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35fa280229
webrtc/call
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Sebastian Jansson 35fa280229 Adds allocated rate without feedback to new congestion controller. 
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.

To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.

Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
2018-10-01 07:48:02 +00:00
..
test Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe 2018-08-29 11:05:19 +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 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_send_stream.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
audio_send_stream.h Delete root header file typedef.h. 2018-07-25 14:59:26 +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 allocated rate without feedback to new congestion controller. 2018-10-01 07:48:02 +00:00
bitrate_allocator.cc Adds allocated rate without feedback to new congestion controller. 2018-10-01 07:48:02 +00:00
bitrate_allocator.h Adds allocated rate without feedback to new congestion controller. 2018-10-01 07:48:02 +00:00
bitrate_estimator_tests.cc Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
BUILD.gn Remove runtime_enabled_feature. 2018-10-01 07:03:25 +00:00
call_config.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
call_config.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
call_perf_tests.cc Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
call_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
call.cc Adds allocated rate without feedback to new congestion controller. 2018-10-01 07:48:02 +00:00
call.h Delete struct webrtc::PacketTime. 2018-08-07 10:07:15 +00:00
callfactory.cc Remove temporary SetConfig method from NetworkSimulatioInterface. 2018-08-23 10:08:35 +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 Adds packet_size to rtc::SentPacket in testing code. 2018-09-29 22:06:07 +00:00
degraded_call.h Switch webrtc users from deprecated ctors. 2018-08-17 13:54:51 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe 2018-08-29 11:05:19 +00:00
fake_network_pipe.h Cleanup: remove deprecated class shortcuts. 2018-09-25 09:06:47 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
flexfec_receive_stream.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +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 Use SdpVideoFormat in VideoReceiveStream::Decoder 2018-09-11 15:03:04 +00:00
rampup_tests.h Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe 2018-08-22 11:12:40 +00:00
receive_time_calculator_unittest.cc Added receive time calculator under field trial. 2018-03-21 15:40:39 +00:00
receive_time_calculator.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
receive_time_calculator.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Refactoring PayloadRouter. 2018-07-17 14:46:15 +00:00
rtp_config.h Add shared frame id state to RtpVideoSender. 2018-08-08 15:28:20 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." 2018-09-26 10:26:43 +00:00
rtp_payload_params.cc Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." 2018-09-26 10:26:43 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_transport_controller_send_interface.h Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_transport_controller_send.cc Adds allocated rate without feedback to new congestion controller. 2018-10-01 07:48:02 +00:00
rtp_transport_controller_send.h Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_video_sender_interface.h Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_video_sender_unittest.cc Makes new congestion controller work with rtp sender tests. 2018-09-27 15:58:32 +00:00
rtp_video_sender.cc Disable ulpfec when field trial flag is present 2018-09-25 16:07:59 +00:00
rtp_video_sender.h Add shared frame id state to RtpVideoSender. 2018-08-08 15:28:20 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulated_network.cc Move SimulatedNetwork class to separate file. 2018-08-08 09:29:53 +00:00
simulated_network.h Remove temporary SetConfig method from NetworkSimulatioInterface. 2018-08-23 10:08:35 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
video_send_stream.cc New api function CreateVideoStreamEncoder. 2018-07-24 09:14:26 +00:00
video_send_stream.h Delete VideoSendStream::EnableEncodedFrameRecording. 2018-09-03 13:06:32 +00:00