1 /* 2 * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_ 12 #define TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_ 13 14 #include <map> 15 #include <string> 16 17 #include "absl/strings/string_view.h" 18 #include "api/test/audio_quality_analyzer_interface.h" 19 #include "api/test/track_id_stream_info_map.h" 20 #include "api/units/time_delta.h" 21 #include "rtc_base/numerics/samples_stats_counter.h" 22 #include "rtc_base/synchronization/mutex.h" 23 #include "test/testsupport/perf_test.h" 24 25 namespace webrtc { 26 namespace webrtc_pc_e2e { 27 28 struct AudioStreamStats { 29 SamplesStatsCounter expand_rate; 30 SamplesStatsCounter accelerate_rate; 31 SamplesStatsCounter preemptive_rate; 32 SamplesStatsCounter speech_expand_rate; 33 SamplesStatsCounter average_jitter_buffer_delay_ms; 34 SamplesStatsCounter preferred_buffer_size_ms; 35 }; 36 37 class DefaultAudioQualityAnalyzer : public AudioQualityAnalyzerInterface { 38 public: 39 void Start(std::string test_case_name, 40 TrackIdStreamInfoMap* analyzer_helper) override; 41 void OnStatsReports( 42 absl::string_view pc_label, 43 const rtc::scoped_refptr<const RTCStatsReport>& report) override; 44 void Stop() override; 45 46 // Returns audio quality stats per stream label. 47 std::map<std::string, AudioStreamStats> GetAudioStreamsStats() const; 48 49 private: 50 struct StatsSample { 51 uint64_t total_samples_received = 0; 52 uint64_t concealed_samples = 0; 53 uint64_t removed_samples_for_acceleration = 0; 54 uint64_t inserted_samples_for_deceleration = 0; 55 uint64_t silent_concealed_samples = 0; 56 TimeDelta jitter_buffer_delay = TimeDelta::Zero(); 57 TimeDelta jitter_buffer_target_delay = TimeDelta::Zero(); 58 uint64_t jitter_buffer_emitted_count = 0; 59 }; 60 61 std::string GetTestCaseName(const std::string& stream_label) const; 62 void ReportResult(const std::string& metric_name, 63 const std::string& stream_label, 64 const SamplesStatsCounter& counter, 65 const std::string& unit, 66 webrtc::test::ImproveDirection improve_direction) const; 67 68 std::string test_case_name_; 69 TrackIdStreamInfoMap* analyzer_helper_; 70 71 mutable Mutex lock_; 72 std::map<std::string, AudioStreamStats> streams_stats_ RTC_GUARDED_BY(lock_); 73 std::map<std::string, StatsSample> last_stats_sample_ RTC_GUARDED_BY(lock_); 74 }; 75 76 } // namespace webrtc_pc_e2e 77 } // namespace webrtc 78 79 #endif // TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_ 80