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/*
* 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
#define MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
namespace webrtc {
// Stores data for reporting metrics on the API call jitter.
class ApiCallJitterMetrics {
public:
class Jitter {
public:
Jitter();
void Update(int num_api_calls_in_a_row);
void Reset();
int min() const { return min_; }
int max() const { return max_; }
private:
int max_;
int min_;
};
ApiCallJitterMetrics() { Reset(); }
// Update metrics for render API call.
void ReportRenderCall();
// Update and periodically report metrics for capture API call.
void ReportCaptureCall();
// Methods used only for testing.
const Jitter& render_jitter() const { return render_jitter_; }
const Jitter& capture_jitter() const { return capture_jitter_; }
bool WillReportMetricsAtNextCapture() const;
private:
void Reset();
Jitter render_jitter_;
Jitter capture_jitter_;
int num_api_calls_in_a_row_ = 0;
int frames_since_last_report_ = 0;
bool last_call_was_render_ = false;
bool proper_call_observed_ = false;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
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