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diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
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+/*
+ * 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_