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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_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
#include <cstddef>
#include <cstdint>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/audio/audio_frame.h"
#include "api/neteq/neteq.h"
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
namespace webrtc {
namespace test {
class NetEqStatsGetter : public NetEqGetAudioCallback {
public:
// This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
// values stored in double precision.
struct Stats {
double current_buffer_size_ms = 0.0;
double preferred_buffer_size_ms = 0.0;
double jitter_peaks_found = 0.0;
double packet_loss_rate = 0.0;
double expand_rate = 0.0;
double speech_expand_rate = 0.0;
double preemptive_rate = 0.0;
double accelerate_rate = 0.0;
double secondary_decoded_rate = 0.0;
double secondary_discarded_rate = 0.0;
double clockdrift_ppm = 0.0;
double added_zero_samples = 0.0;
double mean_waiting_time_ms = 0.0;
double median_waiting_time_ms = 0.0;
double min_waiting_time_ms = 0.0;
double max_waiting_time_ms = 0.0;
};
struct ConcealmentEvent {
uint64_t duration_ms;
size_t concealment_event_number;
int64_t time_from_previous_event_end_ms;
std::string ToString() const;
};
// Takes a pointer to another callback object, which will be invoked after
// this object finishes. This does not transfer ownership, and null is a
// valid value.
explicit NetEqStatsGetter(std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer);
void set_stats_query_interval_ms(int64_t stats_query_interval_ms) {
stats_query_interval_ms_ = stats_query_interval_ms;
}
void BeforeGetAudio(NetEq* neteq) override;
void AfterGetAudio(int64_t time_now_ms,
const AudioFrame& audio_frame,
bool muted,
NetEq* neteq) override;
double AverageSpeechExpandRate() const;
NetEqDelayAnalyzer* delay_analyzer() const { return delay_analyzer_.get(); }
const std::vector<ConcealmentEvent>& concealment_events() const {
// Do not account for the last concealment event to avoid potential end
// call skewing.
return concealment_events_;
}
const std::vector<std::pair<int64_t, NetEqNetworkStatistics>>* stats() const {
return &stats_;
}
const std::vector<std::pair<int64_t, NetEqLifetimeStatistics>>*
lifetime_stats() const {
return &lifetime_stats_;
}
Stats AverageStats() const;
private:
std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer_;
int64_t stats_query_interval_ms_ = 1000;
int64_t last_stats_query_time_ms_ = 0;
std::vector<std::pair<int64_t, NetEqNetworkStatistics>> stats_;
std::vector<std::pair<int64_t, NetEqLifetimeStatistics>> lifetime_stats_;
size_t current_concealment_event_ = 1;
uint64_t voice_concealed_samples_until_last_event_ = 0;
std::vector<ConcealmentEvent> concealment_events_;
int64_t last_event_end_time_ms_ = 0;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
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