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adaptation
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Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
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2023-05-03 11:09:26 +00:00 |
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test
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Add PeerConnectionInterface::ReconfigureBandwidthEstimation
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2024-02-07 14:10:02 +00:00 |
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audio_receive_stream.cc
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Rename AudioReceiveStream to AudioReceiveStreamInterface
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2022-05-23 08:44:26 +00:00 |
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audio_receive_stream.h
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Add Rust_setIncomingAudioMuted
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2023-09-27 12:16:54 -04:00 |
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audio_send_stream.cc
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audio_send_stream.h
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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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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.cc
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Cleanup merge differences from upstream
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2023-01-25 17:17:55 -08:00 |
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bitrate_allocator.h
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bitrate_allocator_unittest.cc
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Adopt absl::string_view in call/
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2022-05-17 12:00:45 +00:00 |
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bitrate_estimator_tests.cc
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Update CallTests to create Call using Environment
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2023-12-01 13:16:41 +00:00 |
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BUILD.gn
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Add PeerConnectionInterface::ReconfigureBandwidthEstimation
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2024-02-07 14:10:02 +00:00 |
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call.cc
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Merge branch m122
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2024-02-14 22:44:28 -08:00 |
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call.h
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Merge branch m122
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2024-02-14 22:44:28 -08:00 |
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call_config.cc
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Use Environment in RtpTransportyControllerSend
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2023-12-20 14:47:51 +00:00 |
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call_config.h
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Reland "FrameCadenceAdapter: align video encoding to metronome"
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2024-01-08 13:54:56 +00:00 |
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call_perf_tests.cc
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Update CallTests to create Call using Environment
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2023-12-01 13:16:41 +00:00 |
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call_unittest.cc
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Allow to create webrtc::Call with Environment
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2023-11-28 10:26:56 +00:00 |
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create_call.cc
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Pass Clock through Environment when constructing Call
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2023-12-06 19:13:39 +00:00 |
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create_call.h
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Delete CallFactoryInterface as no longer needed
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2023-12-05 15:44:43 +00:00 |
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degraded_call.cc
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Remove internal overrides using old SendRtp and SendRtcp interfaces.
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2023-08-15 13:20:21 +00:00 |
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degraded_call.h
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Remove internal overrides using old SendRtp and SendRtcp interfaces.
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2023-08-15 13:20:21 +00:00 |
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DEPS
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fake_network_pipe.cc
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Delete unused constructor of FakeNetworkPipe
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2023-08-18 13:07:10 +00:00 |
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fake_network_pipe.h
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Delete unused constructor of FakeNetworkPipe
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2023-08-18 13:07:10 +00:00 |
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fake_network_pipe_unittest.cc
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Reland "Delete PacketReceiver::DeliverPacket from all implementations"
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2023-01-25 18:18:29 +00:00 |
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flexfec_receive_stream.cc
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flexfec_receive_stream.h
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stats: implement flexfec fecBytesReceived stats for FlexFEC
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2023-06-21 13:04:31 +00:00 |
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flexfec_receive_stream_impl.cc
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stats: implement flexfec fecBytesReceived stats for FlexFEC
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2023-06-21 13:04:31 +00:00 |
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flexfec_receive_stream_impl.h
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stats: implement flexfec fecBytesReceived stats for FlexFEC
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2023-06-21 13:04:31 +00:00 |
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flexfec_receive_stream_unittest.cc
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Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
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2023-05-03 11:09:26 +00:00 |
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OWNERS
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Finish resolving merge conflicts
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2022-11-11 19:10:59 -05:00 |
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packet_receiver.h
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Allow injecting packets of type Any to Call::DeliverRtpPacket
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2023-03-29 06:36:17 +00:00 |
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rampup_tests.cc
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Update CallTests to create Call using Environment
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2023-12-01 13:16:41 +00:00 |
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rampup_tests.h
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Stop overriding extensions in rampup tests
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2023-01-25 13:18:49 +00:00 |
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receive_stream.h
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Remove rtp header extension from config of Call audio and video receivers
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2023-01-31 11:58:43 +00:00 |
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receive_time_calculator.cc
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Update to 5005 (M102) (#86)
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2022-08-24 11:07:33 -04:00 |
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receive_time_calculator.h
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Update to 5005 (M102) (#86)
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2022-08-24 11:07:33 -04:00 |
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receive_time_calculator_unittest.cc
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Update to 4896 (M100) (#72)
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2022-04-15 17:13:23 -06:00 |
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rtp_bitrate_configurator.cc
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rtp_bitrate_configurator.h
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Update to 4896 (M100) (#72)
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2022-04-15 17:13:23 -06:00 |
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rtp_bitrate_configurator_unittest.cc
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rtp_config.cc
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Update to 4896 (M100) (#72)
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2022-04-15 17:13:23 -06:00 |
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rtp_config.h
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Introduce support for video packet batching.
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2023-05-08 16:24:03 +00:00 |
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rtp_demuxer.cc
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Remove SSRCs from libSRTP when removing them from the rtp_demuxer
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2023-11-08 10:24:10 +00:00 |
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rtp_demuxer.h
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Remove SSRCs from libSRTP when removing them from the rtp_demuxer
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2023-11-08 10:24:10 +00:00 |
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rtp_demuxer_unittest.cc
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Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
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2023-05-03 11:09:26 +00:00 |
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rtp_packet_sink_interface.h
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rtp_payload_params.cc
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Add codec name H265 to support H265 in WebRTC
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2023-09-20 09:25:32 +00:00 |
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rtp_payload_params.h
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Revert "Use shared_frame_id for the VP8 picture ID even if not using the generic header (#39)"
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2023-05-04 11:57:45 -04:00 |
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rtp_payload_params_unittest.cc
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When simulating chains from VP9 codec specific info support first_active_layer > 0
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2023-08-03 13:19:00 +00:00 |
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rtp_stream_receiver_controller.cc
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Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
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2022-12-22 14:04:21 +00:00 |
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rtp_stream_receiver_controller.h
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Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
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2022-12-22 14:04:21 +00:00 |
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rtp_stream_receiver_controller_interface.h
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Demote RtpStreamReceiverController AddSink/RemoveSink to private
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2022-07-06 09:31:54 +00:00 |
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rtp_transport_config.h
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Add PeerConnectionInterface::ReconfigureBandwidthEstimation
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2024-02-07 14:10:02 +00:00 |
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rtp_transport_controller_send.cc
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Add PeerConnectionInterface::ReconfigureBandwidthEstimation
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2024-02-07 14:10:02 +00:00 |
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rtp_transport_controller_send.h
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Add PeerConnectionInterface::ReconfigureBandwidthEstimation
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2024-02-07 14:10:02 +00:00 |
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rtp_transport_controller_send_factory.h
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Use Environment in RtpTransportyControllerSend
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2023-12-20 14:47:51 +00:00 |
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rtp_transport_controller_send_factory_interface.h
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Use Environment in RtpTransportyControllerSend
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2023-12-20 14:47:51 +00:00 |
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rtp_transport_controller_send_interface.h
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Add PeerConnectionInterface::ReconfigureBandwidthEstimation
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2024-02-07 14:10:02 +00:00 |
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rtp_video_sender.cc
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Merge m123/6312
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2024-06-12 22:25:35 -07:00 |
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rtp_video_sender.h
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Refactor RtpVideoSender::SetActiveModules.
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2024-01-26 10:34:46 +00:00 |
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rtp_video_sender_interface.h
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Refactor RtpVideoSender::SetActiveModules.
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2024-01-26 10:34:46 +00:00 |
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rtp_video_sender_unittest.cc
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Consolidate encoded transform mocks into api/test/
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2024-01-26 12:46:34 +00:00 |
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rtx_receive_stream.cc
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Updated associated payload types without recreating receive streams.
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2022-08-16 13:38:24 +00:00 |
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rtx_receive_stream.h
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Updated associated payload types without recreating receive streams.
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2022-08-16 13:38:24 +00:00 |
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rtx_receive_stream_unittest.cc
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simulated_network.cc
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Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
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2022-11-06 13:14:26 +00:00 |
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simulated_network.h
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Export webrtc::SimulatedNetwork for Chrome component builds
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2023-11-27 16:03:23 +00:00 |
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simulated_network_unittest.cc
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Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
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2022-11-06 13:14:26 +00:00 |
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simulated_packet_receiver.h
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syncable.cc
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syncable.h
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Rename AudioReceiveStream to AudioReceiveStreamInterface
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2022-05-23 08:44:26 +00:00 |
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version.cc
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Update WebRTC code version (2024-02-18T04:06:34).
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2024-02-18 05:25:06 +00:00 |
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version.h
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video_receive_stream.cc
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Add missing comma in VideoReceiveStreamInterface::Stats::ToString
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2023-10-17 10:42:06 +00:00 |
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video_receive_stream.h
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Remove default "unknown" encoderImplementation/decoderImplementation
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2023-06-22 11:49:58 +00:00 |
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video_send_stream.cc
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Cleanup usasge of ReportBlockData::report_block accessor
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2023-05-05 09:56:30 +00:00 |
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video_send_stream.h
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Remove VideoSendStream::StartPerRtpStream
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2024-01-26 09:19:50 +00:00 |
