/* * Copyright (c) 2021 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_ #define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_ #include "api/array_view.h" #include "api/rtp_headers.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" namespace webrtc { // // Helper class for interpolating the `AbsoluteCaptureTime` header extension. // // Supports the "timestamp interpolation" optimization: // A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture // timestamp, and RTP timestamp of the most recently received abs-capture-time // packet on each received stream. It can then use that information, in // combination with RTP timestamps of packets without abs-capture-time, to // extrapolate missing capture timestamps. // // See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/ // class AbsoluteCaptureTimeInterpolator { … }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_