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1c7602c65d
Bug: webrtc:13982 Change-Id: I0c4fe63f47d842fc5871baeb1137aa225bc10ea4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288960 Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39039}
46 lines
1.4 KiB
C++
46 lines
1.4 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_VIDEO_CODING_TIMING_INTER_FRAME_DELAY_H_
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#define MODULES_VIDEO_CODING_TIMING_INTER_FRAME_DELAY_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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namespace webrtc {
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class InterFrameDelay {
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public:
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InterFrameDelay();
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// Resets the estimate. Zeros are given as parameters.
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void Reset();
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// Calculates the delay of a frame with the given timestamp.
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// This method is called when the frame is complete.
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absl::optional<TimeDelta> CalculateDelay(uint32_t rtp_timestamp,
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Timestamp now);
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private:
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// The previous rtp timestamp passed to the delay estimate
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int64_t prev_rtp_timestamp_unwrapped_;
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RtpTimestampUnwrapper unwrapper_;
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// The previous wall clock timestamp used by the delay estimate
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absl::optional<Timestamp> prev_wall_clock_;
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};
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_TIMING_INTER_FRAME_DELAY_H_
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