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adaptation
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Implement fake PixelLimitResource for TestBed.
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2020-12-04 10:35:53 +00:00 |
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test
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Adds ability to delay pacer start until media is added.
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2020-09-14 21:42:55 +00:00 |
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audio_receive_stream.cc
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Remove chromium clang style errors affecting sdk/android/media_jni
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2018-04-09 13:55:49 +00:00 |
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audio_receive_stream.h
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Reduce redundant flags for audio stream playout state.
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2021-01-18 09:09:09 +00:00 |
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audio_send_stream.cc
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Include configured mid and rid when logging AudioSendStream config
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2020-12-22 14:19:33 +00:00 |
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audio_send_stream.h
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negotiate RED codec for audio
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2020-06-25 06:24:18 +00:00 |
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audio_sender.h
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Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
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2020-01-13 18:31:30 +00:00 |
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audio_state.cc
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Remove chromium clang style errors affecting sdk/android/media_jni
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2018-04-09 13:55:49 +00:00 |
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audio_state.h
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Async audio processing API
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2020-10-02 12:33:34 +00:00 |
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bitrate_allocator.cc
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Replace DataSize and DataRate factories with newer versions
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2020-02-18 16:09:50 +00:00 |
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bitrate_allocator.h
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Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
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2020-11-23 11:29:36 +00:00 |
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bitrate_allocator_unittest.cc
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In call/ replace mock macros with unified MOCK_METHOD macro
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2020-05-15 13:36:00 +00:00 |
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bitrate_estimator_tests.cc
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Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
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2020-08-06 11:50:08 +00:00 |
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BUILD.gn
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Move deprecated code to their own build targets.
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2021-01-18 13:09:47 +00:00 |
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call.cc
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Add DeliverPacketAsync method to PacketReceiver.
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2021-01-19 11:53:50 +00:00 |
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call.h
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Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
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2020-09-22 16:08:22 +00:00 |
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call_config.cc
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[Cleanup] Add missing #include. Remove useless ones.
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2018-10-23 11:32:56 +00:00 |
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call_config.h
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Remove deprecated constant.
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2020-04-27 10:32:45 +00:00 |
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call_factory.cc
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Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread
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2020-06-30 15:38:35 +00:00 |
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call_factory.h
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Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
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2020-11-23 11:29:36 +00:00 |
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call_perf_tests.cc
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Fix data race on destruction in Audio Video sync tests
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2020-12-30 17:11:31 +00:00 |
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call_unittest.cc
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Move deprecated code to their own build targets.
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2021-01-18 13:09:47 +00:00 |
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degraded_call.cc
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Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
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2020-09-22 16:08:22 +00:00 |
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degraded_call.h
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Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
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2020-09-22 16:08:22 +00:00 |
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DEPS
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Async audio processing API
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2020-10-02 12:33:34 +00:00 |
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fake_network_pipe.cc
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Migrate call/ to webrtc::Mutex.
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2020-07-06 15:48:30 +00:00 |
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fake_network_pipe.h
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Migrate call/ to webrtc::Mutex.
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2020-07-06 15:48:30 +00:00 |
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fake_network_pipe_unittest.cc
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In call/ replace mock macros with unified MOCK_METHOD macro
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2020-05-15 13:36:00 +00:00 |
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flexfec_receive_stream.cc
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[Cleanup] Add missing #include. Remove useless ones.
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2018-10-23 11:32:56 +00:00 |
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flexfec_receive_stream.h
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Format almost everything.
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2019-07-08 13:45:15 +00:00 |
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flexfec_receive_stream_impl.cc
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Revert "Add task queue to RtpRtcpInterface::Configuration."
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2021-01-12 17:47:32 +00:00 |
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flexfec_receive_stream_impl.h
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Remove dependency from RtpRtcp on the Module interface.
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2020-06-04 08:11:21 +00:00 |
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flexfec_receive_stream_unittest.cc
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Use std::make_unique instead of absl::make_unique.
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2019-09-17 15:47:29 +00:00 |
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OWNERS
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Make sprang@ owner in call
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2020-10-19 10:30:03 +00:00 |
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packet_receiver.h
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Add DeliverPacketAsync method to PacketReceiver.
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2021-01-19 11:53:50 +00:00 |
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rampup_tests.cc
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Default enable sending transport sequence numbers on audio packets.
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2020-11-24 09:19:54 +00:00 |
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rampup_tests.h
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Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
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2019-10-21 12:33:27 +00:00 |
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receive_time_calculator.cc
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Use newer version of TimeDelta and TimeStamp factories in webrtc
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2020-02-10 12:21:17 +00:00 |
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receive_time_calculator.h
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Format almost everything.
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2019-07-08 13:45:15 +00:00 |
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receive_time_calculator_unittest.cc
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Format almost everything.
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2019-07-08 13:45:15 +00:00 |
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rtp_bitrate_configurator.cc
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Allow setting a bandwidth cap for relayed connections.
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2020-03-26 20:41:46 +00:00 |
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rtp_bitrate_configurator.h
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Allow setting a bandwidth cap for relayed connections.
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2020-03-26 20:41:46 +00:00 |
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rtp_bitrate_configurator_unittest.cc
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Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
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2020-01-10 16:39:51 +00:00 |
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rtp_config.cc
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Reland "Improve outbound-rtp statistics for simulcast"
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2020-05-05 20:22:19 +00:00 |
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rtp_config.h
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Reland "Improve outbound-rtp statistics for simulcast"
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2020-05-05 20:22:19 +00:00 |
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rtp_demuxer.cc
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Remove lock from RtpStreamReceiverController.
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2021-01-18 09:10:14 +00:00 |
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rtp_demuxer.h
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Remove lock from RtpStreamReceiverController.
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2021-01-18 09:10:14 +00:00 |
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rtp_demuxer_unittest.cc
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Delete callbacks from RtpDemuxer on ssrc binding
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2020-07-17 15:41:39 +00:00 |
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rtp_packet_sink_interface.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_payload_params.cc
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Ignore frame type when calculating generic frame dependencies.
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2020-11-23 10:52:06 +00:00 |
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rtp_payload_params.h
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Ignore frame type when calculating generic frame dependencies.
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2020-11-23 10:52:06 +00:00 |
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rtp_payload_params_unittest.cc
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in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
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2020-11-11 14:00:52 +00:00 |
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rtp_stream_receiver_controller.cc
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Remove lock from RtpStreamReceiverController.
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2021-01-18 09:10:14 +00:00 |
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rtp_stream_receiver_controller.h
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Remove lock from RtpStreamReceiverController.
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2021-01-18 09:10:14 +00:00 |
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rtp_stream_receiver_controller_interface.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_transport_controller_send.cc
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Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
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2020-11-02 11:05:56 +00:00 |
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rtp_transport_controller_send.h
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Adds ability to delay pacer start until media is added.
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2020-09-14 21:42:55 +00:00 |
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rtp_transport_controller_send_interface.h
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Delete unneeded dependencies on the Module abstraction
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2020-12-21 09:09:57 +00:00 |
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rtp_video_sender.cc
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Fix enabling DependencyDescriptor for VP9 with spatial layers
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2021-01-20 15:03:37 +00:00 |
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rtp_video_sender.h
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Use frame rate in video overhead calculation.
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2020-12-01 09:42:40 +00:00 |
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rtp_video_sender_interface.h
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Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender.
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2020-10-19 11:37:23 +00:00 |
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rtp_video_sender_unittest.cc
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Fix enabling DependencyDescriptor for VP9 with spatial layers
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2021-01-20 15:03:37 +00:00 |
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rtx_receive_stream.cc
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Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
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2019-12-03 21:10:53 +00:00 |
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rtx_receive_stream.h
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IWYU: uint32_t is defined in cstdint
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2020-05-07 17:04:15 +00:00 |
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rtx_receive_stream_unittest.cc
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Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
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2019-12-03 21:10:53 +00:00 |
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simulated_network.cc
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Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
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2020-11-09 10:47:55 +00:00 |
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simulated_network.h
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Migrate call/ to webrtc::Mutex.
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2020-07-06 15:48:30 +00:00 |
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simulated_network_unittest.cc
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Replace DataSize and DataRate factories with newer versions
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2020-02-18 16:09:50 +00:00 |
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simulated_packet_receiver.h
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Calculate next process time in simulated network.
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2019-02-08 19:33:17 +00:00 |
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syncable.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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syncable.h
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Make AV sync robust to failures to set a desired minimum delay
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2020-09-09 15:44:47 +00:00 |
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version.cc
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Update WebRTC code version (2021-01-24T04:03:52).
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2021-01-24 05:50:19 +00:00 |
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version.h
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Add WebRTC code freshness version string.
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2020-12-14 16:22:35 +00:00 |
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video_receive_stream.cc
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Add commas between codec parameters in VideoReceiveStream logging.
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2020-03-09 02:45:34 +00:00 |
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video_receive_stream.h
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Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
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2020-08-27 09:53:18 +00:00 |
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video_send_stream.cc
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Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
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2020-11-09 10:47:55 +00:00 |
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video_send_stream.h
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Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
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2020-08-27 09:53:18 +00:00 |
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