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adaptation
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[Adaptation] Refactor AdaptationTarget. Peek next restrictions.
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2020-03-14 11:29:03 +00:00 |
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test
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Insert frame transformer between Encoded and Packetizer.
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2020-02-28 07:43:13 +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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Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
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2020-03-11 12:08:32 +00:00 |
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audio_send_stream.cc
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Delete media transport integration.
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2019-11-26 19:19:36 +00:00 |
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audio_send_stream.h
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Insert audio frame transformer between encoder and packetizer.
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2020-03-31 11:14:00 +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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[getStats] Implement "media-source" audio levels, fixing Chrome bug.
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2019-07-04 08:13:45 +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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Converts const methods in BitrateAllocator to non-member functions.
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2019-09-25 11:55:13 +00:00 |
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bitrate_allocator_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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bitrate_estimator_tests.cc
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Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
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2019-12-07 00:54:26 +00:00 |
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BUILD.gn
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Insert audio frame transformer between encoder and packetizer.
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2020-03-31 11:14:00 +00:00 |
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call.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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call.h
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Remove MediaTransport from Call.
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2019-08-08 10:58:57 +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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Adds injectable trials from peerconnection down to transport controller.
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2019-11-21 12:41:45 +00:00 |
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call_factory.cc
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DegradedCall: fake network using TaskQueue instead of ProcessThread
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2019-08-06 15:05:30 +00:00 |
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call_factory.h
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(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
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2019-01-11 17:11:39 +00:00 |
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call_perf_tests.cc
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Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
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2020-01-07 13:02:52 +00:00 |
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call_unittest.cc
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Trials should always be populated in call config.
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2019-12-03 10:34:55 +00:00 |
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degraded_call.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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degraded_call.h
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Make fake network degradation work also for sent audio
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2019-08-12 15:20:18 +00:00 |
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DEPS
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Make fec controller plug-able.
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2018-01-22 11:48:16 +00:00 |
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fake_network_pipe.cc
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Make fake network degradation work also for sent audio
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2019-08-12 15:20:18 +00:00 |
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fake_network_pipe.h
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Make fake network degradation work also for sent audio
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2019-08-12 15:20:18 +00:00 |
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fake_network_pipe_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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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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Concatenate string literals at compile time.
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2020-01-14 14:47:48 +00:00 |
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flexfec_receive_stream_impl.h
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Injecting Clock in video receive.
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2019-03-04 21:53:57 +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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Add terelius as OWNER in call/
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2020-03-23 09:55:34 +00:00 |
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packet_receiver.h
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(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
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2019-01-11 17:11:39 +00:00 |
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rampup_tests.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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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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rtcp_demuxer.cc
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Format almost everything.
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2019-07-08 13:45:15 +00:00 |
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rtcp_demuxer.h
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +00:00 |
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rtcp_demuxer_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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rtcp_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_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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[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
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2020-03-24 13:31:54 +00:00 |
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rtp_config.h
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[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
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2020-03-24 13:31:54 +00:00 |
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rtp_demuxer.cc
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Log details when RtpDemuxer fails to deliver a packet
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2019-04-16 00:47:53 +00:00 |
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rtp_demuxer.h
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Log details when RtpDemuxer fails to deliver a packet
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2019-04-16 00:47:53 +00:00 |
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rtp_demuxer_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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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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Ship GenericDescriptor00 by default.
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2020-02-18 11:11:48 +00:00 |
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rtp_payload_params.h
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Populate generic descriptor based on GenericFrameInfo when available.
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2020-02-12 10:55:41 +00:00 |
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rtp_payload_params_unittest.cc
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Ship GenericDescriptor00 by default.
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2020-02-18 11:11:48 +00:00 |
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rtp_rtcp_demuxer_helper.cc
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Replace rtc::Optional with absl::optional in audio, call and video
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2018-06-15 12:09:49 +00:00 |
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rtp_rtcp_demuxer_helper.h
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[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
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2018-11-28 18:25:07 +00:00 |
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rtp_rtcp_demuxer_helper_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_stream_receiver_controller.cc
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Concatenate string literals at compile time.
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2020-01-14 14:47:48 +00:00 |
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rtp_stream_receiver_controller.h
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(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
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2019-01-11 17:11:39 +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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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_transport_controller_send.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_transport_controller_send_interface.h
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Insert frame transformer between Encoded and Packetizer.
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2020-02-28 07:43:13 +00:00 |
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rtp_video_sender.cc
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[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
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2020-03-24 13:31:54 +00:00 |
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rtp_video_sender.h
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Reland "Reland "Refactors UlpFec and FlexFec to use a common interface.""
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2020-03-09 13:41:35 +00:00 |
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rtp_video_sender_interface.h
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Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
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2019-10-15 14:40:48 +00:00 |
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rtp_video_sender_unittest.cc
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Insert frame transformer between Encoded and Packetizer.
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2020-02-28 07:43:13 +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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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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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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Adds UpdateConfig to SimulatedNetwork
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2020-03-16 15:58:43 +00:00 |
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simulated_network.h
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Adds UpdateConfig to SimulatedNetwork
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2020-03-16 15:58:43 +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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ssrc_binding_observer.h
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Delete unneeded includes of basictypes.h.
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2018-05-21 19:35:08 +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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Add periodic logging of sync delays.
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2020-02-11 09:43:49 +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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[InsertableStreams] Set video frame transformer if RTP stream already started.
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2020-03-31 14:07:29 +00:00 |
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video_send_stream.cc
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[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
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2020-03-24 13:31:54 +00:00 |
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video_send_stream.h
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[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
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2020-03-24 13:31:54 +00:00 |
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