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webrtc_m130/call
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Piotr (Peter) Slatala 179a3923b9 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. 
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.


Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
2018-11-16 19:31:11 +00:00
..
test
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
audio_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_receive_stream.h
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
audio_send_stream.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
audio_send_stream.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
audio_state.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_state.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator_unittest.cc
Adds BitrateAllocation struct to OnBitrateUpdated.
2018-10-25 13:51:11 +00:00
bitrate_allocator.cc
Fixes bug in AudioPriorityBitrateAllocationStrategy field trial.
2018-10-25 13:59:22 +00:00
bitrate_allocator.h
Adds BitrateAllocation struct to OnBitrateUpdated.
2018-10-25 13:51:11 +00:00
bitrate_estimator_tests.cc
Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
2018-11-08 19:10:47 +00:00
BUILD.gn
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
2018-11-16 19:31:11 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Delete VideoReceiveStream::EnableEncodedFrameRecording
2018-10-02 10:31:46 +00:00
call_perf_tests.cc
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +00:00
call_unittest.cc
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
2018-11-16 19:31:11 +00:00
call.cc
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
call.h
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
callfactory.cc
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
2018-10-08 12:19:31 +00:00
callfactory.h
Rename Call::Config to CallConfig, keep old name as alias.
2018-02-14 15:14:39 +00:00
degraded_call.cc
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
degraded_call.h
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Allow FakeNetworkPipe to wake up its processing thread
2018-10-31 15:20:57 +00:00
fake_network_pipe.h
Allow FakeNetworkPipe to wake up its processing thread
2018-10-31 15:20:57 +00:00
flexfec_receive_stream_impl.cc
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
flexfec_receive_stream_impl.h
…
flexfec_receive_stream_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
OWNERS
Makes srte owner in call/ and test/
2018-07-13 08:39:41 +00:00
packet_receiver.h
Delete webrtc::PacketTime and backwards compatibility.
2018-08-17 15:14:03 +00:00
rampup_tests.cc
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +00:00
rampup_tests.h
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +00:00
receive_time_calculator_unittest.cc
New method for precise packet reception time measurement.
2018-10-15 11:25:26 +00:00
receive_time_calculator.cc
New method for precise packet reception time measurement.
2018-10-15 11:25:26 +00:00
receive_time_calculator.h
New method for precise packet reception time measurement.
2018-10-15 11:25:26 +00:00
rtcp_demuxer_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtcp_demuxer.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +00:00
rtcp_demuxer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h
…
rtp_bitrate_configurator_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
rtp_config.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_config.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_demuxer_unittest.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +00:00
rtp_demuxer.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +00:00
rtp_demuxer.h
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Get frame type, width and height from the generic descriptor.
2018-10-17 13:31:09 +00:00
rtp_payload_params.cc
Get frame type, width and height from the generic descriptor.
2018-10-17 13:31:09 +00:00
rtp_payload_params.h
Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
2018-09-26 10:26:43 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtp_stream_receiver_controller.h
…
rtp_transport_controller_send_interface.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_transport_controller_send.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_transport_controller_send.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_video_sender_interface.h
Move FecController to RtpVideoSender.
2018-10-05 14:39:01 +00:00
rtp_video_sender_unittest.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_video_sender.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_video_sender.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
rtx_receive_stream.h
…
simulated_network.cc
Fixing bug in SimulatedNetwork where packets stop.
2018-11-06 19:04:37 +00:00
simulated_network.h
Fixing bug in SimulatedNetwork where packets stop.
2018-11-06 19:04:37 +00:00
simulated_packet_receiver.h
Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
2018-08-22 11:12:40 +00:00
ssrc_binding_observer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
syncable.cc
…
syncable.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_receive_stream.cc
Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
2018-09-28 08:48:02 +00:00
video_receive_stream.h
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
video_send_stream.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
video_send_stream.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00