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/*
* Copyright (c) 2012 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.
*/
#include "system_wrappers/include/rtp_to_ntp_estimator.h"
#include <stddef.h>
#include <cmath>
#include <vector>
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
// Maximum number of RTCP SR reports to use to map between RTP and NTP.
const size_t kNumRtcpReportsToUse = 20;
// Don't allow NTP timestamps to jump more than 1 hour. Chosen arbitrary as big
// enough to not affect normal use-cases. Yet it is smaller than RTP wrap-around
// half-period (90khz RTP clock wrap-arounds every 13.25 hours). After half of
// wrap-around period it is impossible to unwrap RTP timestamps correctly.
const int kMaxAllowedRtcpNtpIntervalMs = 60 * 60 * 1000;
bool Contains(const std::list<RtpToNtpEstimator::RtcpMeasurement>& measurements,
const RtpToNtpEstimator::RtcpMeasurement& other) {
for (const auto& measurement : measurements) {
if (measurement.IsEqual(other))
return true;
}
return false;
}
// Given x[] and y[] writes out such k and b that line y=k*x+b approximates
// given points in the best way (Least Squares Method).
bool LinearRegression(rtc::ArrayView<const double> x,
rtc::ArrayView<const double> y,
double* k,
double* b) {
size_t n = x.size();
if (n < 2)
return false;
if (y.size() != n)
return false;
double avg_x = 0;
double avg_y = 0;
for (size_t i = 0; i < n; ++i) {
avg_x += x[i];
avg_y += y[i];
}
avg_x /= n;
avg_y /= n;
double variance_x = 0;
double covariance_xy = 0;
for (size_t i = 0; i < n; ++i) {
double normalized_x = x[i] - avg_x;
double normalized_y = y[i] - avg_y;
variance_x += normalized_x * normalized_x;
covariance_xy += normalized_x * normalized_y;
}
if (std::fabs(variance_x) < 1e-8)
return false;
*k = static_cast<double>(covariance_xy / variance_x);
*b = static_cast<double>(avg_y - (*k) * avg_x);
return true;
}
} // namespace
RtpToNtpEstimator::RtcpMeasurement::RtcpMeasurement(uint32_t ntp_secs,
uint32_t ntp_frac,
int64_t unwrapped_timestamp)
: ntp_time(ntp_secs, ntp_frac),
unwrapped_rtp_timestamp(unwrapped_timestamp) {}
bool RtpToNtpEstimator::RtcpMeasurement::IsEqual(
const RtcpMeasurement& other) const {
// Use || since two equal timestamps will result in zero frequency and in
// RtpToNtpMs, |rtp_timestamp_ms| is estimated by dividing by the frequency.
return (ntp_time == other.ntp_time) ||
(unwrapped_rtp_timestamp == other.unwrapped_rtp_timestamp);
}
// Class for converting an RTP timestamp to the NTP domain.
RtpToNtpEstimator::RtpToNtpEstimator() : consecutive_invalid_samples_(0) {}
RtpToNtpEstimator::~RtpToNtpEstimator() {}
void RtpToNtpEstimator::UpdateParameters() {
if (measurements_.size() < 2)
return;
std::vector<double> x;
std::vector<double> y;
x.reserve(measurements_.size());
y.reserve(measurements_.size());
for (auto it = measurements_.begin(); it != measurements_.end(); ++it) {
x.push_back(it->unwrapped_rtp_timestamp);
y.push_back(it->ntp_time.ToMs());
}
double slope, offset;
if (!LinearRegression(x, y, &slope, &offset)) {
return;
}
params_.emplace(1 / slope, offset);
}
bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
uint32_t ntp_frac,
uint32_t rtp_timestamp,
bool* new_rtcp_sr) {
*new_rtcp_sr = false;
int64_t unwrapped_rtp_timestamp = unwrapper_.Unwrap(rtp_timestamp);
RtcpMeasurement new_measurement(ntp_secs, ntp_frac, unwrapped_rtp_timestamp);
if (Contains(measurements_, new_measurement)) {
// RTCP SR report already added.
return true;
}
if (!new_measurement.ntp_time.Valid())
return false;
int64_t ntp_ms_new = new_measurement.ntp_time.ToMs();
bool invalid_sample = false;
if (!measurements_.empty()) {
int64_t old_rtp_timestamp = measurements_.front().unwrapped_rtp_timestamp;
int64_t old_ntp_ms = measurements_.front().ntp_time.ToMs();
if (ntp_ms_new <= old_ntp_ms ||
ntp_ms_new > old_ntp_ms + kMaxAllowedRtcpNtpIntervalMs) {
invalid_sample = true;
} else if (unwrapped_rtp_timestamp <= old_rtp_timestamp) {
RTC_LOG(LS_WARNING)
<< "Newer RTCP SR report with older RTP timestamp, dropping";
invalid_sample = true;
} else if (unwrapped_rtp_timestamp - old_rtp_timestamp > (1 << 25)) {
// Sanity check. No jumps too far into the future in rtp.
invalid_sample = true;
}
}
if (invalid_sample) {
++consecutive_invalid_samples_;
if (consecutive_invalid_samples_ < kMaxInvalidSamples) {
return false;
}
RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
"clearing measurements.";
measurements_.clear();
params_ = absl::nullopt;
}
consecutive_invalid_samples_ = 0;
// Insert new RTCP SR report.
if (measurements_.size() == kNumRtcpReportsToUse)
measurements_.pop_back();
measurements_.push_front(new_measurement);
*new_rtcp_sr = true;
// List updated, calculate new parameters.
UpdateParameters();
return true;
}
bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp,
int64_t* ntp_timestamp_ms) const {
if (!params_)
return false;
int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
// params_calculated_ should not be true unless ms params.frequency_khz has
// been calculated to something non zero.
RTC_DCHECK_NE(params_->frequency_khz, 0.0);
double rtp_ms =
static_cast<double>(rtp_timestamp_unwrapped) / params_->frequency_khz +
params_->offset_ms + 0.5f;
if (rtp_ms < 0)
return false;
*ntp_timestamp_ms = rtp_ms;
return true;
}
const absl::optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params()
const {
return params_;
}
} // namespace webrtc