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Report timing frames info in GetStats.

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Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.

The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
This commit is contained in:
ilnik
2017-07-06 03:06:50 -07:00
committed by Commit Bot
parent 5b7fc8ce42
commit 2edc6845ac
34 changed files with 376 additions and 67 deletions
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10
webrtc/modules/video_coding/frame_buffer2.cc
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@ -149,6 +149,7 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
}
UpdateJitterDelay();
UpdateTimingFrameInfo();
PropagateDecodability(next_frame_it_->second);
// Sanity check for RTP timestamp monotonicity.
@ -534,8 +535,15 @@ void FrameBuffer::UpdateJitterDelay() {
}
}
void FrameBuffer::UpdateTimingFrameInfo() {
TRACE_EVENT0("webrtc", "FrameBuffer::UpdateTimingFrameInfo");
rtc::Optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
if (info)
stats_callback_->OnTimingFrameInfoUpdated(*info);
}
void FrameBuffer::ClearFramesAndHistory() {
TRACE_EVENT0("webrtc", "FrameBuffer::UpdateJitterDelay");
TRACE_EVENT0("webrtc", "FrameBuffer::ClearFramesAndHistory");
frames_.clear();
last_decoded_frame_it_ = frames_.end();
last_continuous_frame_it_ = frames_.end();