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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;
};
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;
};
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, 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_
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