diff --git a/modules/audio_processing/main/test/process_test/Android.mk b/modules/audio_processing/main/test/process_test/Android.mk
index 269c689681..23080aab23 100644
--- a/modules/audio_processing/main/test/process_test/Android.mk
+++ b/modules/audio_processing/main/test/process_test/Android.mk
@@ -42,7 +42,7 @@ LOCAL_SHARED_LIBRARIES := \
     libstlport \
     libwebrtc_audio_preprocessing 
 
-LOCAL_MODULE:= webrtc-apmtest
+LOCAL_MODULE:= webrtc_apm_process_test
 
 include external/stlport/libstlport.mk
 include $(BUILD_EXECUTABLE)
diff --git a/modules/audio_processing/main/test/unit_test/Android.mk b/modules/audio_processing/main/test/unit_test/Android.mk
new file mode 100644
index 0000000000..b2029cfb4d
--- /dev/null
+++ b/modules/audio_processing/main/test/unit_test/Android.mk
@@ -0,0 +1,49 @@
+#  Copyright (c) 2011 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.
+
+LOCAL_PATH:= $(call my-dir)
+
+# apm test app
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_TAGS := tests
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+    unit_test.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    '-DWEBRTC_TARGET_PC' \
+    '-DWEBRTC_LINUX' \
+    '-DWEBRTC_THREAD_RR' \
+    '-DWEBRTC_ANDROID' \
+    '-DANDROID' 
+
+LOCAL_CPPFLAGS := 
+LOCAL_LDFLAGS :=
+LOCAL_C_INCLUDES := \
+    external/gtest/include \
+    $(LOCAL_PATH)/../../../../../system_wrappers/interface \
+    $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface \
+    $(LOCAL_PATH)/../../interface \
+    $(LOCAL_PATH)/../../../../interface \
+    $(LOCAL_PATH)/../../../../..
+
+LOCAL_STATIC_LIBRARIES := \
+    libgtest 
+
+LOCAL_SHARED_LIBRARIES := \
+    libutils \
+    libstlport \
+    libwebrtc_audio_preprocessing 
+
+LOCAL_MODULE:= webrtc_apm_unit_test
+
+include external/stlport/libstlport.mk
+include $(BUILD_EXECUTABLE)
diff --git a/modules/audio_processing/main/test/unit_test/unit_test.cc b/modules/audio_processing/main/test/unit_test/unit_test.cc
index 22ab417c94..eccd6ce101 100644
--- a/modules/audio_processing/main/test/unit_test/unit_test.cc
+++ b/modules/audio_processing/main/test/unit_test/unit_test.cc
@@ -10,15 +10,29 @@
 
 #include "unit_test.h"
 
+#include "tick_util.h"
+#include "cpu_features_wrapper.h"
 #include "event_wrapper.h"
 #include "module_common_types.h"
 #include "thread_wrapper.h"
 #include "trace.h"
 #include "signal_processing_library.h"
-
 #include "audio_processing.h"
 
-namespace webrtc {
+using webrtc::AudioProcessing;
+using webrtc::AudioFrame;
+using webrtc::GainControl;
+using webrtc::NoiseSuppression;
+using webrtc::EchoCancellation;
+using webrtc::TickInterval;
+using webrtc::TickTime;
+using webrtc::EventWrapper;
+using webrtc::Trace;
+using webrtc::LevelEstimator;
+using webrtc::EchoCancellation;
+using webrtc::EchoControlMobile;
+using webrtc::VoiceDetection;
+
 namespace {
 
 // If false, this will write out a new statistics file.
@@ -465,7 +479,7 @@ TEST_F(ApmTest, Process) {
                              sizeof(WebRtc_Word16),
                              render_audio._payloadDataLengthInSamples * 2,
                              far_file_);
-          if (read_count != 
+          if (read_count !=
               static_cast<size_t>
               (render_audio._payloadDataLengthInSamples * 2)) {
             break; // This is expected.
@@ -1011,4 +1025,3 @@ int main(int argc, char** argv) {
 
   return RUN_ALL_TESTS();
 }
-}
diff --git a/modules/audio_processing/main/test/unit_test/unit_test.h b/modules/audio_processing/main/test/unit_test/unit_test.h
index a68083b3d1..6de3bf916a 100644
--- a/modules/audio_processing/main/test/unit_test/unit_test.h
+++ b/modules/audio_processing/main/test/unit_test/unit_test.h
@@ -14,9 +14,9 @@
 #include <gtest/gtest.h>
 
 namespace webrtc {
-
 class AudioProcessing;
 class AudioFrame;
+}
 
 class ApmTest : public ::testing::Test {
  protected:
@@ -28,9 +28,8 @@ class ApmTest : public ::testing::Test {
   FILE* far_file_;
   FILE* near_file_;
   FILE* stat_file_;
-  AudioFrame* frame_;
-  AudioFrame* reverse_frame_;
+  webrtc::AudioFrame* frame_;
+  webrtc::AudioFrame* reverse_frame_;
 };
-} // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_TEST_UNIT_TEST_UNIT_TEST_H_