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webrtc_m130/call
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Evan Shrubsole e331a122aa Move quality rampup experiment to overuse module 
Bug: webrtc:11222
Change-Id: I8d0860bfe8bdfe0a051f5a6165cdcfa0cc25cfb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30465}
2020-02-06 08:38:39 +00:00
..
adaptation
Move quality rampup experiment to overuse module
2020-02-06 08:38:39 +00:00
test
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +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
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_send_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_send_stream.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_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
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
bitrate_allocator_unittest.cc
Propagating TargetRate struct to BitrateAllocator.
2019-09-19 14:03:04 +00:00
bitrate_allocator.cc
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_allocator.h
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_estimator_tests.cc
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
2019-12-07 00:54:26 +00:00
BUILD.gn
Reformat GN files.
2020-01-21 12:13:11 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Adds injectable trials from peerconnection down to transport controller.
2019-11-21 12:41:45 +00:00
call_factory.cc
DegradedCall: fake network using TaskQueue instead of ProcessThread
2019-08-06 15:05:30 +00:00
call_factory.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
call_perf_tests.cc
Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
2020-01-07 13:02:52 +00:00
call_unittest.cc
Trials should always be populated in call config.
2019-12-03 10:34:55 +00:00
call.cc
Reduce some logging at INFO level by moving log statements
2019-12-12 21:54:06 +00:00
call.h
Remove MediaTransport from Call.
2019-08-08 10:58:57 +00:00
degraded_call.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
degraded_call.h
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
fake_network_pipe.cc
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
fake_network_pipe.h
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
flexfec_receive_stream_impl.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
flexfec_receive_stream_impl.h
Injecting Clock in video receive.
2019-03-04 21:53:57 +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
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +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
Do not stop SingleThreadedTaskQueueForTestingTest near the end of the tests
2019-10-31 13:07:30 +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 std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
receive_time_calculator.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.h
Remove api/bitrate_constraints.h.
2019-09-18 06:37:58 +00:00
rtp_config.cc
Negotiate use of RTCP loss notification feedback (LNTF)
2019-05-24 12:44:14 +00:00
rtp_config.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_demuxer_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
rtp_demuxer.cc
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +00:00
rtp_demuxer.h
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +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
Pass picture_id from generic packetizer through codec-specific field
2020-01-28 19:26:28 +00:00
rtp_payload_params.cc
Reland "[VP9] Shift spatial layers on RTP level to always start from 0."
2020-02-04 10:06:44 +00:00
rtp_payload_params.h
Revert "Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused"
2020-01-07 19:16:48 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_rtcp_demuxer_helper.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
rtp_transport_controller_send_interface.h
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +00:00
rtp_transport_controller_send.cc
Reland "Adds trial to use correct overhead calculation in pacer."
2020-01-29 18:45:16 +00:00
rtp_transport_controller_send.h
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +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 scenario test for transport wide feedback based retransmission.
2020-01-14 16:00:14 +00:00
rtp_video_sender.cc
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +00:00
rtp_video_sender.h
Reland "Reland "Only include overhead if using send side bandwidth estimation.""
2020-01-29 18:42:34 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
simulated_network_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
simulated_network.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
simulated_network.h
Exposing more features in the network emulation manager API.
2019-12-06 08:47:19 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
ssrc_binding_observer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.h
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
video_receive_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
video_receive_stream.h
VideoReceiveStream: Enable encoded frame sink.
2019-12-03 15:55:04 +00:00
video_send_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
video_send_stream.h
Delete media transport integration.
2019-11-26 19:19:36 +00:00