1 files changed, 81 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..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_