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webrtc-version-updater ded1b97aeb Update WebRTC code version (2024-05-13T04:02:33). 
Bug: None
Change-Id: I678c11728d58d1fca8705639b83ce95be275c061
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350433
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@{#42278}
2024-05-13 05:08:27 +00:00
..
adaptation
Delete rtc::TaskQueue
2024-02-28 10:22:49 +00:00
test
Add missing header
2024-04-19 06:13:16 +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
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_sender.h
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audio_state.cc
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audio_state.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
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
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
BUILD.gn
Move call/simulated_network to test/network
2024-04-29 09:55:06 +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
Move call/simulated_network to test/network
2024-04-29 09:55:06 +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
Provide Environment for ReceiveSideConfestionController construction
2024-05-07 08:02:36 +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
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
DEPS
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
fake_network_pipe_unittest.cc
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
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
Make perkj owner of call/
2024-05-07 15:13:29 +00:00
packet_receiver.h
Dont create RTX receive stream before media SSRC is known
2024-02-22 14:40:43 +00:00
rampup_tests.cc
In EncoderStreamFactory pass field trials as required parameter
2024-04-17 12:53:30 +00:00
rampup_tests.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
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
Add option to provide Environment for CongestionConroller construction
2024-05-08 12:46:23 +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.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
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 (2024-05-13T04:02:33).
2024-05-13 05:08:27 +00:00
version.h
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video_receive_stream.cc
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video_receive_stream.h
stats: implement remote-outbound-rtp for video
2024-04-15 15:10:54 +00:00
video_send_stream.cc
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video_send_stream.h
Remove VideoSendStream::StartPerRtpStream
2024-01-26 09:19:50 +00:00