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Diffstat (limited to 'third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h')
-rw-r--r-- | third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h | 84 |
1 files changed, 84 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h new file mode 100644 index 0000000000..c59f727422 --- /dev/null +++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h @@ -0,0 +1,84 @@ +/* + * 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_ |