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/*
* Copyright (c) 2016 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 "rtc_base/timestamp_aligner.h"
#include <cstdint>
#include <cstdlib>
#include <limits>
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
TimestampAligner::TimestampAligner()
: frames_seen_(0),
offset_us_(0),
clip_bias_us_(0),
prev_translated_time_us_(std::numeric_limits<int64_t>::min()),
prev_time_offset_us_(0) {}
TimestampAligner::~TimestampAligner() {}
int64_t TimestampAligner::TranslateTimestamp(int64_t capturer_time_us,
int64_t system_time_us) {
const int64_t translated_timestamp = ClipTimestamp(
capturer_time_us + UpdateOffset(capturer_time_us, system_time_us),
system_time_us);
prev_time_offset_us_ = translated_timestamp - capturer_time_us;
return translated_timestamp;
}
int64_t TimestampAligner::TranslateTimestamp(int64_t capturer_time_us) const {
return capturer_time_us + prev_time_offset_us_;
}
int64_t TimestampAligner::UpdateOffset(int64_t capturer_time_us,
int64_t system_time_us) {
// Estimate the offset between system monotonic time and the capturer's
// time. The capturer is assumed to provide more
// accurate timestamps than we get from the system time. But the
// capturer may use its own free-running clock with a large offset and
// a small drift compared to the system clock. So the model is
// basically
//
// y_k = c_0 + c_1 * x_k + v_k
//
// where x_k is the capturer's timestamp, believed to be accurate in its
// own scale. y_k is our reading of the system clock. v_k is the
// measurement noise, i.e., the delay from frame capture until the
// system clock was read.
//
// It's possible to do (weighted) least-squares estimation of both
// c_0 and c_1. Then we get the constants as c_1 = Cov(x,y) /
// Var(x), and c_0 = mean(y) - c_1 * mean(x). Substituting this c_0,
// we can rearrange the model as
//
// y_k = mean(y) + (x_k - mean(x)) + (c_1 - 1) * (x_k - mean(x)) + v_k
//
// Now if we use a weighted average which gradually forgets old
// values, x_k - mean(x) is bounded, of the same order as the time
// constant (and close to constant for a steady frame rate). In
// addition, the frequency error |c_1 - 1| should be small. Cameras
// with a frequency error up to 3000 ppm (3 ms drift per second)
// have been observed, but frequency errors below 100 ppm could be
// expected of any cheap crystal.
//
// Bottom line is that we ignore the c_1 term, and use only the estimator
//
// x_k + mean(y-x)
//
// where mean is plain averaging for initial samples, followed by
// exponential averaging.
// The input for averaging, y_k - x_k in the above notation.
int64_t diff_us = system_time_us - capturer_time_us;
// The deviation from the current average.
int64_t error_us = diff_us - offset_us_;
// If the current difference is far from the currently estimated
// offset, the filter is reset. This could happen, e.g., if the
// capturer's clock is reset, cameras are plugged in and out, or
// the application process is temporarily suspended. Expected to
// happen for the very first timestamp (`frames_seen_` = 0). The
// threshold of 300 ms should make this unlikely in normal
// operation, and at the same time, converging gradually rather than
// resetting the filter should be tolerable for jumps in capturer's time
// below this threshold.
static const int64_t kResetThresholdUs = 300000;
if (std::abs(error_us) > kResetThresholdUs) {
RTC_LOG(LS_INFO) << "Resetting timestamp translation after averaging "
<< frames_seen_ << " frames. Old offset: " << offset_us_
<< ", new offset: " << diff_us;
frames_seen_ = 0;
clip_bias_us_ = 0;
}
static const int kWindowSize = 100;
if (frames_seen_ < kWindowSize) {
++frames_seen_;
}
offset_us_ += error_us / frames_seen_;
return offset_us_;
}
int64_t TimestampAligner::ClipTimestamp(int64_t filtered_time_us,
int64_t system_time_us) {
// Clip to make sure we don't produce timestamps in the future.
int64_t time_us = filtered_time_us - clip_bias_us_;
if (time_us > system_time_us) {
clip_bias_us_ += time_us - system_time_us;
time_us = system_time_us;
}
// Make timestamps monotonic, with a minimum inter-frame interval of 1 ms.
else if (time_us < prev_translated_time_us_ + kMinFrameIntervalUs) {
time_us = prev_translated_time_us_ + kMinFrameIntervalUs;
if (time_us > system_time_us) {
// In the anomalous case that this function is called with values of
// `system_time_us` less than `kMinFrameIntervalUs` apart, we may output
// timestamps with with too short inter-frame interval. We may even return
// duplicate timestamps in case this function is called several times with
// exactly the same `system_time_us`.
RTC_LOG(LS_WARNING) << "too short translated timestamp interval: "
"system time (us) = "
<< system_time_us << ", interval (us) = "
<< system_time_us - prev_translated_time_us_;
time_us = system_time_us;
}
}
RTC_DCHECK_GE(time_us, prev_translated_time_us_);
RTC_DCHECK_LE(time_us, system_time_us);
prev_translated_time_us_ = time_us;
return time_us;
}
} // namespace webrtc
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