chromium/third_party/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.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_processing/aec3/api_call_jitter_metrics.h"

#include <algorithm>
#include <limits>

#include "modules/audio_processing/aec3/aec3_common.h"
#include "system_wrappers/include/metrics.h"

namespace webrtc {
namespace {

bool TimeToReportMetrics(int frames_since_last_report) { … }

}  // namespace

ApiCallJitterMetrics::Jitter::Jitter()
    : … { … }

void ApiCallJitterMetrics::Jitter::Update(int num_api_calls_in_a_row) { … }

void ApiCallJitterMetrics::Jitter::Reset() { … }

void ApiCallJitterMetrics::Reset() { … }

void ApiCallJitterMetrics::ReportRenderCall() { … }

void ApiCallJitterMetrics::ReportCaptureCall() { … }

bool ApiCallJitterMetrics::WillReportMetricsAtNextCapture() const { … }

}  // namespace webrtc