| commit | fcf79cca7bb92d9ec9b8d2f8146d3c076a6ad365 | [log] [tgz] |
|---|---|---|
| author | Åsa Persson <asapersson@webrtc.org> | Tue Oct 22 13:23:44 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Oct 23 07:46:39 2019 |
| tree | 39c8b80ff830e16aaa1453bb5271d0d005c81f5f | |
| parent | 261fc5197ea832181a162b67030fb330c3da6841 [diff] [blame] |
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp Partial implementation: currently only populated when a/v sync is enabled. Bug: webrtc:7065 Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29581}
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 1f8ad10..55c1af7 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h
@@ -87,6 +87,8 @@ double relative_packet_arrival_delay_seconds = 0.0; int32_t interruption_count = 0; int32_t total_interruption_duration_ms = 0; + // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp + absl::optional<int64_t> estimated_playout_ntp_timestamp_ms; }; struct Config {