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webrtc-version-updater 115aea4774 Update WebRTC code version (2024-10-25T04:04:48). 
Bug: None
Change-Id: Ia2ded6a6fa56535ecbb5ef7a5cb94da243e09dd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366625
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@{#43303}
2024-10-25 06:12:34 +00:00
..
adaptation
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
test
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Change timestamps type from int64 to Timestamp in MediaReceiverInfo.
2024-10-16 11:02:37 +00:00
audio_send_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
audio_send_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_state.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_state.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
bitrate_allocator_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
bitrate_allocator.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
bitrate_allocator.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
bitrate_estimator_tests.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
BUILD.gn
Remove temporary video_stream_api target
2024-10-24 08:55:44 +00:00
call_config.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
call_config.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
call_perf_tests.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
call_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
call.cc
Use PayloadTypePicker in WebRtcVoiceEngine
2024-09-09 18:44:21 +00:00
call.h
Move the payload type picker to call/
2024-09-03 12:36:50 +00:00
create_call.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
DEPS
Revert "Use Payload Type suggester for all codec merging"
2024-10-23 11:37:18 +00:00
fake_network_pipe_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
fake_network_pipe.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
fake_network_pipe.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream_impl.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream_impl.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
OWNERS
Make perkj owner of call/
2024-05-07 15:13:29 +00:00
packet_receiver.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
payload_type_picker_unittest.cc
Add comparators unittest, and abandon MatchesForSdp
2024-10-10 14:33:13 +00:00
payload_type_picker.cc
Revert "Use Payload Type suggester for all codec merging"
2024-10-23 11:37:18 +00:00
payload_type_picker.h
Move the payload type picker to call/
2024-09-03 12:36:50 +00:00
payload_type.h
Revert "Use Payload Type suggester for all codec merging"
2024-10-23 11:37:18 +00:00
rampup_tests.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rampup_tests.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
receive_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
receive_time_calculator_unittest.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
receive_time_calculator.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_bitrate_configurator_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_bitrate_configurator.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_bitrate_configurator.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
rtp_config.cc
Revert "Allow sending to separate payload types for each simulcast index."
2024-10-09 16:13:07 +00:00
rtp_config.h
Revert "Allow sending to separate payload types for each simulcast index."
2024-10-09 16:13:07 +00:00
rtp_demuxer_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_demuxer.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_demuxer.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Add dependency descriptor support for H264 when no template information
2024-10-17 14:47:23 +00:00
rtp_payload_params.cc
Add dependency descriptor support for H264 when no template information
2024-10-17 14:47:23 +00:00
rtp_payload_params.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_stream_receiver_controller_interface.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_stream_receiver_controller.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_stream_receiver_controller.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_transport_config.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_transport_controller_send_factory_interface.h
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_factory.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_transport_controller_send_interface.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_transport_controller_send.cc
Prepare TransportFeedbackAdapter for RFC8888
2024-10-18 11:24:47 +00:00
rtp_transport_controller_send.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_video_sender_interface.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_video_sender_unittest.cc
Add dependency descriptor support for H264 when no template information
2024-10-17 14:47:23 +00:00
rtp_video_sender.cc
Revert "Allow sending to separate payload types for each simulcast index."
2024-10-09 16:13:07 +00:00
rtp_video_sender.h
Prune packets from pacer when sending deactivated via bitrate allocation
2024-09-24 16:41:56 +00:00
rtx_receive_stream_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtx_receive_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtx_receive_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
simulated_network.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
simulated_packet_receiver.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
syncable.cc
…
syncable.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
version.cc
Update WebRTC code version (2024-10-25T04:04:48).
2024-10-25 06:12:34 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
video_receive_stream.h
Change timestamps type from int64 to Timestamp in MediaReceiverInfo.
2024-10-16 11:02:37 +00:00
video_send_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
video_send_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00