Add absolute capture time property to rtp sources.

This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}

modules/rtp_rtcp/source/source_tracker.h

@@ -90,6 +90,11 @@ class SourceTracker {
     // specs for `RTCRtpContributingSource` for more info.
     absl::optional<uint8_t> audio_level;
 
+    // Absolute capture time header extension received or interpolated from the
+    // most recent packet used to assemble the frame. For more info see
+    // https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+    absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+
     // RTP timestamp of the most recent packet used to assemble the frame
     // associated with |timestamp_ms|.
     uint32_t rtp_timestamp;