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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_sender.h"
#include <algorithm>
#include <cstdint>
#include <optional>
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
static_assert(
AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval >=
AbsoluteCaptureTimeSender::kInterpolationMaxInterval,
"Receivers should be as willing to interpolate timestamps as senders.");
AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
: clock_(clock) {}
uint32_t AbsoluteCaptureTimeSender::GetSource(uint32_t ssrc,
ArrayView<const uint32_t> csrcs) {
return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
}
std::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
std::optional<int64_t> estimated_capture_clock_offset) {
return OnSendPacket(source, rtp_timestamp, rtp_clock_frequency,
NtpTime(absolute_capture_timestamp),
estimated_capture_clock_offset, /*force=*/false);
}
std::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
std::optional<int64_t> estimated_capture_clock_offset,
bool force) {
Timestamp send_time = clock_->CurrentTime();
if (!(force || ShouldSendExtension(
send_time, source, rtp_timestamp, rtp_clock_frequency_hz,
absolute_capture_time, estimated_capture_clock_offset))) {
return std::nullopt;
}
last_source_ = source;
last_rtp_timestamp_ = rtp_timestamp;
last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz;
last_absolute_capture_time_ = absolute_capture_time;
last_estimated_capture_clock_offset_ = estimated_capture_clock_offset;
last_send_time_ = send_time;
return AbsoluteCaptureTime{
.absolute_capture_timestamp = uint64_t{absolute_capture_time},
.estimated_capture_clock_offset = estimated_capture_clock_offset,
};
}
bool AbsoluteCaptureTimeSender::ShouldSendExtension(
Timestamp send_time,
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
std::optional<int64_t> estimated_capture_clock_offset) const {
// Should if the last sent extension is too old, in particular if we've never
// sent anything before.
if (send_time - last_send_time_ > kInterpolationMaxInterval) {
return true;
}
// Should if the source has changed.
if (last_source_ != source) {
return true;
}
// Should if the RTP clock frequency has changed.
if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) {
return true;
}
// Should if the RTP clock frequency is invalid.
if (rtp_clock_frequency_hz <= 0) {
return true;
}
// Should if the estimated capture clock offset has changed.
if (last_estimated_capture_clock_offset_ != estimated_capture_clock_offset) {
return true;
}
// Should if interpolation would introduce too much error.
const uint64_t interpolated_absolute_capture_timestamp =
AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
uint64_t{last_absolute_capture_time_});
const uint64_t absolute_capture_timestamp = uint64_t{absolute_capture_time};
const int64_t interpolation_error_ms = UQ32x32ToInt64Ms(std::min(
interpolated_absolute_capture_timestamp - absolute_capture_timestamp,
absolute_capture_timestamp - interpolated_absolute_capture_timestamp));
if (interpolation_error_ms > kInterpolationMaxError.ms()) {
return true;
}
return false;
}
} // namespace webrtc
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