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diff --git a/third_party/libwebrtc/modules/audio_processing/test/test_utils.h b/third_party/libwebrtc/modules/audio_processing/test/test_utils.h
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+++ b/third_party/libwebrtc/modules/audio_processing/test/test_utils.h
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+/*
+ *  Copyright (c) 2014 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_TEST_UTILS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
+
+#include <math.h>
+
+#include <iterator>
+#include <limits>
+#include <memory>
+#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/channel_buffer.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
+#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
+
+// Encapsulates samples and metadata for an integer frame.
+struct Int16FrameData {
+  // Max data size that matches the data size of the AudioFrame class, providing
+  // storage for 8 channels of 96 kHz data.
+  static const int kMaxDataSizeSamples = 7680;
+
+  Int16FrameData() {
+    sample_rate_hz = 0;
+    num_channels = 0;
+    samples_per_channel = 0;
+    data.fill(0);
+  }
+
+  void CopyFrom(const Int16FrameData& src) {
+    samples_per_channel = src.samples_per_channel;
+    sample_rate_hz = src.sample_rate_hz;
+    num_channels = src.num_channels;
+
+    const size_t length = samples_per_channel * num_channels;
+    RTC_CHECK_LE(length, kMaxDataSizeSamples);
+    memcpy(data.data(), src.data.data(), sizeof(int16_t) * length);
+  }
+  std::array<int16_t, kMaxDataSizeSamples> data;
+  int32_t sample_rate_hz;
+  size_t num_channels;
+  size_t samples_per_channel;
+};
+
+// Reads ChannelBuffers from a provided WavReader.
+class ChannelBufferWavReader final {
+ public:
+  explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file);
+  ~ChannelBufferWavReader();
+
+  ChannelBufferWavReader(const ChannelBufferWavReader&) = delete;
+  ChannelBufferWavReader& operator=(const ChannelBufferWavReader&) = delete;
+
+  // Reads data from the file according to the `buffer` format. Returns false if
+  // a full buffer can't be read from the file.
+  bool Read(ChannelBuffer<float>* buffer);
+
+ private:
+  std::unique_ptr<WavReader> file_;
+  std::vector<float> interleaved_;
+};
+
+// Writes ChannelBuffers to a provided WavWriter.
+class ChannelBufferWavWriter final {
+ public:
+  explicit ChannelBufferWavWriter(std::unique_ptr<WavWriter> file);
+  ~ChannelBufferWavWriter();
+
+  ChannelBufferWavWriter(const ChannelBufferWavWriter&) = delete;
+  ChannelBufferWavWriter& operator=(const ChannelBufferWavWriter&) = delete;
+
+  void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+  std::unique_ptr<WavWriter> file_;
+  std::vector<float> interleaved_;
+};
+
+// Takes a pointer to a vector. Allows appending the samples of channel buffers
+// to the given vector, by interleaving the samples and converting them to float
+// S16.
+class ChannelBufferVectorWriter final {
+ public:
+  explicit ChannelBufferVectorWriter(std::vector<float>* output);
+  ChannelBufferVectorWriter(const ChannelBufferVectorWriter&) = delete;
+  ChannelBufferVectorWriter& operator=(const ChannelBufferVectorWriter&) =
+      delete;
+  ~ChannelBufferVectorWriter();
+
+  // Creates an interleaved copy of `buffer`, converts the samples to float S16
+  // and appends the result to output_.
+  void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+  std::vector<float> interleaved_buffer_;
+  std::vector<float>* output_;
+};
+
+// Exits on failure; do not use in unit tests.
+FILE* OpenFile(absl::string_view filename, absl::string_view mode);
+
+void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz);
+
+template <typename T>
+void SetContainerFormat(int sample_rate_hz,
+                        size_t num_channels,
+                        Int16FrameData* frame,
+                        std::unique_ptr<ChannelBuffer<T> >* cb) {
+  SetFrameSampleRate(frame, sample_rate_hz);
+  frame->num_channels = num_channels;
+  cb->reset(new ChannelBuffer<T>(frame->samples_per_channel, num_channels));
+}
+
+template <typename T>
+float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
+  float mse = 0;
+  float mean = 0;
+  *variance = 0;
+  for (size_t i = 0; i < length; ++i) {
+    T error = ref[i] - test[i];
+    mse += error * error;
+    *variance += ref[i] * ref[i];
+    mean += ref[i];
+  }
+  mse /= length;
+  *variance /= length;
+  mean /= length;
+  *variance -= mean * mean;
+
+  float snr = 100;  // We assign 100 dB to the zero-error case.
+  if (mse > 0)
+    snr = 10 * log10(*variance / mse);
+  return snr;
+}
+
+// Returns a vector<T> parsed from whitespace delimited values in to_parse,
+// or an empty vector if the string could not be parsed.
+template <typename T>
+std::vector<T> ParseList(absl::string_view to_parse) {
+  std::vector<T> values;
+
+  std::istringstream str(  // no-presubmit-check TODO(webrtc:8982)
+      std::string{to_parse});
+  std::copy(
+      std::istream_iterator<T>(str),  // no-presubmit-check TODO(webrtc:8982)
+      std::istream_iterator<T>(),     // no-presubmit-check TODO(webrtc:8982)
+      std::back_inserter(values));
+
+  return values;
+}
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_