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}
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@ -34,6 +34,7 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
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entry.timestamp_ms = now_ms;
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entry.audio_level = packet_info.audio_level();
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entry.absolute_capture_time = packet_info.absolute_capture_time();
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entry.rtp_timestamp = packet_info.rtp_timestamp();
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}
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@ -42,6 +43,7 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
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entry.timestamp_ms = now_ms;
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entry.audio_level = packet_info.audio_level();
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entry.absolute_capture_time = packet_info.absolute_capture_time();
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entry.rtp_timestamp = packet_info.rtp_timestamp();
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}
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@ -60,8 +62,9 @@ std::vector<RtpSource> SourceTracker::GetSources() const {
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const SourceKey& key = pair.first;
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const SourceEntry& entry = pair.second;
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sources.emplace_back(entry.timestamp_ms, key.source, key.source_type,
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entry.audio_level, entry.rtp_timestamp);
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sources.emplace_back(
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entry.timestamp_ms, key.source, key.source_type, entry.rtp_timestamp,
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RtpSource::Extensions{entry.audio_level, entry.absolute_capture_time});
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}
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return sources;
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