mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/call
History
Tommi f6f4543304 Rename VideoReceiveStream to VideoReceiveStreamInterface 
Bug: webrtc:7484
Change-Id: I653cfe46486e0396897dd333069a894d67e3c07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262769
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36958}
2022-05-22 10:54:38 +00:00
..
adaptation Delete TestListener and top-level thread wrapping. 2022-05-20 15:21:21 +00:00
test Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
audio_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling 2022-05-21 23:06:21 +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 Remove typing detection 2022-03-23 10:23:54 +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 Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
BUILD.gn Delete TestListener and top-level thread wrapping. 2022-05-20 15:21:21 +00:00
call.cc Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
call.h Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +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 legacy PacedSender. 2022-05-13 20:31:06 +00:00
call_factory.h Use SequenceChecker from public API 2021-02-10 15:04:55 +00:00
call_perf_tests.cc Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
call_unittest.cc Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
degraded_call.cc Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
degraded_call.h Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +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 Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +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 Rename ReceiveStream to ReceiveStreamInterface 2022-05-18 07:26:50 +00:00
flexfec_receive_stream_impl.cc Reland: Update local_ssrc without needing to recreate video streams. 2022-05-17 10:59:54 +00:00
flexfec_receive_stream_impl.h Rename ReceiveStream to ReceiveStreamInterface 2022-05-18 07:26:50 +00:00
flexfec_receive_stream_unittest.cc Delete TestListener and top-level thread wrapping. 2022-05-20 15:21:21 +00:00
OWNERS Fix WebRTC version updater bot. 2021-11-22 08:44:15 +00:00
packet_receiver.h Remove DeliverPacketAsync. 2021-05-29 07:37:33 +00:00
rampup_tests.cc Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
rampup_tests.h Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
receive_stream.h Rename MediaReceiveStream to MediaReceiveStreamInterface 2022-05-20 13:17:52 +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 Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +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 Simulate generic dependency structure for VP8 2022-05-19 11:53:08 +00:00
rtp_payload_params.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_payload_params_unittest.cc Simulate generic dependency structure for VP8 2022-05-19 11:53:08 +00:00
rtp_stream_receiver_controller.cc Remove lock from RtpStreamReceiverController. 2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h Use SequenceChecker from public API 2021-02-10 15:04:55 +00:00
rtp_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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 Use ProbeClusterConfig in BitrateProber from GoogCC 2022-05-17 12:29:25 +00:00
rtp_transport_controller_send.h Adopt absl::string_view in call/ 2022-05-17 12:00:45 +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 Remove legacy PacedSender. 2022-05-13 20:31:06 +00:00
rtp_transport_controller_send_interface.h Adopt absl::string_view in call/ 2022-05-17 12:00:45 +00:00
rtp_video_sender.cc Simulate generic dependency structure for VP8 2022-05-19 11:53:08 +00:00
rtp_video_sender.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +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 Simulate generic dependency structure for VP8 2022-05-19 11:53:08 +00:00
rtx_receive_stream.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
rtx_receive_stream.h IWYU: uint32_t is defined in cstdint 2020-05-07 17:04:15 +00:00
rtx_receive_stream_unittest.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
simulated_network.cc Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
simulated_network.h Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +00:00
simulated_network_unittest.cc Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +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 VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
version.cc Update WebRTC code version (2022-05-22T04:05:03). 2022-05-22 05:18:28 +00:00
version.h Add WebRTC code freshness version string. 2020-12-14 16:22:35 +00:00
video_receive_stream.cc Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
video_receive_stream.h Rename VideoReceiveStream to VideoReceiveStreamInterface 2022-05-22 10:54:38 +00:00
video_send_stream.cc Change the type of RTCVideoSourceStats.framesPerSecond 2021-11-16 11:21:41 +00:00
video_send_stream.h Remove more top-level const from parameters in function declarations 2022-02-01 09:15:50 +00:00