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webrtc-version-updater 3cfebdf0df Update WebRTC code version (2024-06-01T04:02:50). 
Bug: None
Change-Id: I9dfdcb0731a662e5df553680ab46a3d148f7cd7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353300
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@{#42424}
2024-06-01 05:42:11 +00:00
..
adaptation
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
test
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audio_receive_stream.cc
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audio_receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
audio_send_stream.cc
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audio_send_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
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
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BUILD.gn
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
call_config.cc
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call_config.h
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call_perf_tests.cc
Ensure test capturer does not set ntp time
2024-05-29 10:26:54 +00:00
call_unittest.cc
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call.cc
Expose network to deliverrtp histogram metric
2024-05-29 08:50:22 +00:00
call.h
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create_call.cc
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create_call.h
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degraded_call.cc
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degraded_call.h
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DEPS
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fake_network_pipe_unittest.cc
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +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
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rampup_tests.cc
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +00:00
rampup_tests.h
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receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
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
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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
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rtp_payload_params.cc
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rtp_payload_params.h
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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
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rtp_transport_controller_send_factory_interface.h
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rtp_transport_controller_send_factory.h
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rtp_transport_controller_send_interface.h
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rtp_transport_controller_send.cc
Add field trial to reset BWE if Network adapter changes instead of if IP address changes
2024-05-16 12:17:51 +00:00
rtp_transport_controller_send.h
Add field trial to reset BWE if Network adapter changes instead of if IP address changes
2024-05-16 12:17:51 +00:00
rtp_video_sender_interface.h
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rtp_video_sender_unittest.cc
Change WebRTC-Video-SimulcastIndependentFrameIds to a kill-switch.
2024-05-23 09:42:23 +00:00
rtp_video_sender.cc
Change WebRTC-Video-SimulcastIndependentFrameIds to a kill-switch.
2024-05-23 09:42:23 +00:00
rtp_video_sender.h
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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
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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 (2024-06-01T04:02:50).
2024-06-01 05:42:11 +00:00
version.h
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video_receive_stream.cc
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video_receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
video_send_stream.cc
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video_send_stream.h
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