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diff --git a/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.h b/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.h
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+++ b/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.h
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+/*
+ *  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 MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+// Collects statistics about the API call durations.
+class ApiCallStatistics {
+ public:
+  enum class CallType { kRender, kCapture };
+
+  // Adds a new datapoint.
+  void Add(int64_t duration_nanos, CallType call_type);
+
+  // Prints out a report of the statistics.
+  void PrintReport() const;
+
+  // Writes the call information to a file.
+  void WriteReportToFile(absl::string_view filename) const;
+
+ private:
+  struct CallData {
+    CallData(int64_t duration_nanos, CallType call_type);
+    int64_t duration_nanos;
+    CallType call_type;
+  };
+  std::vector<CallData> calls_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_