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webrtc-version-updater 1261863f87 Update WebRTC code version (2024-04-15T04:12:23). 
Bug: None
Change-Id: I09a28275d7ab591ddeb0967aba75a632c468475b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347361
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@{#42061}
2024-04-15 05:10:52 +00:00
..
adaptation
Delete rtc::TaskQueue
2024-02-28 10:22:49 +00:00
test
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
audio_receive_stream.cc
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audio_receive_stream.h
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audio_send_stream.cc
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audio_send_stream.h
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audio_sender.h
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audio_state.cc
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audio_state.h
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bitrate_allocator_unittest.cc
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bitrate_allocator.cc
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bitrate_allocator.h
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bitrate_estimator_tests.cc
Update CallTests to create Call using Environment
2023-12-01 13:16:41 +00:00
BUILD.gn
Remove unnecessary RtcEventLog parameter in RtpTransportControllerSend::CreateRtpVideoSender
2024-03-06 16:24:06 +00:00
call_config.cc
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
call_config.h
Reland "FrameCadenceAdapter: align video encoding to metronome"
2024-01-08 13:54:56 +00:00
call_perf_tests.cc
Update expectation on first Sinkwants in CallPerfTest.ReceivesCpuOveruseAndUnderuse
2024-03-22 13:30:46 +00:00
call_unittest.cc
Pass Environment instead of clock to Fake video encoders at construction
2024-04-12 07:42:48 +00:00
call.cc
Propagate Environment into VideoStreamEncoder
2024-03-05 09:33:02 +00:00
call.h
Pass Clock through Environment when constructing Call
2023-12-06 19:13:39 +00:00
create_call.cc
Pass Clock through Environment when constructing Call
2023-12-06 19:13:39 +00:00
create_call.h
Delete CallFactoryInterface as no longer needed
2023-12-05 15:44:43 +00:00
degraded_call.cc
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degraded_call.h
Delete rtc::TaskQueue
2024-02-28 10:22:49 +00:00
DEPS
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fake_network_pipe_unittest.cc
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fake_network_pipe.cc
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fake_network_pipe.h
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flexfec_receive_stream_impl.cc
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flexfec_receive_stream_impl.h
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flexfec_receive_stream_unittest.cc
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flexfec_receive_stream.cc
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flexfec_receive_stream.h
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OWNERS
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packet_receiver.h
Dont create RTX receive stream before media SSRC is known
2024-02-22 14:40:43 +00:00
rampup_tests.cc
Update CallTests to create Call using Environment
2023-12-01 13:16:41 +00:00
rampup_tests.h
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receive_stream.h
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receive_time_calculator_unittest.cc
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receive_time_calculator.cc
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receive_time_calculator.h
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rtp_bitrate_configurator_unittest.cc
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rtp_bitrate_configurator.cc
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rtp_bitrate_configurator.h
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rtp_config.cc
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rtp_config.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_demuxer_unittest.cc
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rtp_demuxer.cc
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rtp_demuxer.h
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rtp_packet_sink_interface.h
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rtp_payload_params_unittest.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_payload_params.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_payload_params.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_stream_receiver_controller_interface.h
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rtp_stream_receiver_controller.cc
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rtp_stream_receiver_controller.h
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rtp_transport_config.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +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
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_interface.h
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_transport_controller_send.cc
Ignore allocated bitrate during initial exponential BWE.
2024-04-12 09:13:44 +00:00
rtp_transport_controller_send.h
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_video_sender_interface.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_video_sender_unittest.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_video_sender.cc
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_video_sender.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtx_receive_stream_unittest.cc
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rtx_receive_stream.cc
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rtx_receive_stream.h
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simulated_network_unittest.cc
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simulated_network.cc
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simulated_network.h
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simulated_packet_receiver.h
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syncable.cc
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syncable.h
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version.cc
Update WebRTC code version (2024-04-15T04:12:23).
2024-04-15 05:10:52 +00:00
version.h
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video_receive_stream.cc
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video_receive_stream.h
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video_send_stream.cc
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video_send_stream.h
Remove VideoSendStream::StartPerRtpStream
2024-01-26 09:19:50 +00:00