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diff --git a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h
new file mode 100644
index 0000000000..e2c1f3e4ba
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+++ b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h
@@ -0,0 +1,82 @@
+/*
+ *  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_AEC_DUMP_BASED_SIMULATOR_H_
+#define MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
+
+#include <fstream>
+#include <string>
+
+#include "modules/audio_processing/test/audio_processing_simulator.h"
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+namespace test {
+
+// Used to perform an audio processing simulation from an aec dump.
+class AecDumpBasedSimulator final : public AudioProcessingSimulator {
+ public:
+  AecDumpBasedSimulator(const SimulationSettings& settings,
+                        rtc::scoped_refptr<AudioProcessing> audio_processing,
+                        std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+  AecDumpBasedSimulator() = delete;
+  AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete;
+  AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete;
+
+  ~AecDumpBasedSimulator() override;
+
+  // Processes the messages in the aecdump file.
+  void Process() override;
+
+  // Analyzes the data in the aecdump file and reports the resulting statistics.
+  void Analyze() override;
+
+ private:
+  void HandleEvent(const webrtc::audioproc::Event& event_msg,
+                   int& num_forward_chunks_processed,
+                   int& init_index);
+  void HandleMessage(const webrtc::audioproc::Init& msg, int init_index);
+  void HandleMessage(const webrtc::audioproc::Stream& msg);
+  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
+  void HandleMessage(const webrtc::audioproc::Config& msg);
+  void HandleMessage(const webrtc::audioproc::RuntimeSetting& msg);
+  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
+  void PrepareReverseProcessStreamCall(
+      const webrtc::audioproc::ReverseStream& msg);
+  void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
+  void MaybeOpenCallOrderFile();
+  enum InterfaceType {
+    kFixedInterface,
+    kFloatInterface,
+    kNotSpecified,
+  };
+
+  FILE* dump_input_file_;
+  std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_;
+  std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_;
+  bool artificial_nearend_eof_reported_ = false;
+  InterfaceType interface_used_ = InterfaceType::kNotSpecified;
+  std::unique_ptr<std::ofstream> call_order_output_file_;
+  bool finished_processing_specified_init_block_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_