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adaptation
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Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
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2023-05-03 11:09:26 +00:00 |
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test
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Revert "Clean up last_packet_received_time_ as it's no longer used."
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2023-09-25 08:49:53 +00:00 |
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audio_receive_stream.cc
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Rename AudioReceiveStream to AudioReceiveStreamInterface
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2022-05-23 08:44:26 +00:00 |
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audio_receive_stream.h
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Propagate time of the last received packet with Timestamp type
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2023-06-02 14:29:19 +00:00 |
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audio_send_stream.cc
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Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
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2021-09-06 14:26:55 +00:00 |
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audio_send_stream.h
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Replace "rcvd" with "received" for readability
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2023-04-24 15:30:07 +00:00 |
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audio_sender.h
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Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
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2020-01-13 18:31:30 +00:00 |
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audio_state.cc
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Remove chromium clang style errors affecting sdk/android/media_jni
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2018-04-09 13:55:49 +00:00 |
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audio_state.h
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Async audio processing API
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2020-10-02 12:33:34 +00:00 |
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bitrate_allocator.cc
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WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
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2022-03-09 13:23:21 +00:00 |
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bitrate_allocator.h
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Use backticks not vertical bars to denote variables in comments for /call
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2021-07-27 18:29:33 +00:00 |
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bitrate_allocator_unittest.cc
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Adopt absl::string_view in call/
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2022-05-17 12:00:45 +00:00 |
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bitrate_estimator_tests.cc
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Reland: FrameGeneratorCapturer: don't generate video before Start is called
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2023-06-29 14:47:05 +00:00 |
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BUILD.gn
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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call.cc
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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call.h
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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call_config.cc
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Add parameter to control the pacer's burst outside of field trials.
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2022-11-15 08:46:30 +00:00 |
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call_config.h
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Introduce support for video packet batching.
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2023-05-08 16:24:03 +00:00 |
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call_factory.cc
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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call_factory.h
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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call_perf_tests.cc
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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call_unittest.cc
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Cleanup Call construction
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2023-10-16 06:34:26 +00:00 |
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degraded_call.cc
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Remove internal overrides using old SendRtp and SendRtcp interfaces.
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2023-08-15 13:20:21 +00:00 |
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degraded_call.h
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Remove internal overrides using old SendRtp and SendRtcp interfaces.
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2023-08-15 13:20:21 +00:00 |
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DEPS
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SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
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2021-08-30 10:20:55 +00:00 |
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fake_network_pipe.cc
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Delete unused constructor of FakeNetworkPipe
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2023-08-18 13:07:10 +00:00 |
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fake_network_pipe.h
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Delete unused constructor of FakeNetworkPipe
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2023-08-18 13:07:10 +00:00 |
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fake_network_pipe_unittest.cc
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Reland "Delete PacketReceiver::DeliverPacket from all implementations"
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2023-01-25 18:18:29 +00:00 |
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flexfec_receive_stream.cc
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[Cleanup] Add missing #include. Remove useless ones.
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2018-10-23 11:32:56 +00:00 |
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flexfec_receive_stream.h
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stats: implement flexfec fecBytesReceived stats for FlexFEC
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2023-06-21 13:04:31 +00:00 |
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flexfec_receive_stream_impl.cc
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stats: implement flexfec fecBytesReceived stats for FlexFEC
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2023-06-21 13:04:31 +00:00 |
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flexfec_receive_stream_impl.h
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stats: implement flexfec fecBytesReceived stats for FlexFEC
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2023-06-21 13:04:31 +00:00 |
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flexfec_receive_stream_unittest.cc
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Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
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2023-05-03 11:09:26 +00:00 |
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OWNERS
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Update OWNERS for call/
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2022-06-03 12:01:46 +00:00 |
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packet_receiver.h
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Allow injecting packets of type Any to Call::DeliverRtpPacket
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2023-03-29 06:36:17 +00:00 |
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rampup_tests.cc
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Replace WebRTC-QuickPerfTest field trial with a flag
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2023-10-10 08:59:10 +00:00 |
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rampup_tests.h
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Stop overriding extensions in rampup tests
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2023-01-25 13:18:49 +00:00 |
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receive_stream.h
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Remove rtp header extension from config of Call audio and video receivers
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2023-01-31 11:58:43 +00:00 |
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receive_time_calculator.cc
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WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
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2022-03-29 10:14:00 +00:00 |
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receive_time_calculator.h
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WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
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2022-03-29 10:14:00 +00:00 |
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receive_time_calculator_unittest.cc
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WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
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2022-03-09 22:17:52 +00:00 |
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rtp_bitrate_configurator.cc
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Allow setting a bandwidth cap for relayed connections.
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2020-03-26 20:41:46 +00:00 |
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rtp_bitrate_configurator.h
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Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
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2022-01-20 11:00:18 +00:00 |
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rtp_bitrate_configurator_unittest.cc
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Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
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2020-01-10 16:39:51 +00:00 |
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rtp_config.cc
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Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
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2021-11-15 21:44:59 +00:00 |
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rtp_config.h
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Introduce support for video packet batching.
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2023-05-08 16:24:03 +00:00 |
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rtp_demuxer.cc
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Adopt absl::string_view in call/
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2022-05-17 12:00:45 +00:00 |
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rtp_demuxer.h
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Adopt absl::string_view in call/
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2022-05-17 12:00:45 +00:00 |
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rtp_demuxer_unittest.cc
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Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
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2023-05-03 11:09:26 +00:00 |
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rtp_packet_sink_interface.h
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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rtp_payload_params.cc
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Add codec name H265 to support H265 in WebRTC
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2023-09-20 09:25:32 +00:00 |
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rtp_payload_params.h
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For VP9 assume max number of spatial layers to simulate generic descriptor
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2022-06-08 11:36:54 +00:00 |
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rtp_payload_params_unittest.cc
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When simulating chains from VP9 codec specific info support first_active_layer > 0
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2023-08-03 13:19:00 +00:00 |
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rtp_stream_receiver_controller.cc
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Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
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2022-12-22 14:04:21 +00:00 |
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rtp_stream_receiver_controller.h
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Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
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2022-12-22 14:04:21 +00:00 |
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rtp_stream_receiver_controller_interface.h
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Demote RtpStreamReceiverController AddSink/RemoveSink to private
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2022-07-06 09:31:54 +00:00 |
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rtp_transport_config.h
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Add parameter to control the pacer's burst outside of field trials.
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2022-11-15 08:46:30 +00:00 |
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rtp_transport_controller_send.cc
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Revert "Clean up last_packet_received_time_ as it's no longer used."
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2023-09-25 08:49:53 +00:00 |
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rtp_transport_controller_send.h
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Revert "Clean up last_packet_received_time_ as it's no longer used."
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2023-09-25 08:49:53 +00:00 |
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rtp_transport_controller_send_factory.h
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Refactor some config plumbing in call/.
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2022-11-16 09:18:40 +00:00 |
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rtp_transport_controller_send_factory_interface.h
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Delete Call dependency on ProcessThread as unused
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2022-06-21 08:59:38 +00:00 |
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rtp_transport_controller_send_interface.h
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Revert "Clean up last_packet_received_time_ as it's no longer used."
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2023-09-25 08:49:53 +00:00 |
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rtp_video_sender.cc
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Move field trial check WebRTC-DisableRtxRateLimiter
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2023-10-11 10:34:17 +00:00 |
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rtp_video_sender.h
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Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
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2023-06-01 07:51:56 +00:00 |
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rtp_video_sender_interface.h
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Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
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2022-12-02 12:03:25 +00:00 |
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rtp_video_sender_unittest.cc
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Rename EncodedImage property Timetamp to RtpTimestamp
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2023-09-24 20:06:48 +00:00 |
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rtx_receive_stream.cc
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Updated associated payload types without recreating receive streams.
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2022-08-16 13:38:24 +00:00 |
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rtx_receive_stream.h
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Updated associated payload types without recreating receive streams.
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2022-08-16 13:38:24 +00:00 |
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rtx_receive_stream_unittest.cc
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Store RtpPacketReceived::arrival_time as Timestamp.
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2021-05-05 16:22:33 +00:00 |
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simulated_network.cc
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Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
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2022-11-06 13:14:26 +00:00 |
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simulated_network.h
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Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
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2022-11-06 13:14:26 +00:00 |
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simulated_network_unittest.cc
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Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
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2022-11-06 13:14:26 +00:00 |
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simulated_packet_receiver.h
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Calculate next process time in simulated network.
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2019-02-08 19:33:17 +00:00 |
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syncable.cc
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Fixing WebRTC after moving from src/webrtc to src/
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2017-09-15 05:02:56 +00:00 |
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syncable.h
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Rename AudioReceiveStream to AudioReceiveStreamInterface
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2022-05-23 08:44:26 +00:00 |
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version.cc
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Update WebRTC code version (2023-10-16T04:06:36).
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2023-10-16 05:19:37 +00:00 |
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version.h
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Add WebRTC code freshness version string.
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2020-12-14 16:22:35 +00:00 |
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video_receive_stream.cc
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Add missing comma in VideoReceiveStreamInterface::Stats::ToString
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2023-10-17 10:42:06 +00:00 |
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video_receive_stream.h
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Remove default "unknown" encoderImplementation/decoderImplementation
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2023-06-22 11:49:58 +00:00 |
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video_send_stream.cc
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Cleanup usasge of ReportBlockData::report_block accessor
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2023-05-05 09:56:30 +00:00 |
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video_send_stream.h
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Remove default "unknown" encoderImplementation/decoderImplementation
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2023-06-22 11:49:58 +00:00 |
c868fd3b3e