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/*
* Copyright (c) 2019 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 "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
#include <limits>
#include "rtc_base/checks.h"
namespace webrtc {
AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
: clock_(clock) {}
uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
uint32_t ssrc,
rtc::ArrayView<const uint32_t> csrcs) {
if (csrcs.empty()) {
return ssrc;
}
return csrcs[0];
}
absl::optional<AbsoluteCaptureTime>
AbsoluteCaptureTimeInterpolator::OnReceivePacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
const absl::optional<AbsoluteCaptureTime>& received_extension) {
const Timestamp receive_time = clock_->CurrentTime();
MutexLock lock(&mutex_);
if (received_extension == absl::nullopt) {
if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
rtp_clock_frequency_hz)) {
last_receive_time_ = Timestamp::MinusInfinity();
return absl::nullopt;
}
return AbsoluteCaptureTime{
.absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp(
rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
last_received_extension_.absolute_capture_timestamp),
.estimated_capture_clock_offset =
last_received_extension_.estimated_capture_clock_offset,
};
} else {
last_source_ = source;
last_rtp_timestamp_ = rtp_timestamp;
last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz;
last_received_extension_ = *received_extension;
last_receive_time_ = receive_time;
return received_extension;
}
}
uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
uint32_t last_rtp_timestamp,
uint64_t last_absolute_capture_timestamp) {
RTC_DCHECK_GT(rtp_clock_frequency_hz, 0);
return last_absolute_capture_timestamp +
static_cast<int64_t>(uint64_t{rtp_timestamp - last_rtp_timestamp}
<< 32) /
rtp_clock_frequency_hz;
}
bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension(
Timestamp receive_time,
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz) const {
// Shouldn't if the last received extension is not eligible for interpolation,
// in particular if we don't have a previously received extension stored.
if (receive_time - last_receive_time_ > kInterpolationMaxInterval) {
return false;
}
// Shouldn't if the source has changed.
if (last_source_ != source) {
return false;
}
// Shouldn't if the RTP clock frequency has changed.
if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) {
return false;
}
// Shouldn't if the RTP clock frequency is invalid.
if (rtp_clock_frequency_hz <= 0) {
return false;
}
return true;
}
} // namespace webrtc
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