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}
diff --git a/modules/rtp_rtcp/source/source_tracker.cc b/modules/rtp_rtcp/source/source_tracker.cc
index 2878b11..22b887c 100644
--- a/modules/rtp_rtcp/source/source_tracker.cc
+++ b/modules/rtp_rtcp/source/source_tracker.cc
@@ -34,6 +34,7 @@
 
       entry.timestamp_ms = now_ms;
       entry.audio_level = packet_info.audio_level();
+      entry.absolute_capture_time = packet_info.absolute_capture_time();
       entry.rtp_timestamp = packet_info.rtp_timestamp();
     }
 
@@ -42,6 +43,7 @@
 
     entry.timestamp_ms = now_ms;
     entry.audio_level = packet_info.audio_level();
+    entry.absolute_capture_time = packet_info.absolute_capture_time();
     entry.rtp_timestamp = packet_info.rtp_timestamp();
   }
 
@@ -60,8 +62,9 @@
     const SourceKey& key = pair.first;
     const SourceEntry& entry = pair.second;
 
-    sources.emplace_back(entry.timestamp_ms, key.source, key.source_type,
-                         entry.audio_level, entry.rtp_timestamp);
+    sources.emplace_back(
+        entry.timestamp_ms, key.source, key.source_type, entry.rtp_timestamp,
+        RtpSource::Extensions{entry.audio_level, entry.absolute_capture_time});
   }
 
   return sources;