| .. |
|
adaptation
|
Add powerEfficientDecoder and powerEfficientEncoder stats
|
2022-10-19 13:15:31 +00:00 |
|
test
|
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
|
2022-10-10 11:56:52 +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
|
[Stats] Expose totalPacketSendDelay for audio as well.
|
2022-10-27 10:33:16 +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
|
|
|
|
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
|
Update rtc::Event::Wait call sites to use TimeDelta.
|
2022-08-19 10:07:28 +00:00 |
|
BUILD.gn
|
Revert "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-10-17 13:11:34 +00:00 |
|
call.cc
|
Replace Thread::Invoke with Thread::BlockingCall
|
2022-09-09 10:44:17 +00:00 |
|
call.h
|
Delete Call dependency on ProcessThread as unused
|
2022-06-21 08:59:38 +00:00 |
|
call_config.cc
|
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
|
2021-06-01 06:57:31 +00:00 |
|
call_config.h
|
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
|
2022-03-29 10:14:00 +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
|
RtpEncodingParameters::request_resolution patch 2
|
2022-09-29 14:10:44 +00:00 |
|
call_unittest.cc
|
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.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 |
|
fake_network_pipe_unittest.cc
|
Revert "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-10-17 13:11:34 +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
|
Replace Thread::Invoke with Thread::BlockingCall
|
2022-09-09 10:44:17 +00:00 |
|
flexfec_receive_stream_impl.h
|
Add SetPayloadType to FlexfecReceiveStream.
|
2022-07-28 21:24:50 +00:00 |
|
flexfec_receive_stream_unittest.cc
|
test: fix flexfec test
|
2022-07-06 10:37:19 +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
|
RtpEncodingParameters::request_resolution patch 2
|
2022-09-29 14:10:44 +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
|
Add SetTransportCc to ReceiveStreamInterface.
|
2022-05-30 14:07:04 +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
|
|
|
|
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_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_stream_receiver_controller.cc
|
Demote RtpStreamReceiverController AddSink/RemoveSink to private
|
2022-07-06 09:31:54 +00:00 |
|
rtp_stream_receiver_controller.h
|
Demote RtpStreamReceiverController AddSink/RemoveSink to private
|
2022-07-06 09:31:54 +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
|
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
|
2022-03-29 10:14:00 +00:00 |
|
rtp_transport_controller_send.cc
|
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
|
2022-10-10 11:56:52 +00:00 |
|
rtp_transport_controller_send.h
|
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
|
2022-10-10 11:56:52 +00:00 |
|
rtp_transport_controller_send_factory.h
|
Remove legacy PacedSender.
|
2022-05-13 20:31:06 +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
|
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
|
2022-10-10 11:56:52 +00:00 |
|
rtp_video_sender.h
|
Have RTPSenderVideoFrameTransformerDelegate use new TQ for HW encoders
|
2022-10-10 09:57:08 +00:00 |
|
rtp_video_sender_interface.h
|
Remove top-level const from parameters in function declarations.
|
2022-01-26 11:05:25 +00:00 |
|
rtp_video_sender_unittest.cc
|
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
|
2022-10-10 11:56:52 +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
|
Revert "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-10-17 13:11:34 +00:00 |
|
simulated_network.h
|
Revert "Add documentation, tests and simplify webrtc::SimulatedNetwork."
|
2022-10-17 13:11:34 +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 (2022-11-02T04:07:27).
|
2022-11-02 05:20:18 +00:00 |
|
version.h
|
Add WebRTC code freshness version string.
|
2020-12-14 16:22:35 +00:00 |
|
video_receive_stream.cc
|
Remove unused VideoReceiveStreamInterface::Config::target_delay_ms field.
|
2022-05-30 09:30:23 +00:00 |
|
video_receive_stream.h
|
Add powerEfficientDecoder and powerEfficientEncoder stats
|
2022-10-19 13:15:31 +00:00 |
|
video_send_stream.cc
|
Change the type of RTCVideoSourceStats.framesPerSecond
|
2021-11-16 11:21:41 +00:00 |
|
video_send_stream.h
|
[Stats] Update totalPacketSendDelay to only cover time in pacer queue.
|
2022-10-26 21:29:20 +00:00 |
4915bf869f