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diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
new file mode 100644
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+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -0,0 +1,176 @@
+/*
+ *  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_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+
+#include <fstream>
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+enum LossModes {
+  kNoLoss,
+  kUniformLoss,
+  kGilbertElliotLoss,
+  kFixedLoss,
+  kLastLossMode
+};
+
+class LossModel {
+ public:
+  virtual ~LossModel() {}
+  virtual bool Lost(int now_ms) = 0;
+};
+
+class NoLoss : public LossModel {
+ public:
+  bool Lost(int now_ms) override;
+};
+
+class UniformLoss : public LossModel {
+ public:
+  UniformLoss(double loss_rate);
+  bool Lost(int now_ms) override;
+  void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
+
+ private:
+  double loss_rate_;
+};
+
+class GilbertElliotLoss : public LossModel {
+ public:
+  GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
+  ~GilbertElliotLoss() override;
+  bool Lost(int now_ms) override;
+
+ private:
+  // Prob. of losing current packet, when previous packet is lost.
+  double prob_trans_11_;
+  // Prob. of losing current packet, when previous packet is not lost.
+  double prob_trans_01_;
+  bool lost_last_;
+  std::unique_ptr<UniformLoss> uniform_loss_model_;
+};
+
+struct FixedLossEvent {
+  int start_ms;
+  int duration_ms;
+  FixedLossEvent(int start_ms, int duration_ms)
+      : start_ms(start_ms), duration_ms(duration_ms) {}
+};
+
+struct FixedLossEventCmp {
+  bool operator()(const FixedLossEvent& l_event,
+                  const FixedLossEvent& r_event) const {
+    return l_event.start_ms < r_event.start_ms;
+  }
+};
+
+class FixedLossModel : public LossModel {
+ public:
+  FixedLossModel(std::set<FixedLossEvent, FixedLossEventCmp> loss_events);
+  ~FixedLossModel() override;
+  bool Lost(int now_ms) override;
+
+ private:
+  std::set<FixedLossEvent, FixedLossEventCmp> loss_events_;
+  std::set<FixedLossEvent, FixedLossEventCmp>::iterator loss_events_it_;
+};
+
+class NetEqQualityTest : public ::testing::Test {
+ protected:
+  NetEqQualityTest(
+      int block_duration_ms,
+      int in_sampling_khz,
+      int out_sampling_khz,
+      const SdpAudioFormat& format,
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory =
+          webrtc::CreateBuiltinAudioDecoderFactory());
+  ~NetEqQualityTest() override;
+
+  void SetUp() override;
+
+  // EncodeBlock(...) does the following:
+  // 1. encodes a block of audio, saved in `in_data` and has a length of
+  // `block_size_samples` (samples per channel),
+  // 2. save the bit stream to `payload` of `max_bytes` bytes in size,
+  // 3. returns the length of the payload (in bytes),
+  virtual int EncodeBlock(int16_t* in_data,
+                          size_t block_size_samples,
+                          rtc::Buffer* payload,
+                          size_t max_bytes) = 0;
+
+  // PacketLost(...) determines weather a packet sent at an indicated time gets
+  // lost or not.
+  bool PacketLost();
+
+  // DecodeBlock() decodes a block of audio using the payload stored in
+  // `payload_` with the length of `payload_size_bytes_` (bytes). The decoded
+  // audio is to be stored in `out_data_`.
+  int DecodeBlock();
+
+  // Transmit() uses `rtp_generator_` to generate a packet and passes it to
+  // `neteq_`.
+  int Transmit();
+
+  // Runs encoding / transmitting / decoding.
+  void Simulate();
+
+  // Write to log file. Usage Log() << ...
+  std::ofstream& Log();
+
+  SdpAudioFormat audio_format_;
+  const size_t channels_;
+
+ private:
+  int decoded_time_ms_;
+  int decodable_time_ms_;
+  double drift_factor_;
+  int packet_loss_rate_;
+  const int block_duration_ms_;
+  const int in_sampling_khz_;
+  const int out_sampling_khz_;
+
+  // Number of samples per channel in a frame.
+  const size_t in_size_samples_;
+
+  size_t payload_size_bytes_;
+  size_t max_payload_bytes_;
+
+  std::unique_ptr<InputAudioFile> in_file_;
+  std::unique_ptr<AudioSink> output_;
+  std::ofstream log_file_;
+
+  std::unique_ptr<RtpGenerator> rtp_generator_;
+  std::unique_ptr<NetEq> neteq_;
+  std::unique_ptr<LossModel> loss_model_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  rtc::Buffer payload_;
+  AudioFrame out_frame_;
+  RTPHeader rtp_header_;
+
+  size_t total_payload_size_bytes_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_