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webrtc/call
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Artem Titov 8a9f3a8f53 Reland "Remove dependency of video_replay on TestADM." 
This reverts commit f9e3bdd2ce.

Reason for revert: reland with fix

Original change's description:
> Revert "Remove dependency of video_replay on TestADM."
>
> This reverts commit 01716663a9.
>
> Reason for revert:  breaking CallPerfTest
> https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview 
>
> Original change's description:
> > Remove dependency of video_replay on TestADM.
> >
> > This should remove requirement to build TestADM in chromium build.
> >
> > Bug: b/272350185, webrtc:15081
> > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39934}
>
> Bug: b/272350185, webrtc:15081
> Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#39939}

Bug: b/272350185, webrtc:15081
Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:39:22 +00:00
..
adaptation Apply resolution-bitrate limits collected from field trial (cl/294600) for AV1. 2023-03-16 19:04:32 +00:00
test [WebRTC-SendPacketsOnWorkerThread] Cleanup RtpTransportControllerSend 2023-04-17 11:41:15 +00:00
audio_receive_stream.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
audio_receive_stream.h Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +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
audio_state.cc
audio_state.h
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 Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +00:00
BUILD.gn Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +00:00
call.cc Allow injecting packets of type Any to Call::DeliverRtpPacket 2023-03-29 06:36:17 +00:00
call.h Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +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 Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +00:00
call_unittest.cc Allow injecting packets of type Any to Call::DeliverRtpPacket 2023-03-29 06:36:17 +00:00
degraded_call.cc Migrate PostTask+Wait to BlockingCall to avoid deadlock in DegradedCall. 2023-04-19 09:40:33 +00:00
degraded_call.h Reland "Delete PacketReceiver::DeliverPacket from all implementations" 2023-01-25 18:18:29 +00:00
DEPS SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +00:00
fake_network_pipe.cc Reland "Delete PacketReceiver::DeliverPacket from all implementations" 2023-01-25 18:18:29 +00:00
fake_network_pipe.h Reland "Delete PacketReceiver::DeliverPacket from all implementations" 2023-01-25 18:18:29 +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
flexfec_receive_stream.h Add SetPayloadType to FlexfecReceiveStream. 2022-07-28 21:24:50 +00:00
flexfec_receive_stream_impl.cc Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
flexfec_receive_stream_impl.h Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
flexfec_receive_stream_unittest.cc Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
OWNERS Update OWNERS for call/ 2022-06-03 12:01:46 +00:00
packet_receiver.h Allow injecting packets of type Any to Call::DeliverRtpPacket 2023-03-29 06:36:17 +00:00
rampup_tests.cc Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +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
rtp_bitrate_configurator.h Remove RTC_DISALLOW_COPY_AND_ASSIGN more. 2022-01-20 11:00:18 +00:00
rtp_bitrate_configurator_unittest.cc
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.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_demuxer_unittest.cc Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
rtp_packet_sink_interface.h
rtp_payload_params.cc Reland "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)." 2023-02-21 18:30:35 +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_payload_params_unittest.cc Introduce EncodedImage.SimulcastIndex(). 2023-02-15 15:02:57 +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 parameter to control the pacer's burst outside of field trials. 2022-11-15 08:46:30 +00:00
rtp_transport_controller_send.cc [WebRTC-SendPacketsOnWorkerThread] Cleanup RtpTransportControllerSend 2023-04-17 11:41:15 +00:00
rtp_transport_controller_send.h [WebRTC-SendPacketsOnWorkerThread] Cleanup RtpTransportControllerSend 2023-04-17 11:41:15 +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_factory_interface.h Delete Call dependency on ProcessThread as unused 2022-06-21 08:59:38 +00:00
rtp_transport_controller_send_interface.h [WebRTC-SendPacketsOnWorkerThread] Cleanup RtpTransportControllerSend 2023-04-17 11:41:15 +00:00
rtp_video_sender.cc [WebRTC-SendPacketsOnWorkerThread] Cleanup RtpTransportControllerSend 2023-04-17 11:41:15 +00:00
rtp_video_sender.h Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" 2022-12-02 12:03:25 +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 [WebRTC-SendPacketsOnWorkerThread] Cleanup RtpTransportControllerSend 2023-04-17 11:41:15 +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 Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." 2022-11-06 13:14:26 +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
syncable.cc
syncable.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
version.cc Update WebRTC code version (2023-04-24T04:05:22). 2023-04-24 06:02:33 +00:00
version.h
video_receive_stream.cc Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
video_receive_stream.h Allow RTX ssrc to be updated on receive streams 2023-02-01 12:54:46 +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