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webrtc/call
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Danil Chapovalov 2ee83c1784 Provide Environment for ReceiveSideConfestionController construction 
Environment includes propagated field trials that can be later passed to
RemoteBitrateEstimators member, and would allow not to rely on the global field trial string

Bug: webrtc:42220378
Change-Id: Icf75a433c20352b2c22829c2148c92f69a2517aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349645
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42242}
2024-05-07 08:02:36 +00:00
..
adaptation Delete rtc::TaskQueue 2024-02-28 10:22:49 +00:00
test Add missing header 2024-04-19 06:13:16 +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 Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +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 Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +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 Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
BUILD.gn Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
call.cc Provide Environment for ReceiveSideConfestionController construction 2024-05-07 08:02:36 +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 Move call/simulated_network to test/network 2024-04-29 09:55:06 +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 Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
DEPS Move call/simulated_network to test/network 2024-04-29 09:55:06 +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 Move call/simulated_network to test/network 2024-04-29 09:55:06 +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 In EncoderStreamFactory pass field trials as required parameter 2024-04-17 12:53:30 +00:00
rampup_tests.h Move call/simulated_network to test/network 2024-04-29 09:55:06 +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 Reland "Ignore allocated bitrate during initial exponential BWE." 2024-04-16 15:34:49 +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.h Move call/simulated_network to test/network 2024-04-29 09:55:06 +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 (2024-05-07T04:06:18). 2024-05-07 05:45:40 +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 stats: implement remote-outbound-rtp for video 2024-04-15 15:10:54 +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