Seperate NetEq stats getter to use in other tools.

Bug: webrtc:9147
Change-Id: I251618bbb542d89b3d38c3ea424b1e55c0a5f2b2
Reviewed-on: https://webrtc-review.googlesource.com/69806
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22971}

modules/audio_coding/neteq/tools/neteq_stats_getter.h

@@ -0,0 +1,93 @@
+/*
+ *  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 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_;
+  }
+
+  Stats AverageStats() const;
+
+ private:
+  std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer_;
+  size_t counter_ = 0;
+  std::vector<NetEqNetworkStatistics> 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_