chromium/third_party/webrtc/video/stream_synchronization.cc

/*
 *  Copyright (c) 2012 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/stream_synchronization.h"

#include <stdlib.h>

#include <algorithm>

#include "rtc_base/logging.h"

namespace webrtc {

static const int kMaxChangeMs = …;
static const int kMaxDeltaDelayMs = …;
static const int kFilterLength = …;
// Minimum difference between audio and video to warrant a change.
static const int kMinDeltaMs = …;

StreamSynchronization::StreamSynchronization(uint32_t video_stream_id,
                                             uint32_t audio_stream_id)
    : … { … }

bool StreamSynchronization::ComputeRelativeDelay(
    const Measurements& audio_measurement,
    const Measurements& video_measurement,
    int* relative_delay_ms) { … }

bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
                                          int current_audio_delay_ms,
                                          int* total_audio_delay_target_ms,
                                          int* total_video_delay_target_ms) { … }

void StreamSynchronization::SetTargetBufferingDelay(int target_delay_ms) { … }

void StreamSynchronization::ReduceAudioDelay() { … }

void StreamSynchronization::ReduceVideoDelay() { … }

}  // namespace webrtc