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Per K eb688d6e80 Remove dependency to NetworkStateEstimator from TransportSequenceNumberFeedbackGenerator 
NetworkStateEstimator is not used by WebRTC from receive side.

ReceiveSidesCongestionController::SetTransportOverhead is not needed either since NetworkStateEstimator is removed.
Note, CongestionControlFeedbackGenerator is used with RFC 8888 only and feedback frequency will be refactored in later cl.


Bug: webrtc:42220808
Change-Id: I08980aa19117e1de7a9b7896d05d07715dd9f962
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375460
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43821}
2025-01-29 08:43:47 -08:00
..
adaptation
Replace gunit.h macros with WaitUntil in call/
2025-01-07 02:42:31 -08:00
test
Propagate desicion if RTP packet should be ECT(1) marked to socket
2024-12-19 01:59:49 -08:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Change timestamps type from int64 to Timestamp in MediaReceiverInfo.
2024-10-16 11:02:37 +00:00
audio_send_stream.cc
Remove all references to codec-level transport-cc functions and flags.
2024-11-18 10:20:01 +00:00
audio_send_stream.h
Remove all references to codec-level transport-cc functions and flags.
2024-11-18 10:20:01 +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
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
bitrate_allocator_unittest.cc
Comment unused variables in implemented functions 8\n
2024-10-28 12:05:18 +00:00
bitrate_allocator.cc
Comment unused variables in implemented functions 8\n
2024-10-28 12:05:18 +00:00
bitrate_allocator.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
bitrate_estimator_tests.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
BUILD.gn
Reland "Use PayloadTypePicker for video PT assignment"
2024-12-12 16:37:30 -08:00
call_config.cc
Format /call folder
2025-01-08 02:04:16 -08:00
call_config.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
call_perf_tests.cc
In tests replace AudioProcessingBuilder with BuiltinAudioProcessingBuilder
2024-11-01 12:38:34 +00:00
call_unittest.cc
Cleanup AssociateSendStream for audio.
2025-01-10 05:12:40 -08:00
call.cc
Remove dependency to NetworkStateEstimator from TransportSequenceNumberFeedbackGenerator
2025-01-29 08:43:47 -08:00
call.h
Verify that transport-cc is used when RFC8888 field trial is off.
2024-11-19 13:27:58 +00:00
DEPS
Reland "Use PayloadTypePicker for video PT assignment"
2024-12-12 16:37:30 -08:00
fake_network_pipe_unittest.cc
Comment unused variables in implemented functions 9\n
2024-10-29 17:23:30 +00:00
fake_network_pipe.cc
Comment unused variables in implemented functions 9\n
2024-10-29 17:23:30 +00:00
fake_network_pipe.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
fake_payload_type_suggester.h
Reland "Use PayloadTypePicker for video PT assignment"
2024-12-12 16:37:30 -08:00
flexfec_receive_stream_impl.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream_impl.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream_unittest.cc
Fix some clang-tidy issues in call/
2025-01-09 00:30:36 -08:00
flexfec_receive_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
flexfec_receive_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
OWNERS
Make perkj owner of call/
2024-05-07 15:13:29 +00:00
packet_receiver.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
payload_type_picker_unittest.cc
Reland "Use PayloadTypePicker for video PT assignment"
2024-12-12 16:37:30 -08:00
payload_type_picker.cc
Fix some clang-tidy issues in call/
2025-01-09 00:30:36 -08:00
payload_type_picker.h
Guard against redefining a PT within a single codec list.
2024-10-28 12:16:13 +00:00
payload_type.h
Reland "Use PayloadTypePicker for video PT assignment"
2024-12-12 16:37:30 -08:00
rampup_tests.cc
Comment unused variables in implemented functions 9\n
2024-10-29 17:23:30 +00:00
rampup_tests.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
receive_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
receive_time_calculator_unittest.cc
Comment unused variables in implemented functions 9\n
2024-10-29 17:23:30 +00:00
receive_time_calculator.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_bitrate_configurator_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_bitrate_configurator.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_bitrate_configurator.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
rtp_config.cc
Reland "Allow sending to separate payload types for each simulcast index."
2025-01-22 07:10:46 -08:00
rtp_config.h
Reland "Allow sending to separate payload types for each simulcast index."
2025-01-22 07:10:46 -08:00
rtp_demuxer_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_demuxer.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_demuxer.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +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
Fix some clang-tidy issues in call/
2025-01-09 00:30:36 -08:00
rtp_payload_params.cc
Fix some clang-tidy issues in call/
2025-01-09 00:30:36 -08:00
rtp_payload_params.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_stream_receiver_controller_interface.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_stream_receiver_controller.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_stream_receiver_controller.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_transport_config.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +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_factory.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_transport_controller_send_interface.h
Propagate desicion if RTP packet should be ECT(1) marked to socket
2024-12-19 01:59:49 -08:00
rtp_transport_controller_send.cc
Propagate desicion if RTP packet should be ECT(1) marked to socket
2024-12-19 01:59:49 -08:00
rtp_transport_controller_send.h
Propagate desicion if RTP packet should be ECT(1) marked to socket
2024-12-19 01:59:49 -08:00
rtp_video_sender_interface.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtp_video_sender_unittest.cc
Reland "Allow sending to separate payload types for each simulcast index."
2025-01-22 07:10:46 -08:00
rtp_video_sender.cc
Reland "Allow sending to separate payload types for each simulcast index."
2025-01-22 07:10:46 -08:00
rtp_video_sender.h
Reland "Allow sending to separate payload types for each simulcast index."
2025-01-22 07:10:46 -08:00
rtx_receive_stream_unittest.cc
Fix some clang-tidy issues in call/
2025-01-09 00:30:36 -08:00
rtx_receive_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
rtx_receive_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
simulated_network.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
simulated_packet_receiver.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
version.cc
Update WebRTC code version (2025-01-29T04:03:12).
2025-01-28 22:19:21 -08:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
video_receive_stream.h
Change timestamps type from int64 to Timestamp in MediaReceiverInfo.
2024-10-16 11:02:37 +00:00
video_send_stream.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
video_send_stream.h
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00