chromium/third_party/webrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc

/*
 *  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)
    : … { … }

uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
    uint32_t ssrc,
    rtc::ArrayView<const uint32_t> csrcs) { … }

absl::optional<AbsoluteCaptureTime>
AbsoluteCaptureTimeInterpolator::OnReceivePacket(
    uint32_t source,
    uint32_t rtp_timestamp,
    int rtp_clock_frequency_hz,
    const absl::optional<AbsoluteCaptureTime>& 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) { … }

bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension(
    Timestamp receive_time,
    uint32_t source,
    uint32_t rtp_timestamp,
    int rtp_clock_frequency_hz) const { … }

}  // namespace webrtc