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diff --git a/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h
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+++ b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 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_PERFORMANCE_TIMER_H_
+#define MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+class PerformanceTimer {
+ public:
+  explicit PerformanceTimer(int num_frames_to_process);
+  ~PerformanceTimer();
+
+  void StartTimer();
+  void StopTimer();
+
+  double GetDurationAverage() const;
+  double GetDurationStandardDeviation() const;
+
+  // These methods are the same as those above, but they ignore the first
+  // `number_of_warmup_samples` measurements.
+  double GetDurationAverage(size_t number_of_warmup_samples) const;
+  double GetDurationStandardDeviation(size_t number_of_warmup_samples) const;
+
+ private:
+  webrtc::Clock* clock_;
+  absl::optional<int64_t> start_timestamp_us_;
+  std::vector<int64_t> timestamps_us_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_