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;