diff options
Diffstat (limited to 'third_party/libwebrtc/test/pc/e2e/analyzer/audio')
| -rw-r--r-- | third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc | 175 | ||||
| -rw-r--r-- | third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h | 81 |
2 files changed, 256 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc new file mode 100644 index 0000000000..98d0c533c2 --- /dev/null +++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc @@ -0,0 +1,175 @@ +/* + * 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. + */ + +#include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h" + +#include "api/stats/rtc_stats.h" +#include "api/stats/rtcstats_objects.h" +#include "api/test/metrics/metric.h" +#include "api/test/track_id_stream_info_map.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "test/pc/e2e/metric_metadata_keys.h" + +namespace webrtc { +namespace webrtc_pc_e2e { + +using ::webrtc::test::ImprovementDirection; +using ::webrtc::test::Unit; + +DefaultAudioQualityAnalyzer::DefaultAudioQualityAnalyzer( + test::MetricsLogger* const metrics_logger) + : metrics_logger_(metrics_logger) { + RTC_CHECK(metrics_logger_); +} + +void DefaultAudioQualityAnalyzer::Start(std::string test_case_name, + TrackIdStreamInfoMap* analyzer_helper) { + test_case_name_ = std::move(test_case_name); + analyzer_helper_ = analyzer_helper; +} + +void DefaultAudioQualityAnalyzer::OnStatsReports( + absl::string_view pc_label, + const rtc::scoped_refptr<const RTCStatsReport>& report) { + auto stats = report->GetStatsOfType<RTCInboundRTPStreamStats>(); + + for (auto& stat : stats) { + if (!stat->kind.is_defined() || + !(*stat->kind == RTCMediaStreamTrackKind::kAudio)) { + continue; + } + + StatsSample sample; + sample.total_samples_received = + stat->total_samples_received.ValueOrDefault(0ul); + sample.concealed_samples = stat->concealed_samples.ValueOrDefault(0ul); + sample.removed_samples_for_acceleration = + stat->removed_samples_for_acceleration.ValueOrDefault(0ul); + sample.inserted_samples_for_deceleration = + stat->inserted_samples_for_deceleration.ValueOrDefault(0ul); + sample.silent_concealed_samples = + stat->silent_concealed_samples.ValueOrDefault(0ul); + sample.jitter_buffer_delay = + TimeDelta::Seconds(stat->jitter_buffer_delay.ValueOrDefault(0.)); + sample.jitter_buffer_target_delay = + TimeDelta::Seconds(stat->jitter_buffer_target_delay.ValueOrDefault(0.)); + sample.jitter_buffer_emitted_count = + stat->jitter_buffer_emitted_count.ValueOrDefault(0ul); + + TrackIdStreamInfoMap::StreamInfo stream_info = + analyzer_helper_->GetStreamInfoFromTrackId(*stat->track_identifier); + + MutexLock lock(&lock_); + stream_info_.emplace(stream_info.stream_label, stream_info); + StatsSample prev_sample = last_stats_sample_[stream_info.stream_label]; + RTC_CHECK_GE(sample.total_samples_received, + prev_sample.total_samples_received); + double total_samples_diff = static_cast<double>( + sample.total_samples_received - prev_sample.total_samples_received); + if (total_samples_diff == 0) { + return; + } + + AudioStreamStats& audio_stream_stats = + streams_stats_[stream_info.stream_label]; + audio_stream_stats.expand_rate.AddSample( + (sample.concealed_samples - prev_sample.concealed_samples) / + total_samples_diff); + audio_stream_stats.accelerate_rate.AddSample( + (sample.removed_samples_for_acceleration - + prev_sample.removed_samples_for_acceleration) / + total_samples_diff); + audio_stream_stats.preemptive_rate.AddSample( + (sample.inserted_samples_for_deceleration - + prev_sample.inserted_samples_for_deceleration) / + total_samples_diff); + + int64_t speech_concealed_samples = + sample.concealed_samples - sample.silent_concealed_samples; + int64_t prev_speech_concealed_samples = + prev_sample.concealed_samples - prev_sample.silent_concealed_samples; + audio_stream_stats.speech_expand_rate.AddSample( + (speech_concealed_samples - prev_speech_concealed_samples) / + total_samples_diff); + + int64_t jitter_buffer_emitted_count_diff = + sample.jitter_buffer_emitted_count - + prev_sample.jitter_buffer_emitted_count; + if (jitter_buffer_emitted_count_diff > 0) { + TimeDelta jitter_buffer_delay_diff = + sample.jitter_buffer_delay - prev_sample.jitter_buffer_delay; + TimeDelta jitter_buffer_target_delay_diff = + sample.jitter_buffer_target_delay - + prev_sample.jitter_buffer_target_delay; + audio_stream_stats.average_jitter_buffer_delay_ms.AddSample( + jitter_buffer_delay_diff.ms<double>() / + jitter_buffer_emitted_count_diff); + audio_stream_stats.preferred_buffer_size_ms.AddSample( + jitter_buffer_target_delay_diff.ms<double>() / + jitter_buffer_emitted_count_diff); + } + + last_stats_sample_[stream_info.stream_label] = sample; + } +} + +std::string DefaultAudioQualityAnalyzer::GetTestCaseName( + const std::string& stream_label) const { + return test_case_name_ + "/" + stream_label; +} + +void DefaultAudioQualityAnalyzer::Stop() { + MutexLock lock(&lock_); + for (auto& item : streams_stats_) { + const TrackIdStreamInfoMap::StreamInfo& stream_info = + stream_info_[item.first]; + // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey. + std::map<std::string, std::string> metric_metadata{ + {MetricMetadataKey::kAudioStreamMetadataKey, item.first}, + {MetricMetadataKey::kPeerMetadataKey, stream_info.receiver_peer}, + {MetricMetadataKey::kReceiverMetadataKey, stream_info.receiver_peer}, + {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}}; + + metrics_logger_->LogMetric("expand_rate", GetTestCaseName(item.first), + item.second.expand_rate, Unit::kUnitless, + ImprovementDirection::kSmallerIsBetter, + metric_metadata); + metrics_logger_->LogMetric("accelerate_rate", GetTestCaseName(item.first), + item.second.accelerate_rate, Unit::kUnitless, + ImprovementDirection::kSmallerIsBetter, + metric_metadata); + metrics_logger_->LogMetric("preemptive_rate", GetTestCaseName(item.first), + item.second.preemptive_rate, Unit::kUnitless, + ImprovementDirection::kSmallerIsBetter, + metric_metadata); + metrics_logger_->LogMetric( + "speech_expand_rate", GetTestCaseName(item.first), + item.second.speech_expand_rate, Unit::kUnitless, + ImprovementDirection::kSmallerIsBetter, metric_metadata); + metrics_logger_->LogMetric( + "average_jitter_buffer_delay_ms", GetTestCaseName(item.first), + item.second.average_jitter_buffer_delay_ms, Unit::kMilliseconds, + ImprovementDirection::kNeitherIsBetter, metric_metadata); + metrics_logger_->LogMetric( + "preferred_buffer_size_ms", GetTestCaseName(item.first), + item.second.preferred_buffer_size_ms, Unit::kMilliseconds, + ImprovementDirection::kNeitherIsBetter, metric_metadata); + } +} + +std::map<std::string, AudioStreamStats> +DefaultAudioQualityAnalyzer::GetAudioStreamsStats() const { + MutexLock lock(&lock_); + return streams_stats_; +} + +} // namespace webrtc_pc_e2e +} // namespace webrtc 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..9e427afed8 --- /dev/null +++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h @@ -0,0 +1,81 @@ +/* + * 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, 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_ |