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webrtc_m130/call
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Erik Språng ab89d02b05 Cleans up code related to early loss detection flag. 
Bug: webrtc:10676
Change-Id: I32dab4d4cfc4e8e09c3611b1484fb4394b7efbad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186563
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32295}
2020-10-02 15:37:39 +00:00
..
adaptation
Reland "[Adaptation] Remove QualityScalerResource when disabled."
2020-08-26 06:33:43 +00:00
test
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +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
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
audio_send_stream.cc
Log audio network adaptor and DSCP in AudioSendStream.
2020-08-13 14:05:46 +00:00
audio_send_stream.h
negotiate RED codec for audio
2020-06-25 06:24:18 +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_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
bitrate_allocator.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
bitrate_allocator.h
Introduce RTC_NO_UNIQUE_ADDRESS.
2020-09-30 09:52:49 +00:00
bitrate_estimator_tests.cc
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
BUILD.gn
Async audio processing API
2020-10-02 12:33:34 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Remove deprecated constant.
2020-04-27 10:32:45 +00:00
call_factory.cc
Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread
2020-06-30 15:38:35 +00:00
call_factory.h
Introduce RTC_NO_UNIQUE_ADDRESS.
2020-09-30 09:52:49 +00:00
call_perf_tests.cc
Change gtest name to allow filtering based on the story name.
2020-09-11 14:11:27 +00:00
call_unittest.cc
[Adaptation] Multi-processor support for injected Resources.
2020-07-02 10:28:11 +00:00
call.cc
Introduce RTC_NO_UNIQUE_ADDRESS.
2020-09-30 09:52:49 +00:00
call.h
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
degraded_call.cc
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
degraded_call.h
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
DEPS
Async audio processing API
2020-10-02 12:33:34 +00:00
fake_network_pipe_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
fake_network_pipe.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
fake_network_pipe.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
flexfec_receive_stream_impl.cc
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
flexfec_receive_stream_impl.h
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
flexfec_receive_stream_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
OWNERS
Add terelius as OWNER in call/
2020-03-23 09:55:34 +00:00
packet_receiver.h
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
rampup_tests.cc
Set up a new rtc::Thread instance per test.
2020-05-15 09:13:02 +00:00
rampup_tests.h
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
receive_time_calculator_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
receive_time_calculator.cc
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
receive_time_calculator.h
Format almost everything.
2019-07-08 13:45:15 +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_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_config.cc
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_config.h
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_demuxer_unittest.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.h
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +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_unittest.cc
Remove framemarking RTP extension.
2020-06-15 11:18:00 +00:00
rtp_payload_params.cc
Propagate active decode targets bitmask into DependencyDescriptor
2020-06-29 12:54:43 +00:00
rtp_payload_params.h
Delete field trial WebRTC-GenericDescriptor
2020-06-03 13:00:30 +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_stream_receiver_controller.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
rtp_stream_receiver_controller.h
Rename CriticalSection to RecursiveCriticalSection.
2020-07-17 09:19:50 +00:00
rtp_transport_controller_send_interface.h
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
rtp_transport_controller_send.cc
Fix flag parsing for lazy pacer starting
2020-09-18 17:07:33 +00:00
rtp_transport_controller_send.h
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
rtp_video_sender_interface.h
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +00:00
rtp_video_sender_unittest.cc
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
rtp_video_sender.cc
Cleans up code related to early loss detection flag.
2020-10-02 15:37:39 +00:00
rtp_video_sender.h
Cleans up code related to early loss detection flag.
2020-10-02 15:37:39 +00:00
rtx_receive_stream_unittest.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.h
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
simulated_network.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +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
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
video_receive_stream.cc
Add commas between codec parameters in VideoReceiveStream logging.
2020-03-09 02:45:34 +00:00
video_receive_stream.h
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00
video_send_stream.cc
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
video_send_stream.h
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00