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e9126c18bf
Additionally, * Moved to its own GN target. * Added unittests. * Removed unused variable `_zeroWallClock`. * Renamed variables to match style guide. * Moved fields _dTS and _wrapArounds to variables. Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36147}
71 lines
2.4 KiB
C++
71 lines
2.4 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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#include "modules/video_coding/inter_frame_delay.h"
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#include "absl/types/optional.h"
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#include "api/units/frequency.h"
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#include "api/units/time_delta.h"
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#include "modules/include/module_common_types_public.h"
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namespace webrtc {
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namespace {
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constexpr Frequency k90kHz = Frequency::KiloHertz(90);
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}
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VCMInterFrameDelay::VCMInterFrameDelay() {
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Reset();
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}
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// Resets the delay estimate.
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void VCMInterFrameDelay::Reset() {
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prev_wall_clock_ = absl::nullopt;
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prev_rtp_timestamp_unwrapped_ = 0;
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}
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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> VCMInterFrameDelay::CalculateDelay(
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uint32_t rtp_timestamp,
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Timestamp now) {
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int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
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if (!prev_wall_clock_) {
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// First set of data, initialization, wait for next frame.
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prev_wall_clock_ = now;
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prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
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return TimeDelta::Zero();
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}
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// Account for reordering in jitter variance estimate in the future?
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// Note that this also captures incomplete frames which are grabbed for
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// decoding after a later frame has been complete, i.e. real packet losses.
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uint32_t cropped_last = static_cast<uint32_t>(prev_rtp_timestamp_unwrapped_);
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if (rtp_timestamp_unwrapped < prev_rtp_timestamp_unwrapped_ ||
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!IsNewerTimestamp(rtp_timestamp, cropped_last)) {
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return absl::nullopt;
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}
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// Compute the compensated timestamp difference.
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int64_t d_rtp_ticks = rtp_timestamp_unwrapped - prev_rtp_timestamp_unwrapped_;
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TimeDelta dts = d_rtp_ticks / k90kHz;
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TimeDelta dt = now - *prev_wall_clock_;
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// frameDelay is the difference of dT and dTS -- i.e. the difference of the
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// wall clock time difference and the timestamp difference between two
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// following frames.
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TimeDelta delay = dt - dts;
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prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
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prev_wall_clock_ = now;
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return delay;
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}
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} // namespace webrtc
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