chromium/third_party/webrtc/video/rtp_streams_synchronizer2.cc

/*
 *  Copyright (c) 2020 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 "video/rtp_streams_synchronizer2.h"

#include "absl/types/optional.h"
#include "call/syncable.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/rtp_to_ntp_estimator.h"

namespace webrtc {
namespace internal {
namespace {
// Time interval for logging stats.
constexpr int64_t kStatsLogIntervalMs = …;
constexpr TimeDelta kSyncInterval = …;

bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const Syncable::Info& info) { … }

}  // namespace

RtpStreamsSynchronizer::RtpStreamsSynchronizer(TaskQueueBase* main_queue,
                                               Syncable* syncable_video)
    : … { … }

RtpStreamsSynchronizer::~RtpStreamsSynchronizer() { … }

void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) { … }

void RtpStreamsSynchronizer::UpdateDelay() { … }

// TODO(https://bugs.webrtc.org/7065): Move RtpToNtpEstimator out of
// RtpStreamsSynchronizer and into respective receive stream to always populate
// the estimated playout timestamp.
bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
    uint32_t rtp_timestamp,
    int64_t render_time_ms,
    int64_t* video_playout_ntp_ms,
    int64_t* stream_offset_ms,
    double* estimated_freq_khz) const { … }

}  // namespace internal
}  // namespace webrtc