webrtc/call
Evan Shrubsole bd4a718667 [Adaptation] Make resource most limited if kLimitReached hit
This occurs when a resource causes an adaptation down but the current
adaptations can not be adapted any more. Any further adaptation will result in the status kLimitReached,
and so any resource that adapts down should also be most limited.

Bug: webrtc:11695
Change-Id: Idfdf23f482b1b4a132cec49a9be76adc0aec4361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181586
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31933}
2020-08-14 10:16:03 +00:00
..
adaptation [Adaptation] Make resource most limited if kLimitReached hit 2020-08-14 10:16:03 +00:00
test In call/ replace mock macros with unified MOCK_METHOD macro 2020-05-15 13:36:00 +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 Insert audio frame transformer between depacketizer and decoder. 2020-04-01 08:15:53 +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 [getStats] Implement "media-source" audio levels, fixing Chrome bug. 2019-07-04 08:13:45 +00:00
bitrate_allocator.cc Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
bitrate_allocator.h Converts const methods in BitrateAllocator to non-member functions. 2019-09-25 11:55:13 +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_estimator_tests.cc Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 2020-08-06 11:50:08 +00:00
BUILD.gn Delete callbacks from RtpDemuxer on ssrc binding 2020-07-17 15:41:39 +00:00
call.cc [Adaptation] Multi-processor support for injected Resources. 2020-07-02 10:28:11 +00:00
call.h Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread 2020-06-30 15:38:35 +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 Add SharedModuleThread class to share a module thread across Call instances. 2020-05-25 17:21:56 +00:00
call_perf_tests.cc Migrate call/ to webrtc::Mutex. 2020-07-06 15:48:30 +00:00
call_unittest.cc [Adaptation] Multi-processor support for injected Resources. 2020-07-02 10:28:11 +00:00
degraded_call.cc [Adaptation] Adding adaptation resources from Call. 2020-06-11 12:43:21 +00:00
degraded_call.h [Adaptation] Adding adaptation resources from Call. 2020-06-11 12:43:21 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +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
fake_network_pipe_unittest.cc In call/ replace mock macros with unified MOCK_METHOD macro 2020-05-15 13:36:00 +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
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
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.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
receive_time_calculator_unittest.cc Format almost everything. 2019-07-08 13:45:15 +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_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 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.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_demuxer_unittest.cc 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.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_payload_params_unittest.cc Remove framemarking RTP extension. 2020-06-15 11:18:00 +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_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_transport_controller_send.cc Optionally allows TaskQueuePacedSender to coalesce send events. 2020-05-19 17:23:30 +00:00
rtp_transport_controller_send.h Allow setting a bandwidth cap for relayed connections. 2020-03-26 20:41:46 +00:00
rtp_transport_controller_send_interface.h Insert frame transformer between Encoded and Packetizer. 2020-02-28 07:43:13 +00:00
rtp_video_sender.cc Stop mentioning RTPFragmentationHeader in call/ 2020-08-12 13:59:38 +00:00
rtp_video_sender.h Stop mentioning RTPFragmentationHeader in call/ 2020-08-12 13:59:38 +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 Stop mentioning RTPFragmentationHeader in call/ 2020-08-12 13:59:38 +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
rtx_receive_stream_unittest.cc Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets 2019-12-03 21:10:53 +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_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 Add periodic logging of sync delays. 2020-02-11 09:43:49 +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 Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 2020-08-06 11:50:08 +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 [Adaptation] Adding adaptation resources from Call. 2020-06-11 12:43:21 +00:00