| .. |
|
adaptation
|
Delete rtc::TaskQueue
|
2024-02-28 10:22:49 +00:00 |
|
test
|
Add missing header
|
2024-04-19 06:13:16 +00:00 |
|
audio_receive_stream.cc
|
Rename AudioReceiveStream to AudioReceiveStreamInterface
|
2022-05-23 08:44:26 +00:00 |
|
audio_receive_stream.h
|
Add Rust_setIncomingAudioMuted
|
2023-09-27 12:16:54 -04:00 |
|
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
|
Merge remote-tracking branch 'google/branch-heads/6478'
|
2024-06-21 16:31:45 -07:00 |
|
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
|
Move webrtc::AudioDeviceModule include to api/ folder
|
2024-04-22 08:56:31 +00:00 |
|
bitrate_allocator.cc
|
Cleanup merge differences from upstream
|
2023-01-25 17:17:55 -08:00 |
|
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
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +00:00 |
|
BUILD.gn
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +00:00 |
|
call.cc
|
Merge remote-tracking branch 'google/branch-heads/6478'
|
2024-06-21 16:31:45 -07:00 |
|
call.h
|
Merge branch m122
|
2024-02-14 22:44:28 -08:00 |
|
call_config.cc
|
Use Environment in RtpTransportyControllerSend
|
2023-12-20 14:47:51 +00:00 |
|
call_config.h
|
Reland "FrameCadenceAdapter: align video encoding to metronome"
|
2024-01-08 13:54:56 +00:00 |
|
call_perf_tests.cc
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +00:00 |
|
call_unittest.cc
|
Pass Environment instead of clock to Fake video encoders at construction
|
2024-04-12 07:42:48 +00:00 |
|
create_call.cc
|
Pass Clock through Environment when constructing Call
|
2023-12-06 19:13:39 +00:00 |
|
create_call.h
|
Delete CallFactoryInterface as no longer needed
|
2023-12-05 15:44:43 +00:00 |
|
degraded_call.cc
|
Remove internal overrides using old SendRtp and SendRtcp interfaces.
|
2023-08-15 13:20:21 +00:00 |
|
degraded_call.h
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +00:00 |
|
DEPS
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +00:00 |
|
fake_network_pipe.cc
|
Delete unused constructor of FakeNetworkPipe
|
2023-08-18 13:07:10 +00:00 |
|
fake_network_pipe.h
|
Delete unused constructor of FakeNetworkPipe
|
2023-08-18 13:07:10 +00:00 |
|
fake_network_pipe_unittest.cc
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +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
|
stats: implement flexfec fecBytesReceived stats for FlexFEC
|
2023-06-21 13:04:31 +00:00 |
|
flexfec_receive_stream_impl.cc
|
stats: implement flexfec fecBytesReceived stats for FlexFEC
|
2023-06-21 13:04:31 +00:00 |
|
flexfec_receive_stream_impl.h
|
stats: implement flexfec fecBytesReceived stats for FlexFEC
|
2023-06-21 13:04:31 +00:00 |
|
flexfec_receive_stream_unittest.cc
|
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
|
2023-05-03 11:09:26 +00:00 |
|
OWNERS
|
Make perkj owner of call/
|
2024-05-07 15:13:29 +00:00 |
|
packet_receiver.h
|
Dont create RTX receive stream before media SSRC is known
|
2024-02-22 14:40:43 +00:00 |
|
rampup_tests.cc
|
In EncoderStreamFactory pass field trials as required parameter
|
2024-04-17 12:53:30 +00:00 |
|
rampup_tests.h
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +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
|
Update to 5005 (M102) (#86)
|
2022-08-24 11:07:33 -04:00 |
|
receive_time_calculator.h
|
Update to 5005 (M102) (#86)
|
2022-08-24 11:07:33 -04:00 |
|
receive_time_calculator_unittest.cc
|
Update to 4896 (M100) (#72)
|
2022-04-15 17:13:23 -06:00 |
|
rtp_bitrate_configurator.cc
|
Update to WebRTC 4103 (M83) (#12)
|
2020-06-25 11:14:34 -07:00 |
|
rtp_bitrate_configurator.h
|
Update to 4896 (M100) (#72)
|
2022-04-15 17:13:23 -06:00 |
|
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
|
Update to 4896 (M100) (#72)
|
2022-04-15 17:13:23 -06:00 |
|
rtp_config.h
|
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
|
2024-03-19 10:03:36 +00:00 |
|
rtp_demuxer.cc
|
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
|
2023-11-08 10:24:10 +00:00 |
|
rtp_demuxer.h
|
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
|
2023-11-08 10:24:10 +00:00 |
|
rtp_demuxer_unittest.cc
|
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
|
2023-05-03 11:09:26 +00:00 |
|
rtp_packet_sink_interface.h
|
|
|
|
rtp_payload_params.cc
|
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
|
2024-03-19 10:03:36 +00:00 |
|
rtp_payload_params.h
|
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
|
2024-03-19 10:03:36 +00:00 |
|
rtp_payload_params_unittest.cc
|
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
|
2024-03-19 10:03:36 +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 PeerConnectionInterface::ReconfigureBandwidthEstimation
|
2024-02-07 14:10:02 +00:00 |
|
rtp_transport_controller_send.cc
|
Add option to provide Environment for CongestionConroller construction
|
2024-05-08 12:46:23 +00:00 |
|
rtp_transport_controller_send.h
|
PacketRouter directly notify RtpTransportControllerSender when sending
|
2024-03-28 09:27:43 +00:00 |
|
rtp_transport_controller_send_factory.h
|
Use Environment in RtpTransportyControllerSend
|
2023-12-20 14:47:51 +00:00 |
|
rtp_transport_controller_send_factory_interface.h
|
Use Environment in RtpTransportyControllerSend
|
2023-12-20 14:47:51 +00:00 |
|
rtp_transport_controller_send_interface.h
|
PacketRouter directly notify RtpTransportControllerSender when sending
|
2024-03-28 09:27:43 +00:00 |
|
rtp_video_sender.cc
|
Merge remote-tracking branch 'google/branch-heads/6478'
|
2024-06-21 16:31:45 -07:00 |
|
rtp_video_sender.h
|
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
|
2024-03-19 10:03:36 +00:00 |
|
rtp_video_sender_interface.h
|
Refactor RtpVideoSender::SetActiveModules.
|
2024-01-26 10:34:46 +00:00 |
|
rtp_video_sender_unittest.cc
|
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
|
2024-03-19 10:03:36 +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.h
|
Move call/simulated_network to test/network
|
2024-04-29 09:55:06 +00:00 |
|
simulated_packet_receiver.h
|
Calculate next process time in simulated network.
|
2019-02-08 19:33:17 +00:00 |
|
syncable.cc
|
|
|
|
syncable.h
|
Rename AudioReceiveStream to AudioReceiveStreamInterface
|
2022-05-23 08:44:26 +00:00 |
|
version.cc
|
Update WebRTC code version (2024-05-13T04:02:33).
|
2024-05-13 05:08:27 +00:00 |
|
version.h
|
Update to WebRTC 4389 (e7d9f74)
|
2021-04-16 13:26:31 -07:00 |
|
video_receive_stream.cc
|
Add missing comma in VideoReceiveStreamInterface::Stats::ToString
|
2023-10-17 10:42:06 +00:00 |
|
video_receive_stream.h
|
stats: implement remote-outbound-rtp for video
|
2024-04-15 15:10:54 +00:00 |
|
video_send_stream.cc
|
Cleanup usasge of ReportBlockData::report_block accessor
|
2023-05-05 09:56:30 +00:00 |
|
video_send_stream.h
|
Remove VideoSendStream::StartPerRtpStream
|
2024-01-26 09:19:50 +00:00 |
