TimestampExtrapolator maps RTP timestamps of received video frames to local timestamps. As part of this mapping, the clock drift between the local and remote clock is estimated. Add the histogram WebRTC.Video.EstimatedClockDrift_ppm to log the relative clock drift in points per million. Bug: b/363166487 Change-Id: I0c2e628ef72c05a93e1f3138c8f71c77467130b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368342 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43413}
191 lines
6.0 KiB
C++
191 lines
6.0 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/timing/timestamp_extrapolator.h"
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#include <algorithm>
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#include <optional>
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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constexpr int kMinimumSamplesToLogEstimatedClockDrift =
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3000; // 100 seconds at 30 fps.
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constexpr double kLambda = 1;
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constexpr int kStartUpFilterDelayInPackets = 2;
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constexpr double kAlarmThreshold = 60e3;
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// in timestamp ticks, i.e. 15 ms
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constexpr double kAccDrift = 6600;
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constexpr double kAccMaxError = 7000;
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constexpr double kP11 = 1e10;
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} // namespace
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TimestampExtrapolator::TimestampExtrapolator(Timestamp start)
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: start_(Timestamp::Zero()),
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prev_(Timestamp::Zero()),
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packet_count_(0),
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detector_accumulator_pos_(0),
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detector_accumulator_neg_(0) {
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Reset(start);
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}
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TimestampExtrapolator::~TimestampExtrapolator() {
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if (packet_count_ >= kMinimumSamplesToLogEstimatedClockDrift) {
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// Relative clock drift per million (ppm).
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double clock_drift_ppm = 1e6 * (w_[0] - 90.0) / 90.0;
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RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.EstimatedClockDrift_ppm",
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static_cast<int>(std::abs(clock_drift_ppm)));
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}
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}
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void TimestampExtrapolator::Reset(Timestamp start) {
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start_ = start;
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prev_ = start_;
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first_unwrapped_timestamp_ = std::nullopt;
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prev_unwrapped_timestamp_ = std::nullopt;
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w_[0] = 90.0;
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w_[1] = 0;
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p_[0][0] = 1;
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p_[1][1] = kP11;
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p_[0][1] = p_[1][0] = 0;
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unwrapper_ = RtpTimestampUnwrapper();
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packet_count_ = 0;
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detector_accumulator_pos_ = 0;
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detector_accumulator_neg_ = 0;
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}
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void TimestampExtrapolator::Update(Timestamp now, uint32_t ts90khz) {
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if (now - prev_ > TimeDelta::Seconds(10)) {
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// Ten seconds without a complete frame.
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// Reset the extrapolator
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Reset(now);
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} else {
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prev_ = now;
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}
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// Remove offset to prevent badly scaled matrices
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const TimeDelta offset = now - start_;
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double t_ms = offset.ms();
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int64_t unwrapped_ts90khz = unwrapper_.Unwrap(ts90khz);
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if (!first_unwrapped_timestamp_) {
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// Make an initial guess of the offset,
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// should be almost correct since t_ms - start
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// should about zero at this time.
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w_[1] = -w_[0] * t_ms;
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first_unwrapped_timestamp_ = unwrapped_ts90khz;
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}
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double residual =
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(static_cast<double>(unwrapped_ts90khz) - *first_unwrapped_timestamp_) -
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t_ms * w_[0] - w_[1];
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if (DelayChangeDetection(residual) &&
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packet_count_ >= kStartUpFilterDelayInPackets) {
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// A sudden change of average network delay has been detected.
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// Force the filter to adjust its offset parameter by changing
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// the offset uncertainty. Don't do this during startup.
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p_[1][1] = kP11;
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}
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if (prev_unwrapped_timestamp_ &&
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unwrapped_ts90khz < prev_unwrapped_timestamp_) {
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// Drop reordered frames.
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return;
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}
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// T = [t(k) 1]';
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// that = T'*w;
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// K = P*T/(lambda + T'*P*T);
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double K[2];
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K[0] = p_[0][0] * t_ms + p_[0][1];
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K[1] = p_[1][0] * t_ms + p_[1][1];
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double TPT = kLambda + t_ms * K[0] + K[1];
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K[0] /= TPT;
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K[1] /= TPT;
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// w = w + K*(ts(k) - that);
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w_[0] = w_[0] + K[0] * residual;
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w_[1] = w_[1] + K[1] * residual;
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// P = 1/lambda*(P - K*T'*P);
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double p00 =
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1 / kLambda * (p_[0][0] - (K[0] * t_ms * p_[0][0] + K[0] * p_[1][0]));
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double p01 =
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1 / kLambda * (p_[0][1] - (K[0] * t_ms * p_[0][1] + K[0] * p_[1][1]));
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p_[1][0] =
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1 / kLambda * (p_[1][0] - (K[1] * t_ms * p_[0][0] + K[1] * p_[1][0]));
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p_[1][1] =
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1 / kLambda * (p_[1][1] - (K[1] * t_ms * p_[0][1] + K[1] * p_[1][1]));
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p_[0][0] = p00;
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p_[0][1] = p01;
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prev_unwrapped_timestamp_ = unwrapped_ts90khz;
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if (packet_count_ < kStartUpFilterDelayInPackets ||
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packet_count_ < kMinimumSamplesToLogEstimatedClockDrift) {
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packet_count_++;
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}
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}
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std::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
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uint32_t timestamp90khz) const {
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int64_t unwrapped_ts90khz = unwrapper_.PeekUnwrap(timestamp90khz);
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if (!first_unwrapped_timestamp_) {
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return std::nullopt;
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}
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if (packet_count_ < kStartUpFilterDelayInPackets) {
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constexpr double kRtpTicksPerMs = 90;
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TimeDelta diff = TimeDelta::Millis(
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(unwrapped_ts90khz - *prev_unwrapped_timestamp_) / kRtpTicksPerMs);
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if (prev_.us() + diff.us() < 0) {
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// Prevent the construction of a negative Timestamp.
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// This scenario can occur when the RTP timestamp wraps around.
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return std::nullopt;
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}
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return prev_ + diff;
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}
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if (w_[0] < 1e-3) {
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return start_;
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}
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double timestamp_diff =
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static_cast<double>(unwrapped_ts90khz - *first_unwrapped_timestamp_);
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TimeDelta diff = TimeDelta::Millis(
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static_cast<int64_t>((timestamp_diff - w_[1]) / w_[0] + 0.5));
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if (start_.us() + diff.us() < 0) {
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// Prevent the construction of a negative Timestamp.
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// This scenario can occur when the RTP timestamp wraps around.
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return std::nullopt;
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}
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return start_ + diff;
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}
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bool TimestampExtrapolator::DelayChangeDetection(double error) {
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// CUSUM detection of sudden delay changes
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error = (error > 0) ? std::min(error, kAccMaxError)
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: std::max(error, -kAccMaxError);
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detector_accumulator_pos_ =
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std::max(detector_accumulator_pos_ + error - kAccDrift, double{0});
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detector_accumulator_neg_ =
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std::min(detector_accumulator_neg_ + error + kAccDrift, double{0});
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if (detector_accumulator_pos_ > kAlarmThreshold ||
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detector_accumulator_neg_ < -kAlarmThreshold) {
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// Alarm
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detector_accumulator_pos_ = detector_accumulator_neg_ = 0;
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return true;
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}
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return false;
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}
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} // namespace webrtc
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