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test
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Include absl/memory/memory.h if absl::make_unique is used
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2019-01-08 20:08:32 +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 PeerConnection option to enable RTX handling in the audio jitter buffer.
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2019-01-10 16:28:43 +00:00 |
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audio_send_stream.cc
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Remove more unneeded things from ChannelSend
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2018-11-21 09:04:07 +00:00 |
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audio_send_stream.h
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Simulcast work to enable RID mux.
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2018-12-21 20:59:23 +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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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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bitrate_allocator_unittest.cc
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Don't trigger too many probes when max allocated bitrate changes.
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2018-12-11 16:19:53 +00:00 |
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bitrate_allocator.cc
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Signals start rate when registering to TargetTransferRateObserver.
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2018-12-12 09:22:32 +00:00 |
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bitrate_allocator.h
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Signals start rate when registering to TargetTransferRateObserver.
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2018-12-12 09:22:32 +00:00 |
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bitrate_estimator_tests.cc
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Include absl/memory/memory.h if absl::make_unique is used
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2019-01-08 20:08:32 +00:00 |
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BUILD.gn
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Include absl/memory/memory.h if absl::make_unique is used
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2019-01-08 20:08:32 +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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Delete VideoReceiveStream::EnableEncodedFrameRecording
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2018-10-02 10:31:46 +00:00 |
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call_perf_tests.cc
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Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
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2018-11-22 11:23:37 +00:00 |
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call_unittest.cc
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Remove Transport implementation from ChannelSend
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2018-12-18 09:34:52 +00:00 |
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call.cc
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Register stat callbacks after rate observer is registered.
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2018-12-19 21:19:01 +00:00 |
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call.h
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Delete unneeded includes of common_types.h and gn deps on webrtc_common.
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2018-11-20 16:28:39 +00:00 |
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callfactory.cc
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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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callfactory.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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degraded_call.cc
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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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degraded_call.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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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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[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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fake_network_pipe.h
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Delete rtc::PacketTime (was an alias for int64_t)
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2019-01-02 09:07:51 +00:00 |
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flexfec_receive_stream_impl.cc
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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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flexfec_receive_stream_impl.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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flexfec_receive_stream_unittest.cc
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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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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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[Cleanup] Add missing #include. Remove useless ones.
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2018-10-23 11:32:56 +00:00 |
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OWNERS
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Makes srte owner in call/ and test/
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2018-07-13 08:39:41 +00:00 |
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packet_receiver.h
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Delete unneeded includes of common_types.h and gn deps on webrtc_common.
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2018-11-20 16:28:39 +00:00 |
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rampup_tests.cc
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Include absl/memory/memory.h if absl::make_unique is used
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2019-01-08 20:08:32 +00:00 |
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rampup_tests.h
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Refactor bitrate configuration in CallTest
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2018-11-13 16:03:00 +00:00 |
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receive_time_calculator_unittest.cc
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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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receive_time_calculator.cc
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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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receive_time_calculator.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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rtcp_demuxer_unittest.cc
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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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rtcp_demuxer.cc
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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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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_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_unittest.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_bitrate_configurator.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_bitrate_configurator.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_config.cc
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Adding rtcp report interval into RTCConfiguration.
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2018-11-12 20:00:00 +00:00 |
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rtp_config.h
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Simulcast work to enable RID mux.
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2018-12-21 20:59:23 +00:00 |
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rtp_demuxer_unittest.cc
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Delete unneeded includes of common_types.h and gn deps on webrtc_common.
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2018-11-20 16:28:39 +00:00 |
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rtp_demuxer.cc
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Allow all "token" chars from RFC 4566 when checking for legal mid names.
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2018-08-01 18:20:42 +00:00 |
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rtp_demuxer.h
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Don't check MIDs when demuxing RTP packets in Call
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2018-03-29 20:36:08 +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_unittest.cc
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Send and receive color space information if available
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2018-12-03 21:07:45 +00:00 |
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rtp_payload_params.cc
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Send and receive color space information if available
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2018-12-03 21:07:45 +00:00 |
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rtp_payload_params.h
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Delete unneeded includes of common_types.h and gn deps on webrtc_common.
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2018-11-20 16:28:39 +00:00 |
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rtp_rtcp_demuxer_helper_unittest.cc
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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.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_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_stream_receiver_controller.cc
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Use absl::make_unique and absl::WrapUnique directly
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2018-07-05 10:59:49 +00:00 |
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rtp_stream_receiver_controller.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_interface.h
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Removing redundant argument for SSRCs from ctor of RtpVideoSender.
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2018-12-21 23:33:39 +00:00 |
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rtp_transport_controller_send.cc
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Include pacing buffer size in congestion window.
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2019-01-09 11:16:58 +00:00 |
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rtp_transport_controller_send.h
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Include pacing buffer size in congestion window.
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2019-01-09 11:16:58 +00:00 |
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rtp_video_sender_interface.h
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Move FecController to RtpVideoSender.
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2018-10-05 14:39:01 +00:00 |
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rtp_video_sender_unittest.cc
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Include absl/memory/memory.h if absl::make_unique is used
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2019-01-08 20:08:32 +00:00 |
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rtp_video_sender.cc
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Add field trial kill switch for packetization overhead subtraction.
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2019-01-09 10:19:22 +00:00 |
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rtp_video_sender.h
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Add field trial kill switch for packetization overhead subtraction.
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2019-01-09 10:19:22 +00:00 |
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rtx_receive_stream_unittest.cc
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Reformat the WebRTC code base
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2018-06-19 14:00:39 +00:00 |
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rtx_receive_stream.cc
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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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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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simulated_network.cc
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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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simulated_network.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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simulated_packet_receiver.h
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Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
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2018-08-22 11:12:40 +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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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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video_receive_stream.cc
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New statistic added to VideoReceiveStream to determine latency to first decode.
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2018-12-10 18:49:34 +00:00 |
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video_receive_stream.h
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Remove unused VideoReceiveStream::Config::AddRtxBinding.
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2019-01-09 10:16:12 +00:00 |
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video_send_stream.cc
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Delete pre_encode_callback from VideoSendStream::Config
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2019-01-02 09:14:32 +00:00 |
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video_send_stream.h
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Delete pre_encode_callback from VideoSendStream::Config
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2019-01-02 09:14:32 +00:00 |
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