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webrtc_m130/call
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Olga Sharonova be5c7135f9 Revert "Remove use of ReceiveStreamRtpConfig:transport_cc" 
This reverts commit 97ba853295578975a04fc504315cccd465f9f0bd.

Reason for revert: Suspected in breaking WebRTC into Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks

Original change's description:
> Remove use of ReceiveStreamRtpConfig:transport_cc
>
> With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
> I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
>
>
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
>
> Bug: webrtc:14802
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38980}

Bug: webrtc:14802
Change-Id: I2b04274466a5a81d767a48ff2e001b0a04f7f541
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288943
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Auto-Submit: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38988}
2023-01-03 16:18:08 +00:00
..
adaptation
Add powerEfficientDecoder and powerEfficientEncoder stats
2022-10-19 13:15:31 +00:00
test
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
audio: make packets lost a signed integer
2022-11-01 11:46:49 +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
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
audio_sender.h
…
audio_state.cc
…
audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +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_estimator_tests.cc
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +00:00
BUILD.gn
Add scalability mode to RTCOutboundRtpStreamStats stats
2022-12-08 11:46:06 +00:00
call_config.cc
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
call_config.h
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
call_factory.cc
Remove CoDel from webrtc::SimulatedNetwork.
2022-09-08 06:51:05 +00:00
call_factory.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
call_perf_tests.cc
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
call_unittest.cc
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
call.cc
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
call.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
degraded_call.cc
Add ability to set bitrate of DegradedCall via PeerConnection::SetBitrate
2022-10-19 14:09:07 +00:00
degraded_call.h
Add ability to set bitrate of DegradedCall via PeerConnection::SetBitrate
2022-10-19 14:09:07 +00:00
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
fake_network_pipe_unittest.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
fake_network_pipe.cc
Delete some unneeded references to ProcessThread.
2022-01-03 15:36:02 +00:00
fake_network_pipe.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
flexfec_receive_stream_impl.cc
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
flexfec_receive_stream_impl.h
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
flexfec_receive_stream_unittest.cc
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
flexfec_receive_stream.cc
…
flexfec_receive_stream.h
Add SetPayloadType to FlexfecReceiveStream.
2022-07-28 21:24:50 +00:00
OWNERS
Update OWNERS for call/
2022-06-03 12:01:46 +00:00
packet_receiver.h
Remove DeliverPacketAsync.
2021-05-29 07:37:33 +00:00
rampup_tests.cc
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
rampup_tests.h
Migrate call_perf_tests to new perf metrics export API
2022-09-26 13:02:40 +00:00
receive_stream.h
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +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
rtp_bitrate_configurator_unittest.cc
…
rtp_bitrate_configurator.cc
…
rtp_bitrate_configurator.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +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
Update old TODO comments
2022-07-05 09:59:33 +00:00
rtp_demuxer_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_payload_params.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_payload_params.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_stream_receiver_controller_interface.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +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_transport_config.h
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
rtp_transport_controller_send_factory_interface.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
rtp_transport_controller_send_factory.h
Refactor some config plumbing in call/.
2022-11-16 09:18:40 +00:00
rtp_transport_controller_send_interface.h
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
2022-10-10 11:56:52 +00:00
rtp_transport_controller_send.cc
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
rtp_transport_controller_send.h
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
rtp_video_sender_interface.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +00:00
rtp_video_sender_unittest.cc
Remove pending packets from the pacer when an RTP module is removed.
2022-12-13 11:32:58 +00:00
rtp_video_sender.cc
Remove pending packets from the pacer when an RTP module is removed.
2022-12-13 11:32:58 +00:00
rtp_video_sender.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +00:00
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +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
simulated_network_unittest.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.h
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_packet_receiver.h
…
syncable.cc
…
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2023-01-03T04:11:28).
2023-01-03 05:58:45 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
video_receive_stream.h
Reland "Report total and squared inter frame delays measured in OnRenderedFrame"
2022-12-20 14:24:34 +00:00
video_send_stream.cc
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
video_send_stream.h
Add scalability mode to RTCOutboundRtpStreamStats stats
2022-12-08 11:46:06 +00:00