webrtc/call
Guido Urdaneta 61b2993aaf Revert "[Adaptation] Remove QualityScalerResource when disabled."
This reverts commit ba8abbb630.

Reason for revert: Suspect of causing Chormium trybots to fail, preventing rolls. Will reland if the revert does not fix it.

Original change's description:
> [Adaptation] Remove QualityScalerResource when disabled.
> 
> Bug: webrtc:11843
> Change-Id: I2d3e40356c266f189db0242f3c7590e6d83e4456
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181369
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31924}

TBR=ilnik@webrtc.org,eshr@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11843
Change-Id: Idc3950be209c6edce0dbe72d98c9b4becae0049f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181880
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31952}
2020-08-17 13:46:30 +00:00
..
adaptation Revert "[Adaptation] Remove QualityScalerResource when disabled." 2020-08-17 13:46:30 +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_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 Converts const methods in BitrateAllocator to non-member functions. 2019-09-25 11:55:13 +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_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
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
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_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 Insert frame transformer between Encoded and Packetizer. 2020-02-28 07:43:13 +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_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
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
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 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