Upgrade to webrtc m145 by ipavlidakis · Pull Request #81 · GetStream/webrtc

ipavlidakis · 2026-04-28T15:42:08Z

No description provided.

These implementations were missed in https://webrtc-review.googlesource.com/c/src/+/430920 Bug: webrtc:465197113 Change-Id: Idecd88ca73cc1bb8d35c63419e3ee77012477f86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431120 Auto-Submit: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46406}

Bug: None Change-Id: I47ccb42054d07e9f9e14dae378f471c56839da60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431480 Bot-Commit: 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/main@{#46410}

This is the first step to add a phase to build WebRTC with -std=c++23 on "more configs" bots. Once this is landed [1] will enable the new bot phase. [1] - https://chromium-review.googlesource.com/c/chromium/tools/build/+/7232116 Bug: chromium:388070065 Change-Id: I4f2bdfc0a18f7c2fba39a31d5c8a9dbb613f4916 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431580 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46412}

Bug: None Change-Id: If8c364ce9b6100b566d7629932154689e10618ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431760 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#46414}

Bug: None Change-Id: I86ed94299edce88905200d3db8fe137eac6be440 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431981 Bot-Commit: 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/main@{#46415}

With these tests, I think we have enough tests to close that bug. Bug: webrtc:463528061 Change-Id: Ie3f3d98737dca123a39fc123593755336974aa96 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431200 Auto-Submit: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46417}

which allows correlating this logging with pcaps more easily. Bug: b:422493053 Change-Id: I20634534dc6b46ac9cbeaa994c7967a98d50b21b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431381 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46418}

No-try: true Bug: none Change-Id: Ia19923b21668aaae246284a53a8618f29c091bfb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431581 Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46419}

Allow users to provide custom RtpTransport impls eg for platforms which provide external SRTP encryption, following the same pattern as the injectable DtlsTransportFactory. Bug: chromium:443019066 Change-Id: I17dceb2e273808331f2b812de286d02609cf47b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/430722 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#46420}

The `is_pc_closed_` variable and `SetPeerConnectionClosed`, were redundant. PeerConnection stops all transceivers when it closes, checking the `stopped_` flag and calls StopInternal(), which is sufficient to determine if the connection is closed. Bug: none Change-Id: I23757c21c6607a6c650a66b20db0a10c0efeac06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431281 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46421}

The mid is available on the transceivers instead. Using channel properties in the stats code, complicates things a bit due to the transceiver->channel hierarchy and the order of object creation. The mid property is redundant on the channel object and should also be accessed consistently on the network thread. Also updates FakePeerConnectionForStats to ensure transceivers have MIDs set when created. Bug: none Change-Id: I669a049e67a14106f0f0c4f17fa4d0ad86da345e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431380 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46422}

Bug: webrtc:447037083 Change-Id: Ifbb61f8578af186259f84092298507a87d8c3e06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432120 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46423}

Dependency in Chrome has been removed; attempt to remove the .h files not needed downstream from the target. Remove legacy dependencies inside webrtc; they were not needed. Bug: webrtc:463418937 Change-Id: I5c0ca6959186af0cd42fb98874d79a05082b7b07 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431821 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46427}

Bug: None Change-Id: I63d87b6fd0a45df047ad56f00a4a54b02d46205f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432360 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#46429}

This seems to have no purpose at the moment. Bug: None Change-Id: I82cff122af9034e14f4c544a4f9000cdf0872543 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432280 Auto-Submit: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46430}

This change adds utility functions to test::RunLoop for allowing to waiting. Bug: webrtc:381524905 Change-Id: Idb6bfd8f4b39c9908c95809768fd695fe63e2bbf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/430681 Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46432}

I have noticed that some videos have more complicated numerator and denominator than just 30:1 or 25:1. Therefore, increasing the number of bytes needed to read in the header. Bug: webrtc:358039777 Change-Id: I20a840b492697b8606700bac43e3805cc4067081 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/431241 Reviewed-by: Fanny Linderborg <linderborg@webrtc.org> Commit-Queue: Emil Vardar (xWF) <vardar@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46433}

AAudio use CLOCK_MONOTONIC, AAudioWrapper queries time to compare it with time provided by AAudio. Using `clock_gettime` ensures the same clock is used, removes assumption webrtc::TimeNanos() use the same CLOCK_MONOTONIC as aaudio uses. While modifying this function, clarify the time units with a suffix Bug: webrtc:42223992 Change-Id: I23a1525aad7518d4f4a8df379d0726b806665f14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432500 Reviewed-by: Zoé Lepaul <zlep@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46434}

Bug: None Change-Id: Ib665f154c3fbe04ced52d791a006b605107708f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432300 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46435}

Update the PeerConnectionJsepTest suite to utilize separate threads for signaling, worker, and network operations. Previously, all these responsibilities were handled by the current thread, which does not accurately reflect real-world PeerConnection usage. Bug: webrtc:42222804 Change-Id: I0203d67a649faaa18825cc174162d9cc6b7d09cc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441141 Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46631}

Bug: webrtc:447037083 Change-Id: I6c0d481d909fbeb4cff38e6ed56d45cc2c93e364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/440945 Commit-Queue: Per Kjellander <perkj@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Auto-Submit: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46632}

using find audio/ -name BUILD.gn | xargs tools_webrtc/gn_check_autofix.py -r followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and picking the non-controversial changes (all of them) Bug: webrtc:467294026 Change-Id: I4131f6bb2689cac9b2e913b8a8b800dc20c83caa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441023 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46633}

This change ties the capture session to the editor window when a slide show is chosen for capture. This ensures that the present session continues by capturing the editor window in the case when the user has exited the slide show. Bug: chromium:409473386 Change-Id: I8c141f80c193d53acb3a763b4c6d5976034d607b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407423 Commit-Queue: Palak Agarwal <agpalak@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46634}

using find p2p/ -name BUILD.gn | xargs tools_webrtc/gn_check_autofix.py -r followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and picking the non-controversial changes (all but the benchmark dependency on test_main) Bug: webrtc:467294026 Change-Id: I1964d018f379869773ca335068e012d4431cdc03 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441120 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/main@{#46635}

This is the seventh CL in the CL chain for adding the mixing capability for stereo signals. This completes the mixer, but additional CLs will be added for using the class in the AudioBuffer class. Change-Id: I6205a53b4f4b8757ea06b9ac4540046355778885 Bug: webrtc:42221468 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432700 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#46636}

This CL wires up the `CorruptionDetectFrameSelectorSettings` class (using the "WebRTC-CorruptionDetectionFrameSelector" field trial) up to the actual `FrameInstrumentationGenerator` implementation. Bug: webrtc:358039777 Change-Id: I8f16b39270a7f2e3dca70bf52b509cf5eba86e9f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441820 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Auto-Submit: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46638}

This reverts commit 99443ea. Reason for revert: Deferral of ReconfigureEncoder removed to not break downstream tests. Original change's description: > Revert "Reland "Request refresh frame on resolution scaling changes"" > > This reverts commit 5f1aeff. > > Reason for revert: This CL may cause downstream tests to fail. > > Original change's description: > > Reland "Request refresh frame on resolution scaling changes" > > > > This reverts commit df1ffee. > > > > Reason for revert: Cl updated so that WPT test passes. > > > > Original change's description: > > > Revert "Request refresh frame on resolution scaling changes" > > > > > > This reverts commit 0ba5208. > > > > > > Reason for revert: Suspect this caused the following error in chrome: > > > > > > external/wpt/webrtc-encoded-transform/tentative/RTCEncodedFrame-timestamps.html > > > > > > https://ci.chromium.org/ui/p/chromium/builders/try/win-rel/1282304/overview > > > > > > Original change's description: > > > > Request refresh frame on resolution scaling changes > > > > > > > > When `scale_resolution_down_to` changes in `ConfigureEncoder`, the input > > > > resolution from the source is expected to change. To handle this > > > > cleanly, `RequestRefreshFrame` is now called to prompt the source for a > > > > frame matching the new constraints. > > > > > > > > Additionally, the immediate `ReconfigureEncoder` call is deferred if the > > > > resolution scaling has changed. The reconfiguration will instead occur > > > > upon the arrival of the next frame, ensuring the encoder is initialized > > > > with the correct resolution. > > > > > > > > Fixed: webrtc:467409350 > > > > Change-Id: Icbd19531a45d130460f4763a945d764df3e5e0d2 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432660 > > > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > Cr-Commit-Position: refs/heads/main@{#46473} > > > > > > Bug: webrtc:467409350 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Change-Id: I710abf84d42641c291fb3fb36bb0d42551504eae > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433740 > > > Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com> > > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > > Cr-Commit-Position: refs/heads/main@{#46482} > > > > Bug: webrtc:467409350 > > Change-Id: I2b4eca73001db9b3516bdbe1fea9df281e41a147 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441020 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#46598} > > Bug: webrtc:467409350 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Change-Id: I36dc644d1be3b5f96ae43763ab72c9932de75573 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441340 > Owners-Override: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#46610} Bug: webrtc:467409350 Change-Id: I8aae24dc60a084b7f80a646e2b40963c820f9e5f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/440946 Auto-Submit: Johannes Kron <kron@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46639}

This is likely not what we want. See this spec issue: rtcweb-wg/jsep#1040 Bug: None Change-Id: Id1f7164fa4e95c55ec14dd5dce9d5e3c5ea4b028 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/440947 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46640}

Align with spec draft 6. Bug: webrtc:447037083 Change-Id: Ic40a64a3f46504de171b4bb5d352581a19868c77 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/440201 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46641}

Calculates a "hold" time a receiver delay sending feedback. Add this hold time to RTT in order to use a more accurate time duration that the reference window represent. For example TWCC when media is sent only in one direction, a receiver waits roughly 90ms from a packet is received until feedback is sent. Also fix a bug where data in flight is not reported to Scream between feedback. Bug: webrtc:447037083 Change-Id: Ibe4f1072a9528952c2af94fb148f8ad79c770219 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/435360 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46642}

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…backend (#82)

…nces in future (#83) * [Fix] Remove deprecated jni_generator_include assignments The jni_zero template no longer accepts jni_generator_include as a parameter (removed upstream in M145). These stale assignments cause GN "assignment had no effect" errors and block the Android AAR build. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Resolve unresolved GN dependencies from M145 merge - Replace deprecated //system_wrappers:field_trial with //api/environment:deprecated_global_field_trials (3 occurrences) - Restore //api/crypto:frame_crypto_transformer target that was lost during merge (sources still exist, target definition was dropped) - Replace //test:scoped_key_value_config with //test:create_test_field_trials (renamed upstream in M145) Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Remove deprecated_global_field_trials from base_jni and java_audio_device_module These targets don't need the field trial dep (upstream M145 doesn't have it). The dep was only valid for peerconnection_jni which is in the target's visibility list. base_jni already has api:field_trials_view. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Remove poison_environment_construction violation and fix crypto deps - audio_device_module_base: replace default_task_queue_factory (which is poisonous) with api/task_queue + api/environment (matching upstream M145) - api/crypto:frame_crypto_transformer: fix task_queue_base dep (no such target; task_queue_base.h lives in the api/task_queue target) Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Add backward-compatible 2-arg VideoCodecInfo constructor The Stream fork changed the @CalledByNative constructor to take 3 args (name, params, scalabilityModes). Code throughout the codebase still calls the 2-arg version. Add a convenience constructor that delegates to the 3-arg version with an empty scalabilityModes list. Co-authored-by: Cursor <cursoragent@cursor.com> * Revert "[Fix] Add backward-compatible 2-arg VideoCodecInfo constructor" This reverts commit ef99cfa. * [Fix] Add missing environment_java target and AudioFrameProcessor source The M145 merge dropped the environment_java target and AudioFrameProcessor.java from peerconnection_java sources. PeerConnectionFactory.java references both classes, causing compilation failure. - Restore environment_java rtc_android_library target - Add generated_environment_jni generate_jni target - Add AudioFrameProcessor.java to peerconnection_java sources - Add environment_java as dep of peerconnection_java Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Update AudioProcessingFactory implementations for M145 API The AudioProcessingFactory interface now requires createNative(long webrtcEnvRef) instead of createNative(). Update both custom implementations to match the new signature. Note: downstream ManagedAudioProcessingFactory implementations (e.g. noise-cancellation library) will also need this update. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Update frame_crypto_transformer for M145 API changes - Replace removed `rtc::` namespace with `webrtc::` (Buffer, Thread, ArrayView, ByteBufferWriter, scoped_refptr, TimeMillis) - Replace `TransformableVideoFrameInterface::header().codec` with `Metadata().GetCodec()` - Replace `header().video_type_header` with `Metadata().GetRTPVideoHeaderCodecSpecifics()` - Add missing BUILD.gn deps for all included headers - Add missing standard library includes in header Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Correct mangled webArrayView -> webrtc::ArrayView The previous bulk replace of rtc::ArrayView matched inside webrtc::ArrayView, producing invalid webArrayView tokens. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Remove E2EE frame cryptor from Android build Remove frame_crypto_transformer, frame_cryptor, and data_packet_cryptor from the Android SDK build - matching iOS which also doesn't include these modules. The E2EE frame cryptor code has M145 incompatibilities that are not needed for the current build. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Remove cricket:: namespace from simulcast_video_encoder_factory In M145, cricket namespace was merged into webrtc namespace. kAv1CodecName is now in webrtc:: which is already imported via 'using namespace webrtc'. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Use ScopedJavaLocalRef instead of JavaParamRef constructor In M145, JavaParamRef is aliased to jni_zero::JavaRef which has a protected (env, obj) constructor. Use ScopedJavaLocalRef::Adopt with a new local ref instead. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Replace rtc::make_ref_counted with webrtc::make_ref_counted In external_audio_processing_factory.cc - the rtc namespace no longer exists in M145. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Update external_audio_processing_factory for M145 namespace changes Replace rtc::scoped_refptr with webrtc::scoped_refptr and convert raw pointer management to scoped_refptr for proper ref-counted lifetime. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Add Environment class to AAR and compile its JNI native impl The Environment.java class was not included in the dist_jar (due to direct_deps_only=true) and its native JNI implementation (environment.cc) was not being compiled, causing NoClassDefFoundError at runtime. Creates a separate environment_jni target matching upstream M145 structure and adds it to libjingle_peerconnection_jni's public_deps. Co-authored-by: Cursor <cursoragent@cursor.com> * [Refactor] Move custom modules to extensions/ to reduce merge conflicts Move simulcast and LiveKit audio processing code out of sdk/android/BUILD.gn into self-contained extensions/ directories: - extensions/android-simulcast/ — SimulcastVideoEncoder, SimulcastVideoEncoderFactory, SimulcastAlignedVideoEncoderFactory, DefaultAlignedVideoEncoderFactory (Java + JNI) - extensions/android-livekit-audio-processing/ — ExternalAudioProcessingFactory (Java + JNI) Add STREAM-CUSTOM marker comments to remaining custom lines in sdk/android/BUILD.gn for easy post-merge verification. This reduces custom lines in sdk/android/BUILD.gn from ~30 scattered entries to ~11 clearly-marked single-line deps, significantly reducing future merge conflict risk. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Correct VP9 dep path in extensions/android-simulcast BUILD.gn Use ../../modules/video_coding:webrtc_vp9 instead of the non-existent ../../modules/video_coding/codecs/vp9:libvpx_vp9 target. Co-authored-by: Cursor <cursoragent@cursor.com> * [Fix] Add missing base_java dep to livekit_audio_java for CalledByNative Co-authored-by: Cursor <cursoragent@cursor.com> * [Refactor] Merge audio processing extensions into android-audio-processing Consolidate android-external-audio-processing and android-livekit-audio-processing into a single extensions/android-audio-processing/ with standard layout: - java/org/webrtc/ — Java API classes - src/ — all C++ sources (native plugin + audio effects) Matches the same structure as android-simulcast. Removes confusing "livekit" naming from public Stream repo. Co-authored-by: Cursor <cursoragent@cursor.com> * [Refactor] Restructure extensions into extensions/android/ and move utility Java files - Move extensions/android-audio-processing → extensions/android/audio-processing - Move extensions/android-simulcast → extensions/android/simulcast - Create extensions/android/utilities for pure Java custom classes: ResolutionAdjustment, ManagedAudioProcessingFactory, DefaultBlacklistedVideoDecoderFactory - Update all relative paths in BUILD.gn files (../../ → ../../../) - Update C++ #include paths for new directory structure - Allows iOS extensions to use extensions/ios/ in the future Co-authored-by: Cursor <cursoragent@cursor.com> * Fix utilities_java build: add missing video_api_java dep DefaultBlacklistedVideoDecoderFactory uses VideoCodecInfo which lives in the video_api_java target. Co-authored-by: Cursor <cursoragent@cursor.com> * Fix simulcast_java build: add utilities_java dep for ResolutionAdjustment SimulcastAlignedVideoEncoderFactory and DefaultAlignedVideoEncoderFactory use ResolutionAdjustment which now lives in the utilities extension. Co-authored-by: Cursor <cursoragent@cursor.com> --------- Co-authored-by: Cursor <cursoragent@cursor.com>

Pass PeerConnectionFactoryDependencies by move in ObjC factory builder tests because the dependency bundle is intentionally non-copyable.

Harald Alvestrand and others added 30 commits December 5, 2025 09:03

Tommi and others added 16 commits January 12, 2026 06:37

Merge official WebRTC M137 baseline into Stream fork history

b91e4ba

[Enhancement] Merge official WebRTC M145 baseline

07cb6a9

Merge conflict fixes

9fb6fec

ipavlidakis self-assigned this Apr 28, 2026

ipavlidakis added the enhancement New feature or request label Apr 28, 2026

Stream Bot and others added 5 commits April 28, 2026 23:41

[Fix] Preserve iOS WebRTC umbrella header in framework bundle

a3935f5

Fix compilation errors

fb06fa1

[Fix] Preserve audio transport registration for active Stream ADM

9118621

fix: fixes objectFit="cover" rendering on iOS using the shared Metal …

ade0016

…backend (#82)

ipavlidakis changed the base branch from main to develop May 12, 2026 13:46

Stream Bot and others added 3 commits May 12, 2026 18:12

Restore crypto components and RTCAudioRenderer

6f40451

Fix move-only factory dependency test setup

6d6b1d9

Pass PeerConnectionFactoryDependencies by move in ObjC factory builder tests because the dependency bundle is intentionally non-copyable.

Iliaspavlidakis/m145/fix ios tests (#87)

5ba7a3a

ipavlidakis force-pushed the iliaspavlidakis/merge-webrtc-m145 branch from bbb2310 to 5ba7a3a Compare May 14, 2026 09:16

Resolve merge conflicts with develop

ad51315

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Upgrade to webrtc m145#81

Upgrade to webrtc m145#81
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ipavlidakis commented Apr 28, 2026

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coderabbitai Bot commented Apr 28, 2026 •

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ipavlidakis commented Apr 28, 2026

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coderabbitai Bot commented Apr 28, 2026 •

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