| /* |
| * 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 <memory> |
| #include <string> |
| #include <vector> |
| |
| #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_ |