diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
new file mode 100644
index 000000000..c0d0a9688
--- /dev/null
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -0,0 +1,252 @@
+/*
+ *  Copyright (c) 2013 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 WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
+
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class MockEchoCancellation : public EchoCancellation {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+  MOCK_METHOD1(enable_drift_compensation,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_drift_compensation_enabled,
+      bool());
+  MOCK_METHOD1(set_device_sample_rate_hz,
+      int(int rate));
+  MOCK_CONST_METHOD0(device_sample_rate_hz,
+      int());
+  MOCK_METHOD1(set_stream_drift_samples,
+      void(int drift));
+  MOCK_CONST_METHOD0(stream_drift_samples,
+      int());
+  MOCK_METHOD1(set_suppression_level,
+      int(SuppressionLevel level));
+  MOCK_CONST_METHOD0(suppression_level,
+      SuppressionLevel());
+  MOCK_CONST_METHOD0(stream_has_echo,
+      bool());
+  MOCK_METHOD1(enable_metrics,
+      int(bool enable));
+  MOCK_CONST_METHOD0(are_metrics_enabled,
+      bool());
+  MOCK_METHOD1(GetMetrics,
+      int(Metrics* metrics));
+  MOCK_METHOD1(enable_delay_logging,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_delay_logging_enabled,
+      bool());
+  MOCK_METHOD2(GetDelayMetrics,
+      int(int* median, int* std));
+  MOCK_CONST_METHOD0(aec_core,
+      struct AecCore*());
+};
+
+class MockEchoControlMobile : public EchoControlMobile {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+  MOCK_METHOD1(set_routing_mode,
+      int(RoutingMode mode));
+  MOCK_CONST_METHOD0(routing_mode,
+      RoutingMode());
+  MOCK_METHOD1(enable_comfort_noise,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_comfort_noise_enabled,
+      bool());
+  MOCK_METHOD2(SetEchoPath,
+      int(const void* echo_path, size_t size_bytes));
+  MOCK_CONST_METHOD2(GetEchoPath,
+      int(void* echo_path, size_t size_bytes));
+};
+
+class MockGainControl : public GainControl {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+  MOCK_METHOD1(set_stream_analog_level,
+      int(int level));
+  MOCK_METHOD0(stream_analog_level,
+      int());
+  MOCK_METHOD1(set_mode,
+      int(Mode mode));
+  MOCK_CONST_METHOD0(mode,
+      Mode());
+  MOCK_METHOD1(set_target_level_dbfs,
+      int(int level));
+  MOCK_CONST_METHOD0(target_level_dbfs,
+      int());
+  MOCK_METHOD1(set_compression_gain_db,
+      int(int gain));
+  MOCK_CONST_METHOD0(compression_gain_db,
+      int());
+  MOCK_METHOD1(enable_limiter,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_limiter_enabled,
+      bool());
+  MOCK_METHOD2(set_analog_level_limits,
+      int(int minimum, int maximum));
+  MOCK_CONST_METHOD0(analog_level_minimum,
+      int());
+  MOCK_CONST_METHOD0(analog_level_maximum,
+      int());
+  MOCK_CONST_METHOD0(stream_is_saturated,
+      bool());
+};
+
+class MockHighPassFilter : public HighPassFilter {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+};
+
+class MockLevelEstimator : public LevelEstimator {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+  MOCK_METHOD0(RMS,
+      int());
+};
+
+class MockNoiseSuppression : public NoiseSuppression {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+  MOCK_METHOD1(set_level,
+      int(Level level));
+  MOCK_CONST_METHOD0(level,
+      Level());
+  MOCK_CONST_METHOD0(speech_probability,
+      float());
+};
+
+class MockVoiceDetection : public VoiceDetection {
+ public:
+  MOCK_METHOD1(Enable,
+      int(bool enable));
+  MOCK_CONST_METHOD0(is_enabled,
+      bool());
+  MOCK_CONST_METHOD0(stream_has_voice,
+      bool());
+  MOCK_METHOD1(set_stream_has_voice,
+      int(bool has_voice));
+  MOCK_METHOD1(set_likelihood,
+      int(Likelihood likelihood));
+  MOCK_CONST_METHOD0(likelihood,
+      Likelihood());
+  MOCK_METHOD1(set_frame_size_ms,
+      int(int size));
+  MOCK_CONST_METHOD0(frame_size_ms,
+      int());
+};
+
+class MockAudioProcessing : public AudioProcessing {
+ public:
+  MockAudioProcessing()
+      : echo_cancellation_(new MockEchoCancellation),
+        echo_control_mobile_(new MockEchoControlMobile),
+        gain_control_(new MockGainControl),
+        high_pass_filter_(new MockHighPassFilter),
+        level_estimator_(new MockLevelEstimator),
+        noise_suppression_(new MockNoiseSuppression),
+        voice_detection_(new MockVoiceDetection) {
+  }
+
+  virtual ~MockAudioProcessing() {
+  }
+
+  MOCK_METHOD0(Initialize,
+      int());
+  MOCK_METHOD1(set_sample_rate_hz,
+      int(int rate));
+  MOCK_CONST_METHOD0(sample_rate_hz,
+      int());
+  MOCK_METHOD2(set_num_channels,
+      int(int input_channels, int output_channels));
+  MOCK_CONST_METHOD0(num_input_channels,
+      int());
+  MOCK_CONST_METHOD0(num_output_channels,
+      int());
+  MOCK_METHOD1(set_num_reverse_channels,
+      int(int channels));
+  MOCK_CONST_METHOD0(num_reverse_channels,
+      int());
+  MOCK_METHOD1(ProcessStream,
+      int(AudioFrame* frame));
+  MOCK_METHOD1(AnalyzeReverseStream,
+      int(AudioFrame* frame));
+  MOCK_METHOD1(set_stream_delay_ms,
+      int(int delay));
+  MOCK_CONST_METHOD0(stream_delay_ms,
+      int());
+  MOCK_METHOD1(set_delay_offset_ms,
+      void(int offset));
+  MOCK_CONST_METHOD0(delay_offset_ms,
+      int());
+  MOCK_METHOD1(StartDebugRecording,
+      int(const char filename[kMaxFilenameSize]));
+  MOCK_METHOD0(StopDebugRecording,
+      int());
+  virtual MockEchoCancellation* echo_cancellation() const {
+    return echo_cancellation_.get();
+  }
+  virtual MockEchoControlMobile* echo_control_mobile() const {
+    return echo_control_mobile_.get();
+  }
+  virtual MockGainControl* gain_control() const {
+    return gain_control_.get();
+  }
+  virtual MockHighPassFilter* high_pass_filter() const {
+    return high_pass_filter_.get();
+  };
+  virtual MockLevelEstimator* level_estimator() const {
+    return level_estimator_.get();
+  };
+  virtual MockNoiseSuppression* noise_suppression() const {
+    return noise_suppression_.get();
+  };
+  virtual MockVoiceDetection* voice_detection() const {
+    return voice_detection_.get();
+  };
+  MOCK_METHOD0(TimeUntilNextProcess,
+      WebRtc_Word32());
+  MOCK_METHOD0(Process,
+      WebRtc_Word32());
+
+ private:
+  scoped_ptr<MockEchoCancellation> echo_cancellation_;
+  scoped_ptr<MockEchoControlMobile> echo_control_mobile_;
+  scoped_ptr<MockGainControl> gain_control_;
+  scoped_ptr<MockHighPassFilter> high_pass_filter_;
+  scoped_ptr<MockLevelEstimator> level_estimator_;
+  scoped_ptr<MockNoiseSuppression> noise_suppression_;
+  scoped_ptr<MockVoiceDetection> voice_detection_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
diff --git a/webrtc/test/fake_common.h b/webrtc/test/fake_common.h
new file mode 100644
index 000000000..a07480856
--- /dev/null
+++ b/webrtc/test/fake_common.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2013 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 WEBRTC_TEST_FAKE_COMMON_H_
+#define WEBRTC_TEST_FAKE_COMMON_H_
+
+// Borrowed from libjingle's talk/media/webrtc/fakewebrtccommon.h.
+
+#include "webrtc/typedefs.h"
+
+#define WEBRTC_STUB(method, args) \
+  virtual int method args OVERRIDE { return 0; }
+
+#define WEBRTC_STUB_CONST(method, args) \
+  virtual int method args const OVERRIDE { return 0; }
+
+#define WEBRTC_BOOL_STUB(method, args) \
+  virtual bool method args OVERRIDE { return true; }
+
+#define WEBRTC_VOID_STUB(method, args) \
+  virtual void method args OVERRIDE {}
+
+#define WEBRTC_FUNC(method, args) \
+  virtual int method args OVERRIDE
+
+#define WEBRTC_FUNC_CONST(method, args) \
+  virtual int method args const OVERRIDE
+
+#define WEBRTC_BOOL_FUNC(method, args) \
+  virtual bool method args OVERRIDE
+
+#define WEBRTC_VOID_FUNC(method, args) \
+  virtual void method args OVERRIDE
+
+#define WEBRTC_CHECK_CHANNEL(channel) \
+  if (channels_.find(channel) == channels_.end()) return -1;
+
+#define WEBRTC_ASSERT_CHANNEL(channel) \
+  ASSERT(channels_.find(channel) != channels_.end());
+
+#endif  // WEBRTC_TEST_FAKE_COMMON_H_
diff --git a/webrtc/typedefs.h b/webrtc/typedefs.h
index d6d401553..b056ab380 100644
--- a/webrtc/typedefs.h
+++ b/webrtc/typedefs.h
@@ -90,4 +90,17 @@ typedef uint16_t            WebRtc_UWord16;
 typedef uint32_t            WebRtc_UWord32;
 typedef uint64_t            WebRtc_UWord64;
 
+// Borrowed from Chromium's base/compiler_specific.h.
+// Annotate a virtual method indicating it must be overriding a virtual
+// method in the parent class.
+// Use like:
+//   virtual void foo() OVERRIDE;
+#if defined(_MSC_VER)
+#define OVERRIDE override
+#elif defined(__clang__)
+#define OVERRIDE override
+#else
+#define OVERRIDE
+#endif
+
 #endif  // WEBRTC_TYPEDEFS_H_
diff --git a/webrtc/voice_engine/include/mock/fake_voe_external_media.h b/webrtc/voice_engine/include/mock/fake_voe_external_media.h
new file mode 100644
index 000000000..8b608ec21
--- /dev/null
+++ b/webrtc/voice_engine/include/mock/fake_voe_external_media.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2013 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 WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
+#define WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
+
+#include <map>
+
+#include "webrtc/test/fake_common.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+
+namespace webrtc {
+
+class FakeVoEExternalMedia : public VoEExternalMedia {
+ public:
+  FakeVoEExternalMedia() {}
+  virtual ~FakeVoEExternalMedia() {}
+
+  WEBRTC_STUB(Release, ());
+  WEBRTC_FUNC(RegisterExternalMediaProcessing,
+      (int channel, ProcessingTypes type, VoEMediaProcess& processObject)) {
+    callback_map_[type] = &processObject;
+    return 0;
+  }
+  WEBRTC_FUNC(DeRegisterExternalMediaProcessing,
+      (int channel, ProcessingTypes type)) {
+    callback_map_.erase(type);
+    return 0;
+  }
+  WEBRTC_STUB(SetExternalRecordingStatus, (bool enable));
+  WEBRTC_STUB(SetExternalPlayoutStatus, (bool enable));
+  WEBRTC_STUB(ExternalRecordingInsertData,
+      (const WebRtc_Word16 speechData10ms[], int lengthSamples,
+       int samplingFreqHz, int current_delay_ms));
+  WEBRTC_STUB(ExternalPlayoutGetData,
+      (WebRtc_Word16 speechData10ms[], int samplingFreqHz,
+       int current_delay_ms, int& lengthSamples));
+  WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
+                              AudioFrame* frame));
+  WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
+
+  // Use this to trigger the Process() callback to a registered media processor.
+  // If |audio| is NULL, a zero array of the correct length will be forwarded.
+  void CallProcess(ProcessingTypes type, int16_t* audio,
+                   int samples_per_channel, int sample_rate_hz,
+                   int num_channels) {
+    const int length = samples_per_channel * num_channels;
+    scoped_array<int16_t> data;
+    if (!audio) {
+      data.reset(new int16_t[length]);
+      memset(data.get(), 0, length * sizeof(data[0]));
+      audio = data.get();
+    }
+
+    std::map<ProcessingTypes, VoEMediaProcess*>::const_iterator it =
+        callback_map_.find(type);
+    if (it != callback_map_.end()) {
+      it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
+                          num_channels == 2 ? true : false);
+    }
+  }
+
+ private:
+  std::map<ProcessingTypes, VoEMediaProcess*> callback_map_;
+};
+
+}  // namespace webrtc
+
+#endif  //  WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
diff --git a/webrtc/voice_engine/include/mock/mock_voe_volume_control.h b/webrtc/voice_engine/include/mock/mock_voe_volume_control.h
new file mode 100644
index 000000000..20b096968
--- /dev/null
+++ b/webrtc/voice_engine/include/mock/mock_voe_volume_control.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2013 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 WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
+#define WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class MockVoEVolumeControl : public VoEVolumeControl {
+ public:
+  MOCK_METHOD0(Release, int());
+  MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
+  MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
+  MOCK_METHOD1(SetSystemOutputMute, int(bool enable));
+  MOCK_METHOD1(GetSystemOutputMute, int(bool &enabled));
+  MOCK_METHOD1(SetMicVolume, int(unsigned int volume));
+  MOCK_METHOD1(GetMicVolume, int(unsigned int& volume));
+  MOCK_METHOD2(SetInputMute, int(int channel, bool enable));
+  MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled));
+  MOCK_METHOD1(SetSystemInputMute, int(bool enable));
+  MOCK_METHOD1(GetSystemInputMute, int(bool& enabled));
+  MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level));
+  MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level));
+  MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level));
+  MOCK_METHOD2(GetSpeechOutputLevelFullRange,
+               int(int channel, unsigned int& level));
+  MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling));
+  MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
+  MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right));
+  MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_