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webrtc/call
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Per K 33cc83595a Ignore allocated bitrate during initial exponential BWE. 
The reason why we want to do this is  because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.

ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
continue up to the max configured bitrate, regardless of of the max
allocated bitrate.

Bug: webrtc:14928
Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42049}
2024-04-12 09:13:44 +00:00
..
adaptation Delete rtc::TaskQueue 2024-02-28 10:22:49 +00:00
test PacketRouter directly notify RtpTransportControllerSender when sending 2024-03-28 09:27:43 +00:00
audio_receive_stream.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
audio_receive_stream.h Propagate time of the last received packet with Timestamp type 2023-06-02 14:29:19 +00:00
audio_send_stream.cc Reland "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-06 14:26:55 +00:00
audio_send_stream.h Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +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 Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h Async audio processing API 2020-10-02 12:33:34 +00:00
bitrate_allocator.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 2022-03-09 13:23:21 +00:00
bitrate_allocator.h Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +00:00
bitrate_allocator_unittest.cc Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
bitrate_estimator_tests.cc Update CallTests to create Call using Environment 2023-12-01 13:16:41 +00:00
BUILD.gn Remove unnecessary RtcEventLog parameter in RtpTransportControllerSend::CreateRtpVideoSender 2024-03-06 16:24:06 +00:00
call.cc Propagate Environment into VideoStreamEncoder 2024-03-05 09:33:02 +00:00
call.h Pass Clock through Environment when constructing Call 2023-12-06 19:13:39 +00:00
call_config.cc Use Environment in RtpTransportyControllerSend 2023-12-20 14:47:51 +00:00
call_config.h Reland "FrameCadenceAdapter: align video encoding to metronome" 2024-01-08 13:54:56 +00:00
call_perf_tests.cc Update expectation on first Sinkwants in CallPerfTest.ReceivesCpuOveruseAndUnderuse 2024-03-22 13:30:46 +00:00
call_unittest.cc Pass Environment instead of clock to Fake video encoders at construction 2024-04-12 07:42:48 +00:00
create_call.cc Pass Clock through Environment when constructing Call 2023-12-06 19:13:39 +00:00
create_call.h Delete CallFactoryInterface as no longer needed 2023-12-05 15:44:43 +00:00
degraded_call.cc Remove internal overrides using old SendRtp and SendRtcp interfaces. 2023-08-15 13:20:21 +00:00
degraded_call.h Delete rtc::TaskQueue 2024-02-28 10:22:49 +00:00
DEPS SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +00:00
fake_network_pipe.cc Delete unused constructor of FakeNetworkPipe 2023-08-18 13:07:10 +00:00
fake_network_pipe.h Delete unused constructor of FakeNetworkPipe 2023-08-18 13:07:10 +00:00
fake_network_pipe_unittest.cc Reland "Delete PacketReceiver::DeliverPacket from all implementations" 2023-01-25 18:18:29 +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 stats: implement flexfec fecBytesReceived stats for FlexFEC 2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.cc stats: implement flexfec fecBytesReceived stats for FlexFEC 2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.h stats: implement flexfec fecBytesReceived stats for FlexFEC 2023-06-21 13:04:31 +00:00
flexfec_receive_stream_unittest.cc Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
OWNERS Update OWNERS for call/ 2022-06-03 12:01:46 +00:00
packet_receiver.h Dont create RTX receive stream before media SSRC is known 2024-02-22 14:40:43 +00:00
rampup_tests.cc Update CallTests to create Call using Environment 2023-12-01 13:16:41 +00:00
rampup_tests.h Stop overriding extensions in rampup tests 2023-01-25 13:18:49 +00:00
receive_stream.h Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
receive_time_calculator.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
receive_time_calculator.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
receive_time_calculator_unittest.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 2022-03-09 22:17:52 +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 Remove RTC_DISALLOW_COPY_AND_ASSIGN more. 2022-01-20 11:00:18 +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_config.cc Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED 2021-11-15 21:44:59 +00:00
rtp_config.h Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
rtp_demuxer.cc Remove SSRCs from libSRTP when removing them from the rtp_demuxer 2023-11-08 10:24:10 +00:00
rtp_demuxer.h Remove SSRCs from libSRTP when removing them from the rtp_demuxer 2023-11-08 10:24:10 +00:00
rtp_demuxer_unittest.cc Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
rtp_packet_sink_interface.h
rtp_payload_params.cc Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
rtp_payload_params.h Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
rtp_payload_params_unittest.cc Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
rtp_stream_receiver_controller.cc Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived 2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller.h Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived 2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller_interface.h Demote RtpStreamReceiverController AddSink/RemoveSink to private 2022-07-06 09:31:54 +00:00
rtp_transport_config.h Add PeerConnectionInterface::ReconfigureBandwidthEstimation 2024-02-07 14:10:02 +00:00
rtp_transport_controller_send.cc Ignore allocated bitrate during initial exponential BWE. 2024-04-12 09:13:44 +00:00
rtp_transport_controller_send.h PacketRouter directly notify RtpTransportControllerSender when sending 2024-03-28 09:27:43 +00:00
rtp_transport_controller_send_factory.h Use Environment in RtpTransportyControllerSend 2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_factory_interface.h Use Environment in RtpTransportyControllerSend 2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_interface.h PacketRouter directly notify RtpTransportControllerSender when sending 2024-03-28 09:27:43 +00:00
rtp_video_sender.cc PacketRouter directly notify RtpTransportControllerSender when sending 2024-03-28 09:27:43 +00:00
rtp_video_sender.h Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
rtp_video_sender_interface.h Refactor RtpVideoSender::SetActiveModules. 2024-01-26 10:34:46 +00:00
rtp_video_sender_unittest.cc Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled. 2024-03-19 10:03:36 +00:00
rtx_receive_stream.cc Updated associated payload types without recreating receive streams. 2022-08-16 13:38:24 +00:00
rtx_receive_stream.h Updated associated payload types without recreating receive streams. 2022-08-16 13:38:24 +00:00
rtx_receive_stream_unittest.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
simulated_network.cc Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." 2022-11-06 13:14:26 +00:00
simulated_network.h Export webrtc::SimulatedNetwork for Chrome component builds 2023-11-27 16:03:23 +00:00
simulated_network_unittest.cc Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." 2022-11-06 13:14:26 +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 Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
version.cc Update WebRTC code version (2024-04-12T04:14:25). 2024-04-12 05:38:06 +00:00
version.h Add WebRTC code freshness version string. 2020-12-14 16:22:35 +00:00
video_receive_stream.cc Add missing comma in VideoReceiveStreamInterface::Stats::ToString 2023-10-17 10:42:06 +00:00
video_receive_stream.h Remove default "unknown" encoderImplementation/decoderImplementation 2023-06-22 11:49:58 +00:00
video_send_stream.cc Cleanup usasge of ReportBlockData::report_block accessor 2023-05-05 09:56:30 +00:00
video_send_stream.h Remove VideoSendStream::StartPerRtpStream 2024-01-26 09:19:50 +00:00