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webrtc/call
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Per K f6ce1d39ee Allow injecting packets of type Any to Call::DeliverRtpPacket 
MediaType::Any will be used by packets that can not be demuxed by
RtpTransport.

Bug: webrtc:14928
Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39710}
2023-03-29 06:36:17 +00:00
..
adaptation Apply resolution-bitrate limits collected from field trial (cl/294600) for AV1. 2023-03-16 19:04:32 +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 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent 2023-03-16 06:46:19 +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 pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +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 Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
BUILD.gn Allow injecting packets of type Any to Call::DeliverRtpPacket 2023-03-29 06:36:17 +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 Ensure CallTest derived tests per default set min/max audio bitrate. 2023-01-26 17:36:01 +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 Reland "Delete PacketReceiver::DeliverPacket from all implementations" 2023-01-25 18:18:29 +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 [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
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 Stop overriding extensions in rampup tests 2023-01-25 13:18:49 +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 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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Stop Posting tasks when we don't need to. 2023-03-06 15:13:39 +00:00
rtp_transport_controller_send.h Stop Posting tasks when we don't need to. 2023-03-06 15:13:39 +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 Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface 2022-10-10 11:56:52 +00:00
rtp_video_sender.cc Add a field trial string to make enable_retransmit_all_layers configurable. 2023-03-03 15:20:41 +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 Introduce EncodedImage.SimulcastIndex(). 2023-02-15 15:02:57 +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 Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
version.cc Update WebRTC code version (2023-03-29T04:08:24). 2023-03-29 06:05:37 +00:00
version.h Add WebRTC code freshness version string. 2020-12-14 16:22:35 +00:00
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