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.h b/modules/rtp_rtcp/source/source_tracker.h
index 3ea16e9..fcf99bf 100644
--- a/modules/rtp_rtcp/source/source_tracker.h
+++ b/modules/rtp_rtcp/source/source_tracker.h
@@ -90,6 +90,11 @@
// 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;