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webrtc-version-updater 7fac89f3f9 Update WebRTC code version (2024-07-11T04:05:56). 
Bug: None
Change-Id: Icba831622f7cb10cab9dde73e45e9a5d18035be2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356960
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@{#42621}
2024-07-11 06:07:00 +00:00
..
adaptation
Deprecate rtc::RefCountInterface
2024-06-07 09:47:26 +00:00
test
Expose a PeerConnection's NetworkControllerInterface instance
2024-06-18 08:04:03 +00:00
audio_receive_stream.cc
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audio_receive_stream.h
Computing and propagating the audio stats totalprocessingdelay.
2024-06-18 08:05:28 +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
Deprecate rtc::RefCountInterface
2024-06-07 09:47:26 +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
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
Reland "FrameCadenceAdapter: align video encoding to metronome"
2024-01-08 13:54:56 +00:00
call_perf_tests.cc
Ensure test capturer does not set ntp time
2024-05-29 10:26:54 +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
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
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
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
…
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
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +00:00
rampup_tests.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
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
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
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rtp_transport_controller_send_factory.h
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rtp_transport_controller_send_interface.h
Expose a PeerConnection's NetworkControllerInterface instance
2024-06-18 08:04:03 +00:00
rtp_transport_controller_send.cc
Fix GoogCcNetworkController::OnNetworkStateEstimate behaviour
2024-06-10 10:43:57 +00:00
rtp_transport_controller_send.h
Expose a PeerConnection's NetworkControllerInterface instance
2024-06-18 08:04:03 +00:00
rtp_video_sender_interface.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
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
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 (2024-07-11T04:05:56).
2024-07-11 06:07:00 +00:00
version.h
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video_receive_stream.cc
…
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
Remove VideoSendStream::StartPerRtpStream
2024-01-26 09:19:50 +00:00