chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc

/*
 *  Copyright (c) 2018 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/audio_coding/neteq/tools/neteq_stats_getter.h"

#include <algorithm>
#include <numeric>
#include <utility>

#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"

namespace webrtc {
namespace test {

std::string NetEqStatsGetter::ConcealmentEvent::ToString() const { … }

NetEqStatsGetter::NetEqStatsGetter(
    std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer)
    : … { … }

void NetEqStatsGetter::BeforeGetAudio(NetEq* neteq) { … }

void NetEqStatsGetter::AfterGetAudio(int64_t time_now_ms,
                                     const AudioFrame& audio_frame,
                                     bool muted,
                                     NetEq* neteq) { … }

double NetEqStatsGetter::AverageSpeechExpandRate() const { … }

NetEqStatsGetter::Stats NetEqStatsGetter::AverageStats() const { … }

}  // namespace test
}  // namespace webrtc