api_call_jitter_metrics.h

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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_