API	📝 🔍	webrtc:467294026
Allow initial BWE probing with Scream without configured media	General	📝 🔍	webrtc:447037083
Fix metadata for Portaudio library for mac	General	📝 🔍	chromium:460543401
Remove deprecated StartSctpTransport method	General	📝 🔍	None
Remove unused ScopedFakeClock from packet_router_unittest	General	📝 🔍	webrtc:42223992
Make Evan and Guido peer_connection/webrtc_sdp co-owners.	General	📝 🔍	None
Make Evan co-owner of WebRTC stats directories.	Stats	📝 🔍	None
Attempt to deflake ScreamTest.LinkCapacity2MbpsRtt50msNoEcn	General	📝 🔍	webrtc:483936079
Add yielding support to ScopedOperationsBatcher	General	📝 🔍	webrtc:42222804
Move ScopedOperationsBatcher to its own source files	Peerconnection	📝 🔍	webrtc:42222804
Enhance ScopedOperationsBatcher to support tasks returning tasks.	Peerconnection	📝 🔍	webrtc:42222804
Change UMA logging to only log once for same kind of error	General	📝 🔍	chromium:409473386
Use GlobalSimulatedTimecontroller in turn_port_unittest	General	📝 🔍	webrtc:469327588, webrtc:42223992, webrtc:381524905
IWYU: regenerate compile_commands.json on every run	Infrastructure	📝 🔍	webrtc:485150285
Fix racy pending frame status in SimulcastEncoderAdapter.	General	📝 🔍	webrtc:467444018
Ensure Scream only treat reported lost packets once	Transport	📝 🔍	webrtc:483936079
Update kMaxPacketBufferSize to support 50ms @ 48kHz in RtpFileReader	General	📝 🔍	None
Remove use of ScopedFakeClock in video_encoder_software_fallback_wrapper_unittest	General	📝 🔍	webrtc:469327588, webrtc:42223992
Move ssrcs_ and payload_types_ to the network thread	General	📝 🔍	webrtc:42222117
Removed the configuration options for multi_channel processing	Audio	📝 🔍	chromium:464314991
VAD SplitFilter may overflow integerr.	Audio	📝 🔍	webrtc:490331112
Remove WebRTC-RtcEventLogEncodeDependencyDescriptor field trial	General	📝 🔍	webrtc:42225280
Use injected clock in ssl_stream_adapter_unittest	General	📝 🔍	webrtc:469327588, webrtc:42223992
Use GlobalSimulatedTimeController in local_network_access_port_unittest	General	📝 🔍	webrtc:469327588, webrtc:42223992
Refactor StunRequestTest to use GlobalSimulatedTimeController	General	📝 🔍	webrtc:42223992, webrtc:469327588, webrtc:381524905
Re-adding apm_data_dumper.h to the list of unsafe buffer exceptions.	General	📝 🔍	webrtc:478086887
Fetch sender parameters asynchronously during GetStats	Transport	📝 🔍	webrtc:492108787
Fix missing mediaSourceId in simulcast outbound-rtp stats	Stats	📝 🔍	webrtc:492108787
Fix integer overflow in WebRtc_CreateBuffer	Security	📝 🔍	None
Make max target input level for input controller -12dB	General	📝 🔍	webrtc:457791164
AEC3: Fix render buffer headroom calculation on underruns	Audio	📝 🔍	b:487988676, webrtc:442444736
Reject frames with inconsistent U/V strides.	General	📝 🔍	chromium:492213293
Move payload_type_demuxing_enabled_ to the network thread	General	📝 🔍	webrtc:42222117
Fix flakiness in ScreamTest.SendVideoOnlyReturnLinkWithBurstLoss	General	📝 🔍	webrtc:447037083
Add encoder switch request callback to VideoMediaSendChannel	Transport	📝 🔍	b/478050997
Increase MSS to 1280bytes to increase BWE ramp up speed	General	📝 🔍	webrtc:447037083
build: clean up api:enable_media deps	Infrastructure	📝 🔍	webrtc:467294026
Replace RtpDemuxerCriteria member with discrete variables	General	📝 🔍	webrtc:42222117
PipeWire capture: add CFI suppression for StopCapture()	General	📝 🔍	chromium:491979284
build: gn_check_autofix should retain blank lines, comments and # keep tagged lines	Infrastructure	📝 🔍	webrtc:467294026
Remove unsupported python event log analyzer	General	📝 🔍	None
Run iwyu on current CL if no files are given on command line	Infrastructure	📝 🔍	None
Add workaround for OpenH264 related to U/V strides.	General	📝 🔍	chromium:491655161
Removed the AEC3 namespace	Audio	📝 🔍	webrtc:42233815
Make scream more resilient to delay spikes	General	📝 🔍	webrtc:447037083
Encapsulate and make RTP sequence number available in RtpPacketInfo.	Transport	📝 🔍	b/462514208
pc: Enhance stability integration test	General	📝 🔍	webrtc:397895867
build: clean up api/BUILD.gn	API	📝 🔍	webrtc:467294026
Make the targets leaking through 'libjingle_peerconnection_api' public	General	📝 🔍	webrtc:467294026
Remove some unused calls from Video Engine classes	Video	📝 🔍	None
Move Call::receive_time_calculator_ to network thread	Transport	📝 🔍	webrtc:11993
Replace ArrayView with std::span in modules/audio_processing/aec3	Audio	📝 🔍	webrtc:439801349
Replace ArrayView with std::span in net/dcsctp/	Infrastructure	📝 🔍	webrtc:439801349
Fix payload type allocation issues for Audio RED and MID recycling.	Audio	📝 🔍	webrtc:360058654
Replace ArrayView with std::span in logging/	Infrastructure	📝 🔍	webrtc:439801349
Consolidate FakeAudioStream and FakeVideoStream classes for tests.	General	📝 🔍	webrtc:42233500
Replace ArrayView with std::span in rtc_base/	Infrastructure	📝 🔍	webrtc:439801349
build: change more tests to use rtc_test instead of rtc_exectuable	Infrastructure	📝 🔍	webrtc:467294026
Remove test code related to unused variables	General	📝 🔍	None
Replace ArrayView with std::span in media/	Infrastructure	📝 🔍	webrtc:439801349
Removed obsolete suppression	General	📝 🔍	webrtc:42232319
Check RtcTransport payload isn't too short before checking it's an RTP packet	Transport	📝 🔍	chromium:488803429
PipeWire: call pw_deinit() only when running against PipeWire 3.49+	General	📝 🔍	chromium:490340738
Remove deprecated charset handling in AsyncHttpUrlConnection	General	📝 🔍	None
Clean up unused members in WebRtcVideoEngine classes	Video	📝 🔍	None
Clean up some video-related OWNERS files.	Infrastructure	📝 🔍	None
Emit CCFB parameters per codec, not wildcard	General	📝 🔍	webrtc:489794442
Use asynchronous posting for PeerConnection signaling tasks	Peerconnection	📝 🔍	webrtc:442220720
Wayland screencast: fix data race in CaptureFrame()	General	📝 🔍	webrtc:42223634
Remove unused members from WebRtcVoiceEngine and channels	General	📝 🔍	None
Replace ArrayView with std::span in p2p/	General	📝 🔍	webrtc:439801349
Remove deprecated charset handling in JniHelper	General	📝 🔍	None
Replace EncoderSwitchRequestCallback with absl::AnyInvocable	General	📝 🔍	b/478050997