default_audio_quality_analyzer.h

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/*

 *  Copyright (c) 2019 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 TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_

#define TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_

#include <map>

#include <string>

#include "absl/strings/string_view.h"

#include "api/numerics/samples_stats_counter.h"

#include "api/test/audio_quality_analyzer_interface.h"

#include "api/test/metrics/metrics_logger.h"

#include "api/test/track_id_stream_info_map.h"

#include "api/units/time_delta.h"

#include "rtc_base/synchronization/mutex.h"

namespace webrtc {

namespace webrtc_pc_e2e {

struct AudioStreamStats {

  SamplesStatsCounter expand_rate;

  SamplesStatsCounter accelerate_rate;

  SamplesStatsCounter preemptive_rate;

  SamplesStatsCounter speech_expand_rate;

  SamplesStatsCounter average_jitter_buffer_delay_ms;

  SamplesStatsCounter preferred_buffer_size_ms;

  SamplesStatsCounter energy;

};

class DefaultAudioQualityAnalyzer : public AudioQualityAnalyzerInterface {

 public:

  explicit DefaultAudioQualityAnalyzer(

      test::MetricsLogger* const metrics_logger);

  void Start(std::string test_case_name,

             TrackIdStreamInfoMap* analyzer_helper) override;

  void OnStatsReports(

      absl::string_view pc_label,

      const rtc::scoped_refptr<const RTCStatsReport>& report) override;

  void Stop() override;

  // Returns audio quality stats per stream label.

  std::map<std::string, AudioStreamStats> GetAudioStreamsStats() const;

 private:

  struct StatsSample {

    uint64_t total_samples_received = 0;

    uint64_t concealed_samples = 0;

    uint64_t removed_samples_for_acceleration = 0;

    uint64_t inserted_samples_for_deceleration = 0;

    uint64_t silent_concealed_samples = 0;

    TimeDelta jitter_buffer_delay = TimeDelta::Zero();

    TimeDelta jitter_buffer_target_delay = TimeDelta::Zero();

    uint64_t jitter_buffer_emitted_count = 0;

    double total_samples_duration = 0.0;

    double total_audio_energy = 0.0;

};

  std::string GetTestCaseName(const std::string& stream_label) const;

  test::MetricsLogger* const metrics_logger_;

  std::string test_case_name_;

  TrackIdStreamInfoMap* analyzer_helper_;

  mutable Mutex lock_;

  std::map<std::string, AudioStreamStats> streams_stats_ RTC_GUARDED_BY(lock_);

  std::map<std::string, TrackIdStreamInfoMap::StreamInfo> stream_info_

      RTC_GUARDED_BY(lock_);

  std::map<std::string, StatsSample> last_stats_sample_ RTC_GUARDED_BY(lock_);

};

}  // namespace webrtc_pc_e2e

}  // namespace webrtc

#endif  // TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_