Sebastian Jansson 68ee4653ef Moving SetPacingFactor and allocation limits to SSCC. ...

This CL adds methods to the SendSideCongestionController (SSCC)
interface for configuring pacing factor and allocation based data rate limits.
This means that old SSCC implement the same interface as the new, task
queue based SSCC. This also allows merging the max total allocated
bit rate into SetAllocatedSendBitrateLimits.

This is done in preparation for an upcoming CL where the SSCC version
is controlled by a field trial.

Bug: webrtc:8415
Change-Id: I4d5446a3bedd5b0c725dbd009fb75815fd661eff
Reviewed-on: https://webrtc-review.googlesource.com/61320
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22408}

2018-03-13 16:58:21 +00:00

..

test

Removing AvailableBandwidth method on transport controller.

2018-03-12 15:53:49 +00:00

audio_receive_stream.h

Delete AudioMonitor and related code.

2018-01-30 09:48:29 +00:00

audio_send_stream.cc

Use rtc::ToString instead of std::to_string

2018-01-29 19:26:09 +00:00

audio_send_stream.h

Move remaining traces of VoiceEngine

2018-01-17 13:27:47 +00:00

audio_state.h

Delete AudioMonitor and related code.

2018-01-30 09:48:29 +00:00

bitrate_allocator_unittest.cc

Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1.

2018-02-28 15:38:15 +00:00

bitrate_allocator.cc

Add a couple of logs.

2018-03-05 16:42:02 +00:00

bitrate_allocator.h

Enables configuration of transmission max bitrate multiplier and fec protection level.

2018-03-02 18:05:29 +00:00

bitrate_constraints.cc

Reland "Moved BitrateConfig out of Call::Config."

2018-02-21 11:38:42 +00:00

bitrate_constraints.h

Reland "Moved BitrateConfig out of Call::Config."

2018-02-21 11:38:42 +00:00

bitrate_estimator_tests.cc

Reland "Add multiplex case to webrtc_perf_tests"

2018-03-10 01:21:04 +00:00

BUILD.gn

Separate test/fake_audio_device on API and implementation. Step 2.

2018-03-12 16:14:39 +00:00

call_perf_tests.cc

Separate test/fake_audio_device on API and implementation. Step 2.

2018-03-12 16:14:39 +00:00

call_unittest.cc

Removed fake rtp transport controller send.

2018-02-21 13:39:16 +00:00

call.cc

Removing AvailableBandwidth method on transport controller.

2018-03-12 15:53:49 +00:00

call.h

Moved bitrate config interface from Call class.

2018-02-21 15:03:45 +00:00

callfactory.cc

Rename Call::Config to CallConfig, keep old name as alias.

2018-02-14 15:14:39 +00:00

callfactory.h

Rename Call::Config to CallConfig, keep old name as alias.

2018-02-14 15:14:39 +00:00

DEPS

Make fec controller plug-able.

2018-01-22 11:48:16 +00:00

flexfec_receive_stream_impl.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +00:00

flexfec_receive_stream_impl.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

flexfec_receive_stream_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

flexfec_receive_stream.h

Fix circular dependencies in webrtc_common.

2017-12-15 14:33:26 +00:00

OWNERS

Added nisse@webrtc.org as owner in call.

2018-02-20 09:39:51 +00:00

rampup_tests.cc

Added some margin to ramp down target in perf test.

2018-02-23 12:15:15 +00:00

rampup_tests.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcp_demuxer_unittest.cc

Fix circular dependencies in webrtc_common.

2017-12-15 14:33:26 +00:00

rtcp_demuxer.cc

Fix circular dependencies in webrtc_common.

2017-12-15 14:33:26 +00:00

rtcp_demuxer.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtcp_packet_sink_interface.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_bitrate_configurator_unittest.cc

Extracted bitrate configuration from call class.

2018-02-21 12:33:02 +00:00

rtp_bitrate_configurator.cc

Extracted bitrate configuration from call class.

2018-02-21 12:33:02 +00:00

rtp_bitrate_configurator.h

Moved bitrate config interface from Call class.

2018-02-21 15:03:45 +00:00

rtp_config.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_config.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_demuxer_unittest.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

rtp_demuxer.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +00:00

rtp_demuxer.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_packet_sink_interface.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_rtcp_demuxer_helper_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_rtcp_demuxer_helper.cc

Optional: Use nullopt and implicit construction in /call

2018-01-11 12:00:00 +00:00

rtp_rtcp_demuxer_helper.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_stream_receiver_controller_interface.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_stream_receiver_controller.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +00:00

rtp_stream_receiver_controller.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_transport_controller_send_interface.h

Removing AvailableBandwidth method on transport controller.

2018-03-12 15:53:49 +00:00

rtp_transport_controller_send.cc

Moving SetPacingFactor and allocation limits to SSCC.

2018-03-13 16:58:21 +00:00

rtp_transport_controller_send.h

Removing AvailableBandwidth method on transport controller.

2018-03-12 15:53:49 +00:00

rtx_receive_stream_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtx_receive_stream.cc

Stop using LOG macros in favor of RTC_ prefixed macros.

2017-11-09 11:56:32 +00:00

rtx_receive_stream.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

ssrc_binding_observer.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

syncable.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

syncable.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

video_config.cc

Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."

2018-03-01 18:07:29 +00:00

video_config.h

Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."

2018-03-01 18:07:29 +00:00

video_receive_stream.cc

Delete member VideoReceiveStream::Config::Rtp::ulpfec.

2017-09-26 09:49:21 +00:00

video_receive_stream.h

Move videosinkinterface.h to common_video to solve a circular dep.

2018-01-04 13:19:49 +00:00

video_send_stream.cc

Make RTCP report interval configurable

2018-02-01 10:12:11 +00:00

video_send_stream.h

Reland "Add hugeFramesSent GetStats metric"

2018-03-06 13:38:11 +00:00