Report timing frames info in GetStats.
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}
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@ -134,9 +134,10 @@ class VideoTimingExtension {
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static constexpr const char* kUri =
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"http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
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static bool Parse(rtc::ArrayView<const uint8_t> data, VideoTiming* timing);
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static size_t ValueSize(const VideoTiming&) { return kValueSizeBytes; }
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static bool Write(uint8_t* data, const VideoTiming& timing);
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static bool Parse(rtc::ArrayView<const uint8_t> data,
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VideoSendTiming* timing);
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static size_t ValueSize(const VideoSendTiming&) { return kValueSizeBytes; }
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static bool Write(uint8_t* data, const VideoSendTiming& timing);
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static size_t ValueSize(uint16_t time_delta_ms, uint8_t idx) {
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return kValueSizeBytes;
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