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webrtc_m130/call
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Erik Språng c96aa30c13 Minor refactoring of RtpVideoSender 
Bug: None
Change-Id: I7cc16d4e435974090310bd3a7cea9eaf0586be8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208761
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33351}
2021-02-26 13:20:52 +00:00
..
adaptation
Use pixels from single active stream if set in CanDecreaseResolutionTo
2021-02-22 10:25:32 +00:00
test
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
audio_receive_stream.cc
…
audio_receive_stream.h
Reduce redundant flags for audio stream playout state.
2021-01-18 09:09:09 +00:00
audio_send_stream.cc
Handle longer AudioSendStream::Config strings
2021-02-10 10:53:29 +00:00
audio_send_stream.h
negotiate RED codec for audio
2020-06-25 06:24:18 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_state.cc
…
audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
bitrate_allocator.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
bitrate_allocator.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
bitrate_estimator_tests.cc
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
BUILD.gn
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
call_config.cc
Inject network thread to Call.
2021-01-31 10:56:14 +00:00
call_config.h
Inject network thread to Call.
2021-01-31 10:56:14 +00:00
call_factory.cc
Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread
2020-06-30 15:38:35 +00:00
call_factory.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
call_perf_tests.cc
Communicate encoder resolutions via rtc::VideoSinkWants.
2021-02-25 11:10:55 +00:00
call_unittest.cc
Move deprecated code to their own build targets.
2021-01-18 13:09:47 +00:00
call.cc
Move Call's histogram reporting code into destructor.
2021-02-18 09:58:54 +00:00
call.h
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
degraded_call.cc
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
degraded_call.h
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
DEPS
Async audio processing API
2020-10-02 12:33:34 +00:00
fake_network_pipe_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
fake_network_pipe.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
fake_network_pipe.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
flexfec_receive_stream_impl.cc
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +00:00
flexfec_receive_stream_impl.h
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
flexfec_receive_stream_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
flexfec_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h
Format almost everything.
2019-07-08 13:45:15 +00:00
OWNERS
Make sprang@ owner in call
2020-10-19 10:30:03 +00:00
packet_receiver.h
Add DeliverPacketAsync method to PacketReceiver.
2021-01-19 11:53:50 +00:00
rampup_tests.cc
Default enable sending transport sequence numbers on audio packets.
2020-11-24 09:19:54 +00:00
rampup_tests.h
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
receive_time_calculator_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
receive_time_calculator.cc
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
receive_time_calculator.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_config.cc
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_config.h
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_demuxer_unittest.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_demuxer.h
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
2020-11-11 14:00:52 +00:00
rtp_payload_params.cc
Ending a statement with a semicolon
2021-02-19 16:53:54 +00:00
rtp_payload_params.h
Ignore frame type when calculating generic frame dependencies.
2020-11-23 10:52:06 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
rtp_transport_controller_send_interface.h
Delete unneeded dependencies on the Module abstraction
2020-12-21 09:09:57 +00:00
rtp_transport_controller_send.cc
Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
2020-11-02 11:05:56 +00:00
rtp_transport_controller_send.h
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
rtp_video_sender_interface.h
Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender.
2020-10-19 11:37:23 +00:00
rtp_video_sender_unittest.cc
Account for extra capacity rtx packet might need
2021-02-06 21:34:08 +00:00
rtp_video_sender.cc
Minor refactoring of RtpVideoSender
2021-02-26 13:20:52 +00:00
rtp_video_sender.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
rtx_receive_stream_unittest.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.h
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
simulated_network.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
syncable.cc
…
syncable.h
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
version.cc
Update WebRTC code version (2021-02-25T04:03:13).
2021-02-25 06:36:17 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Add commas between codec parameters in VideoReceiveStream logging.
2020-03-09 02:45:34 +00:00
video_receive_stream.h
Remove 'secondary sink' concept from webrtc::VideoReceiveStream.
2021-02-15 18:08:17 +00:00
video_send_stream.cc
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
video_send_stream.h
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00