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diff --git a/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.h b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.h
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+/*
+ * Copyright (c) 2017 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_ECHO_CANCELLER_TEST_TOOLS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_ECHO_CANCELLER_TEST_TOOLS_H_
+
+#include <algorithm>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+
+// Randomizes the elements in a vector with values -32767.f:32767.f.
+void RandomizeSampleVector(Random* random_generator, rtc::ArrayView<float> v);
+
+// Randomizes the elements in a vector with values -amplitude:amplitude.
+void RandomizeSampleVector(Random* random_generator,
+ rtc::ArrayView<float> v,
+ float amplitude);
+
+// Class for delaying a signal a fixed number of samples.
+template <typename T>
+class DelayBuffer {
+ public:
+ explicit DelayBuffer(size_t delay) : buffer_(delay) {}
+ ~DelayBuffer() = default;
+
+ // Produces a delayed signal copy of x.
+ void Delay(rtc::ArrayView<const T> x, rtc::ArrayView<T> x_delayed);
+
+ private:
+ std::vector<T> buffer_;
+ size_t next_insert_index_ = 0;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_TEST_ECHO_CANCELLER_TEST_TOOLS_H_