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webrtc_m130/video
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Christoffer Rodbro c2a028887f Enable audio in video_quality_test. 
Allows enabling audio for RunWithAnalyzer method, and prints out audio jitterbuffer performance stats. Also fixes for RunWithRenderer when enabling audio (seg-faulted).

Bug: b/112299470
Change-Id: Ic7c0de1c455891f38cca317001c6c216e82f6ec3
Reviewed-on: https://webrtc-review.googlesource.com/92800
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24208}
2018-08-07 13:49:04 +00:00
..
end_to_end_tests
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
test
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
BUILD.gn
Never invoke rtc::LogMessage::SetLogToStderr outside of main.
2018-07-31 17:24:09 +00:00
call_stats_unittest.cc
Reduce locking for CallStats (preparation for TaskQueue).
2018-03-28 13:24:07 +00:00
call_stats.cc
Reduce locking for CallStats (preparation for TaskQueue).
2018-03-28 13:24:07 +00:00
call_stats.h
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
DEPS
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
encoder_rtcp_feedback_unittest.cc
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
encoder_rtcp_feedback.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
encoder_rtcp_feedback.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
full_stack_tests_plot.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
full_stack_tests.cc
Never invoke rtc::LogMessage::SetLogToStderr outside of main.
2018-07-31 17:24:09 +00:00
overuse_frame_detector_unittest.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
overuse_frame_detector.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
overuse_frame_detector.h
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
OWNERS
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picture_id_tests.cc
Makes some CallTest members private.
2018-07-13 11:15:45 +00:00
quality_scaling_tests.cc
Replaces call config create in tests with modify.
2018-07-13 12:52:45 +00:00
quality_threshold_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quality_threshold.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
quality_threshold.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
receive_statistics_proxy_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
receive_statistics_proxy.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
receive_statistics_proxy.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
replay.cc
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
report_block_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
report_block_stats.cc
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
report_block_stats.h
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
rtp_streams_synchronizer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_streams_synchronizer.h
Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
2018-02-07 10:07:28 +00:00
rtp_video_stream_receiver_unittest.cc
Reland "Remove RTPVideoHeader::h264() accessors."
2018-08-03 09:16:50 +00:00
rtp_video_stream_receiver.cc
Delete RtpVideoStreamReceiver methods GetRtpReceiver and rtp_rtcp
2018-07-31 07:02:18 +00:00
rtp_video_stream_receiver.h
Delete RtpVideoStreamReceiver methods GetRtpReceiver and rtp_rtcp
2018-07-31 07:02:18 +00:00
screenshare_loopback.cc
Never invoke rtc::LogMessage::SetLogToStderr outside of main.
2018-07-31 17:24:09 +00:00
send_delay_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_delay_stats.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
send_delay_stats.h
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
send_statistics_proxy_unittest.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
send_statistics_proxy.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
send_statistics_proxy.h
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
stats_counter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stats_counter.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
stats_counter.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
stream_synchronization_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stream_synchronization.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stream_synchronization.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
sv_loopback.cc
Never invoke rtc::LogMessage::SetLogToStderr outside of main.
2018-07-31 17:24:09 +00:00
transport_adapter.cc
Replace Atomic32 with std::atomic in video/
2017-10-24 23:40:29 +00:00
transport_adapter.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_analyzer.cc
Enable audio in video_quality_test.
2018-08-07 13:49:04 +00:00
video_analyzer.h
Enable audio in video_quality_test.
2018-08-07 13:49:04 +00:00
video_loopback.cc
Never invoke rtc::LogMessage::SetLogToStderr outside of main.
2018-07-31 17:24:09 +00:00
video_quality_observer.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_quality_observer.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_quality_test.cc
Enable audio in video_quality_test.
2018-08-07 13:49:04 +00:00
video_quality_test.h
Enable audio in video_quality_test.
2018-08-07 13:49:04 +00:00
video_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_receive_stream.cc
Delete RtpVideoStreamReceiver methods GetRtpReceiver and rtp_rtcp
2018-07-31 07:02:18 +00:00
video_receive_stream.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_send_stream_impl_unittest.cc
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
video_send_stream_impl.cc
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
video_send_stream_impl.h
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
video_send_stream_tests.cc
Cleanup networkroute.h
2018-07-23 19:35:28 +00:00
video_send_stream.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
video_send_stream.h
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
video_stream_decoder_impl.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_stream_decoder_impl.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_stream_decoder.cc
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
video_stream_decoder.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
video_stream_encoder_unittest.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
video_stream_encoder.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
video_stream_encoder.h
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00