diff --git a/apps/CtsVerifier/Android.mk b/apps/CtsVerifier/Android.mk
index 364f13e..724b4d7 100644
--- a/apps/CtsVerifier/Android.mk
+++ b/apps/CtsVerifier/Android.mk
@@ -42,7 +42,7 @@
 
 LOCAL_AAPT_FLAGS += --version-name "5.0_r1.91 $(BUILD_NUMBER)"
 
-LOCAL_JNI_SHARED_LIBRARIES := libctsverifier_jni
+LOCAL_JNI_SHARED_LIBRARIES := libctsverifier_jni libaudioloopback_jni
 
 LOCAL_PROGUARD_FLAG_FILES := proguard.flags
 
diff --git a/apps/CtsVerifier/AndroidManifest.xml b/apps/CtsVerifier/AndroidManifest.xml
index 5137761..9c8f7b6 100644
--- a/apps/CtsVerifier/AndroidManifest.xml
+++ b/apps/CtsVerifier/AndroidManifest.xml
@@ -1559,6 +1559,18 @@
             -->
         </activity>
 
+        <activity android:name=".audio.AudioLoopbackActivity"
+                  android:label="@string/audio_loopback_test">
+            <intent-filter>
+                <action android:name="android.intent.action.MAIN" />
+                <category android:name="android.cts.intent.category.MANUAL_TEST" />
+            </intent-filter>
+            <meta-data android:name="test_category" android:value="@string/test_category_audio" />
+            <meta-data android:name="test_required_features" android:value="android.hardware.microphone" />
+            <meta-data android:name="test_excluded_features" android:value="android.hardware.type.watch" />
+            <meta-data android:name="test_excluded_features" android:value="android.hardware.type.television" />
+        </activity>
+
         <service android:name=".tv.MockTvInputService"
             android:permission="android.permission.BIND_TV_INPUT">
             <intent-filter>
diff --git a/apps/CtsVerifier/jni/audio_loopback/Android.mk b/apps/CtsVerifier/jni/audio_loopback/Android.mk
new file mode 100644
index 0000000..3dfbc34
--- /dev/null
+++ b/apps/CtsVerifier/jni/audio_loopback/Android.mk
@@ -0,0 +1,28 @@
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE      := libaudioloopback_jni
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES   := \
+	sles.cpp \
+	jni_sles.c
+
+LOCAL_C_INCLUDES := \
+        system/media/audio_utils/include \
+        frameworks/wilhelm/include
+
+LOCAL_SHARED_LIBRARIES := \
+	libutils \
+	libcutils \
+	libOpenSLES \
+	libnbaio \
+	liblog \
+	libaudioutils
+
+LOCAL_PRELINK_MODULE := false
+
+LOCAL_LDFLAGS := -Wl,--hash-style=sysv
+LOCAL_CFLAGS := -DSTDC_HEADERS
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/apps/CtsVerifier/jni/audio_loopback/jni_sles.c b/apps/CtsVerifier/jni/audio_loopback/jni_sles.c
new file mode 100644
index 0000000..a865078
--- /dev/null
+++ b/apps/CtsVerifier/jni/audio_loopback/jni_sles.c
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <android/log.h>
+#include "sles.h"
+#include "jni_sles.h"
+#include <stdio.h>
+#include <stddef.h>
+
+/////
+JNIEXPORT jlong JNICALL Java_com_android_cts_verifier_audio_NativeAudioThread_slesInit
+  (JNIEnv *env __unused, jobject obj __unused, jint samplingRate, jint frameCount, jint micSource) {
+
+    sles_data * pSles = NULL;
+
+    if (slesInit(&pSles, samplingRate, frameCount, micSource) != SLES_FAIL) {
+
+        return (long)pSles;
+    }
+    // FIXME This should be stored as a (long) field in the object,
+    //       so that incorrect Java code could not synthesize a bad sles pointer.
+    return 0;
+}
+
+JNIEXPORT jint JNICALL Java_com_android_cts_verifier_audio_NativeAudioThread_slesProcessNext
+  (JNIEnv *env __unused, jobject obj __unused, jlong sles, jdoubleArray samplesArray,
+          jlong offset) {
+    sles_data * pSles= (sles_data*) ((long)sles);
+
+    long maxSamples = (*env)->GetArrayLength(env, samplesArray);
+    double *pSamples = (*env)->GetDoubleArrayElements(env, samplesArray,0);
+
+    long availableSamples = maxSamples-offset;
+    double *pCurrentSample = pSamples+offset;
+
+    SLES_PRINTF("jni slesProcessNext pSles:%p, currentSample %p, availableSamples %ld ", pSles,
+            pCurrentSample, availableSamples);
+
+    int samplesRead = slesProcessNext(pSles, pCurrentSample, availableSamples);
+
+    return samplesRead;
+}
+
+JNIEXPORT jint JNICALL Java_com_android_cts_verifier_audio_NativeAudioThread_slesDestroy
+  (JNIEnv *env __unused, jobject obj __unused, jlong sles) {
+    sles_data * pSles= (sles_data*) ((long) sles);
+
+    int status = slesDestroy(&pSles);
+
+    return status;
+}
diff --git a/apps/CtsVerifier/jni/audio_loopback/jni_sles.h b/apps/CtsVerifier/jni/audio_loopback/jni_sles.h
new file mode 100644
index 0000000..7bff040
--- /dev/null
+++ b/apps/CtsVerifier/jni/audio_loopback/jni_sles.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <jni.h>
+
+#ifndef _Included_org_drrickorang_loopback_jni
+#define _Included_org_drrickorang_loopback_jni
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+////////////////////////
+JNIEXPORT jlong JNICALL Java_com_android_cts_verifier_audio_NativeAudioThread_slesInit
+  (JNIEnv *, jobject, jint, jint, jint );
+
+JNIEXPORT jint JNICALL Java_com_android_cts_verifier_audio_NativeAudioThread_slesProcessNext
+  (JNIEnv *, jobject , jlong, jdoubleArray, jlong );
+
+JNIEXPORT jint JNICALL Java_com_android_cts_verifier_audio_NativeAudioThread_slesDestroy
+  (JNIEnv *, jobject , jlong );
+
+
+#ifdef __cplusplus
+}
+#endif
+#endif //_Included_org_drrickorang_loopback_jni
diff --git a/apps/CtsVerifier/jni/audio_loopback/sles.cpp b/apps/CtsVerifier/jni/audio_loopback/sles.cpp
new file mode 100644
index 0000000..7859d35
--- /dev/null
+++ b/apps/CtsVerifier/jni/audio_loopback/sles.cpp
@@ -0,0 +1,655 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+////////////////////////////////////////////
+/// Actual sles functions.
+
+
+// Test program to record from default audio input and playback to default audio output.
+// It will generate feedback (Larsen effect) if played through on-device speakers,
+// or acts as a delay if played through headset.
+
+#include "sles.h"
+#include <stdio.h>
+#include <stdlib.h>
+#include <stddef.h>
+
+#include <assert.h>
+#include <pthread.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+
+int slesInit(sles_data ** ppSles, int samplingRate, int frameCount, int micSource) {
+    int status = SLES_FAIL;
+    if (ppSles != NULL) {
+        sles_data * pSles = (sles_data*) calloc(1, sizeof (sles_data));
+
+        SLES_PRINTF("malloc %d bytes at %p",sizeof(sles_data), pSles);
+        *ppSles = pSles;
+        if (pSles != NULL)
+        {
+            SLES_PRINTF("creating server. Sampling rate =%d, frame count = %d",samplingRate,
+                    frameCount);
+            status = slesCreateServer(pSles, samplingRate, frameCount, micSource);
+            SLES_PRINTF("slesCreateServer =%d",status);
+        }
+    }
+    return status;
+}
+int slesDestroy(sles_data ** ppSles) {
+    int status = SLES_FAIL;
+    if (ppSles != NULL) {
+        slesDestroyServer(*ppSles);
+
+        if (*ppSles != NULL)
+        {
+            free(*ppSles);
+            *ppSles = 0;
+        }
+        status = SLES_SUCCESS;
+    }
+    return status;
+}
+
+#define ASSERT_EQ(x, y) do { if ((x) == (y)) ; else { fprintf(stderr, "0x%x != 0x%x\n", \
+        (unsigned) (x), (unsigned) (y)); assert((x) == (y)); } } while (0)
+
+
+// Called after audio recorder fills a buffer with data
+static void recorderCallback(SLAndroidSimpleBufferQueueItf caller __unused, void *context) {
+    sles_data *pSles = (sles_data*) context;
+    if (pSles != NULL) {
+
+
+
+        SLresult result;
+
+        pthread_mutex_lock(&(pSles->mutex));
+        //ee  SLES_PRINTF("<R");
+
+        // We should only be called when a recording buffer is done
+        assert(pSles->rxFront <= pSles->rxBufCount);
+        assert(pSles->rxRear <= pSles->rxBufCount);
+        assert(pSles->rxFront != pSles->rxRear);
+        char *buffer = pSles->rxBuffers[pSles->rxFront];
+
+        // Remove buffer from record queue
+        if (++pSles->rxFront > pSles->rxBufCount) {
+            pSles->rxFront = 0;
+        }
+
+        ssize_t actual = audio_utils_fifo_write(&(pSles->fifo), buffer,
+                (size_t) pSles->bufSizeInFrames);
+        if (actual != (ssize_t) pSles->bufSizeInFrames) {
+            write(1, "?", 1);
+        }
+
+        // This is called by a realtime (SCHED_FIFO) thread,
+        // and it is unsafe to do I/O as it could block for unbounded time.
+        // Flash filesystem is especially notorious for blocking.
+        if (pSles->fifo2Buffer != NULL) {
+            actual = audio_utils_fifo_write(&(pSles->fifo2), buffer,
+                    (size_t) pSles->bufSizeInFrames);
+            if (actual != (ssize_t) pSles->bufSizeInFrames) {
+                write(1, "?", 1);
+            }
+        }
+
+        // Enqueue this same buffer for the recorder to fill again.
+        result = (*(pSles->recorderBufferQueue))->Enqueue(pSles->recorderBufferQueue, buffer,
+                pSles->bufSizeInBytes);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+        // Update our model of the record queue
+        SLuint32 rxRearNext = pSles->rxRear+1;
+        if (rxRearNext > pSles->rxBufCount) {
+            rxRearNext = 0;
+        }
+        assert(rxRearNext != pSles->rxFront);
+        pSles->rxBuffers[pSles->rxRear] = buffer;
+        pSles->rxRear = rxRearNext;
+
+
+
+        //ee  SLES_PRINTF("r>");
+        pthread_mutex_unlock(&(pSles->mutex));
+
+    } //pSles not null
+}
+
+
+// Called after audio player empties a buffer of data
+static void playerCallback(SLBufferQueueItf caller __unused, void *context) {
+    sles_data *pSles = (sles_data*) context;
+    if (pSles != NULL) {
+
+        SLresult result;
+
+        pthread_mutex_lock(&(pSles->mutex));
+        //ee  SLES_PRINTF("<P");
+
+        // Get the buffer that just finished playing
+        assert(pSles->txFront <= pSles->txBufCount);
+        assert(pSles->txRear <= pSles->txBufCount);
+        assert(pSles->txFront != pSles->txRear);
+        char *buffer = pSles->txBuffers[pSles->txFront];
+        if (++pSles->txFront > pSles->txBufCount) {
+            pSles->txFront = 0;
+        }
+
+
+        ssize_t actual = audio_utils_fifo_read(&(pSles->fifo), buffer, pSles->bufSizeInFrames);
+        if (actual != (ssize_t) pSles->bufSizeInFrames) {
+            write(1, "/", 1);
+            // on underrun from pipe, substitute silence
+            memset(buffer, 0, pSles->bufSizeInFrames * pSles->channels * sizeof(short));
+        }
+
+        if (pSles->injectImpulse == -1) {
+            // Experimentally, a single frame impulse was insufficient to trigger feedback.
+            // Also a Nyquist frequency signal was also insufficient, probably because
+            // the response of output and/or input path was not adequate at high frequencies.
+            // This short burst of a few cycles of square wave at Nyquist/4 was found to work well.
+            for (unsigned i = 0; i < pSles->bufSizeInFrames / 8; i += 8) {
+                for (int j = 0; j < 8; j++) {
+                    for (unsigned k = 0; k < pSles->channels; k++) {
+                        ((short *)buffer)[(i+j)*pSles->channels+k] = j < 4 ? 0x7FFF : 0x8000;
+                    }
+                }
+            }
+            pSles->injectImpulse = 0;
+        }
+
+        // Enqueue the filled buffer for playing
+        result = (*(pSles->playerBufferQueue))->Enqueue(pSles->playerBufferQueue, buffer,
+                pSles->bufSizeInBytes);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+        // Update our model of the player queue
+        assert(pSles->txFront <= pSles->txBufCount);
+        assert(pSles->txRear <= pSles->txBufCount);
+        SLuint32 txRearNext = pSles->txRear+1;
+        if (txRearNext > pSles->txBufCount) {
+            txRearNext = 0;
+        }
+        assert(txRearNext != pSles->txFront);
+        pSles->txBuffers[pSles->txRear] = buffer;
+        pSles->txRear = txRearNext;
+
+
+        //ee    SLES_PRINTF("p>");
+        pthread_mutex_unlock(&(pSles->mutex));
+
+    } //pSles not null
+}
+
+int slesCreateServer(sles_data *pSles, int samplingRate, int frameCount, int micSource) {
+    int status = SLES_FAIL;
+
+    if (pSles == NULL) {
+        return status;
+    }
+
+    //        adb shell slesTest_feedback -r1 -t1 -s48000 -f240 -i300 -e3 -o/sdcard/log.wav
+    //            r1 and t1 are the receive and transmit buffer counts, typically 1
+    //            s is the sample rate, typically 48000 or 44100
+    //            f is the frame count per buffer, typically 240 or 256
+    //            i is the number of milliseconds before impulse.  You may need to adjust this.
+    //            e is number of seconds to record
+    //            o is output .wav file name
+
+
+    //        // default values
+    //        SLuint32 rxBufCount = 1;     // -r#
+    //        SLuint32 txBufCount = 1;     // -t#
+    //        SLuint32 bufSizeInFrames = 240;  // -f#
+    //        SLuint32 channels = 1;       // -c#
+    //        SLuint32 sampleRate = 48000; // -s#
+    //        SLuint32 exitAfterSeconds = 3; // -e#
+    //        SLuint32 freeBufCount = 0;   // calculated
+    //        SLuint32 bufSizeInBytes = 0; // calculated
+    //        int injectImpulse = 300; // -i#i
+    //
+    //        // Storage area for the buffer queues
+    //        char **rxBuffers;
+    //        char **txBuffers;
+    //        char **freeBuffers;
+    //
+    //        // Buffer indices
+    //        SLuint32 rxFront;    // oldest recording
+    //        SLuint32 rxRear;     // next to be recorded
+    //        SLuint32 txFront;    // oldest playing
+    //        SLuint32 txRear;     // next to be played
+    //        SLuint32 freeFront;  // oldest free
+    //        SLuint32 freeRear;   // next to be freed
+    //
+    //        audio_utils_fifo fifo; //(*)
+    //        SLAndroidSimpleBufferQueueItf recorderBufferQueue;
+    //        SLBufferQueueItf playerBufferQueue;
+
+    // default values
+    pSles->rxBufCount = 1;     // -r#
+    pSles->txBufCount = 1;     // -t#
+    pSles->bufSizeInFrames = frameCount;//240;  // -f#
+    pSles->channels = 1;       // -c#
+    pSles->sampleRate = samplingRate;//48000; // -s#
+    pSles->exitAfterSeconds = 3; // -e#
+    pSles->freeBufCount = 0;   // calculated
+    pSles->bufSizeInBytes = 0; // calculated
+    pSles->injectImpulse = 300; // -i#i
+
+    // Storage area for the buffer queues
+    //        char **rxBuffers;
+    //        char **txBuffers;
+    //        char **freeBuffers;
+
+    // Buffer indices
+    pSles->rxFront;    // oldest recording
+    pSles->rxRear;     // next to be recorded
+    pSles->txFront;    // oldest playing
+    pSles->txRear;     // next to be played
+    pSles->freeFront;  // oldest free
+    pSles->freeRear;   // next to be freed
+
+    pSles->fifo; //(*)
+    pSles->fifo2Buffer = NULL;
+    pSles->recorderBufferQueue;
+    pSles->playerBufferQueue;
+
+    // compute total free buffers as -r plus -t
+    pSles->freeBufCount = pSles->rxBufCount + pSles->txBufCount;
+    // compute buffer size
+    pSles->bufSizeInBytes = pSles->channels * pSles->bufSizeInFrames * sizeof(short);
+
+    // Initialize free buffers
+    pSles->freeBuffers = (char **) calloc(pSles->freeBufCount+1, sizeof(char *));
+    unsigned j;
+    for (j = 0; j < pSles->freeBufCount; ++j) {
+        pSles->freeBuffers[j] = (char *) malloc(pSles->bufSizeInBytes);
+    }
+    pSles->freeFront = 0;
+    pSles->freeRear = pSles->freeBufCount;
+    pSles->freeBuffers[j] = NULL;
+
+    // Initialize record queue
+    pSles->rxBuffers = (char **) calloc(pSles->rxBufCount+1, sizeof(char *));
+    pSles->rxFront = 0;
+    pSles->rxRear = 0;
+
+    // Initialize play queue
+    pSles->txBuffers = (char **) calloc(pSles->txBufCount+1, sizeof(char *));
+    pSles->txFront = 0;
+    pSles->txRear = 0;
+
+    size_t frameSize = pSles->channels * sizeof(short);
+#define FIFO_FRAMES 1024
+    pSles->fifoBuffer = new short[FIFO_FRAMES * pSles->channels];
+    audio_utils_fifo_init(&(pSles->fifo), FIFO_FRAMES, frameSize, pSles->fifoBuffer);
+
+    //        SNDFILE *sndfile;
+    //        if (outFileName != NULL) {
+    // create .wav writer
+    //            SF_INFO info;
+    //            info.frames = 0;
+    //            info.samplerate = sampleRate;
+    //            info.channels = channels;
+    //            info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+    //            sndfile = sf_open(outFileName, SFM_WRITE, &info);
+    //            if (sndfile != NULL) {
+#define FIFO2_FRAMES 65536
+    pSles->fifo2Buffer = new short[FIFO2_FRAMES * pSles->channels];
+    audio_utils_fifo_init(&(pSles->fifo2), FIFO2_FRAMES, frameSize, pSles->fifo2Buffer);
+    //            } else {
+    //                fprintf(stderr, "sf_open failed\n");
+    //            }
+    //        } else {
+    //            sndfile = NULL;
+    //        }
+
+    SLresult result;
+
+    // create engine
+    pSles->engineObject;
+    result = slCreateEngine(&(pSles->engineObject), 0, NULL, 0, NULL, NULL);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->engineObject))->Realize(pSles->engineObject, SL_BOOLEAN_FALSE);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    SLEngineItf engineEngine;
+    result = (*(pSles->engineObject))->GetInterface(pSles->engineObject, SL_IID_ENGINE,
+            &engineEngine);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    // create output mix
+    pSles->outputmixObject;
+    result = (*engineEngine)->CreateOutputMix(engineEngine, &(pSles->outputmixObject), 0, NULL,
+            NULL);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->outputmixObject))->Realize(pSles->outputmixObject, SL_BOOLEAN_FALSE);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    // create an audio player with buffer queue source and output mix sink
+    SLDataSource audiosrc;
+    SLDataSink audiosnk;
+    SLDataFormat_PCM pcm;
+    SLDataLocator_OutputMix locator_outputmix;
+    SLDataLocator_BufferQueue locator_bufferqueue_tx;
+    locator_bufferqueue_tx.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
+    locator_bufferqueue_tx.numBuffers = pSles->txBufCount;
+    locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+    locator_outputmix.outputMix = pSles->outputmixObject;
+    pcm.formatType = SL_DATAFORMAT_PCM;
+    pcm.numChannels = pSles->channels;
+    pcm.samplesPerSec = pSles->sampleRate * 1000;
+    pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+    pcm.containerSize = 16;
+    pcm.channelMask = pSles->channels == 1 ? SL_SPEAKER_FRONT_CENTER :
+            (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT);
+    pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    audiosrc.pLocator = &locator_bufferqueue_tx;
+    audiosrc.pFormat = &pcm;
+    audiosnk.pLocator = &locator_outputmix;
+    audiosnk.pFormat = NULL;
+    pSles->playerObject = NULL;
+    pSles->recorderObject = NULL;
+    SLInterfaceID ids_tx[1] = {SL_IID_BUFFERQUEUE};
+    SLboolean flags_tx[1] = {SL_BOOLEAN_TRUE};
+    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &(pSles->playerObject),
+            &audiosrc, &audiosnk, 1, ids_tx, flags_tx);
+    if (SL_RESULT_CONTENT_UNSUPPORTED == result) {
+        fprintf(stderr, "Could not create audio player (result %x), check sample rate\n",
+                result);
+        SLES_PRINTF("ERROR: Could not create audio player (result %x), check sample rate\n",
+                result);
+        goto cleanup;
+    }
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->playerObject))->Realize(pSles->playerObject, SL_BOOLEAN_FALSE);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    SLPlayItf playerPlay;
+    result = (*(pSles->playerObject))->GetInterface(pSles->playerObject, SL_IID_PLAY,
+            &playerPlay);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->playerObject))->GetInterface(pSles->playerObject, SL_IID_BUFFERQUEUE,
+            &(pSles->playerBufferQueue));
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->playerBufferQueue))->RegisterCallback(pSles->playerBufferQueue,
+            playerCallback, pSles);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    // Enqueue some zero buffers for the player
+    for (j = 0; j < pSles->txBufCount; ++j) {
+
+        // allocate a free buffer
+        assert(pSles->freeFront != pSles->freeRear);
+        char *buffer = pSles->freeBuffers[pSles->freeFront];
+        if (++pSles->freeFront > pSles->freeBufCount) {
+            pSles->freeFront = 0;
+        }
+
+        // put on play queue
+        SLuint32 txRearNext = pSles->txRear + 1;
+        if (txRearNext > pSles->txBufCount) {
+            txRearNext = 0;
+        }
+        assert(txRearNext != pSles->txFront);
+        pSles->txBuffers[pSles->txRear] = buffer;
+        pSles->txRear = txRearNext;
+        result = (*(pSles->playerBufferQueue))->Enqueue(pSles->playerBufferQueue,
+                buffer, pSles->bufSizeInBytes);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    }
+
+    result = (*playerPlay)->SetPlayState(playerPlay, SL_PLAYSTATE_PLAYING);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    // Create an audio recorder with microphone device source and buffer queue sink.
+    // The buffer queue as sink is an Android-specific extension.
+
+    SLDataLocator_IODevice locator_iodevice;
+    SLDataLocator_AndroidSimpleBufferQueue locator_bufferqueue_rx;
+    locator_iodevice.locatorType = SL_DATALOCATOR_IODEVICE;
+    locator_iodevice.deviceType = SL_IODEVICE_AUDIOINPUT;
+    locator_iodevice.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT;
+    locator_iodevice.device = NULL;
+    audiosrc.pLocator = &locator_iodevice;
+    audiosrc.pFormat = NULL;
+    locator_bufferqueue_rx.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    locator_bufferqueue_rx.numBuffers = pSles->rxBufCount;
+    audiosnk.pLocator = &locator_bufferqueue_rx;
+    audiosnk.pFormat = &pcm;
+    {
+        SLInterfaceID ids_rx[2] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+                SL_IID_ANDROIDCONFIGURATION};
+        SLboolean flags_rx[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+        result = (*engineEngine)->CreateAudioRecorder(engineEngine, &(pSles->recorderObject),
+                &audiosrc, &audiosnk, 2, ids_rx, flags_rx);
+        if (SL_RESULT_SUCCESS != result) {
+            fprintf(stderr, "Could not create audio recorder (result %x), "
+                    "check sample rate and channel count\n", result);
+            status = SLES_FAIL;
+
+            SLES_PRINTF("ERROR: Could not create audio recorder (result %x), "
+                    "check sample rate and channel count\n", result);
+            goto cleanup;
+        }
+    }
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    {
+        /* Get the Android configuration interface which is explicit */
+        SLAndroidConfigurationItf configItf;
+        result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject,
+                SL_IID_ANDROIDCONFIGURATION, (void*)&configItf);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+        SLuint32 presetValue = micSource;
+        /* Use the configuration interface to configure the recorder before it's realized */
+        if (presetValue != SL_ANDROID_RECORDING_PRESET_NONE) {
+            result = (*configItf)->SetConfiguration(configItf, SL_ANDROID_KEY_RECORDING_PRESET,
+                    &presetValue, sizeof(SLuint32));
+            ASSERT_EQ(SL_RESULT_SUCCESS, result);
+        }
+
+    }
+
+    result = (*(pSles->recorderObject))->Realize(pSles->recorderObject, SL_BOOLEAN_FALSE);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    SLRecordItf recorderRecord;
+    result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject, SL_IID_RECORD,
+            &recorderRecord);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject,
+            SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &(pSles->recorderBufferQueue));
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    result = (*(pSles->recorderBufferQueue))->RegisterCallback(pSles->recorderBufferQueue,
+            recorderCallback, pSles);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    // Enqueue some empty buffers for the recorder
+    for (j = 0; j < pSles->rxBufCount; ++j) {
+
+        // allocate a free buffer
+        assert(pSles->freeFront != pSles->freeRear);
+        char *buffer = pSles->freeBuffers[pSles->freeFront];
+        if (++pSles->freeFront > pSles->freeBufCount) {
+            pSles->freeFront = 0;
+        }
+
+        // put on record queue
+        SLuint32 rxRearNext = pSles->rxRear + 1;
+        if (rxRearNext > pSles->rxBufCount) {
+            rxRearNext = 0;
+        }
+        assert(rxRearNext != pSles->rxFront);
+        pSles->rxBuffers[pSles->rxRear] = buffer;
+        pSles->rxRear = rxRearNext;
+        result = (*(pSles->recorderBufferQueue))->Enqueue(pSles->recorderBufferQueue,
+                buffer, pSles->bufSizeInBytes);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    }
+
+    // Kick off the recorder
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_RECORDING);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    // Tear down the objects and exit
+    status = SLES_SUCCESS;
+    cleanup:
+    SLES_PRINTF("Finished initialization with status: %d", status);
+
+    return status;
+}
+
+int slesProcessNext(sles_data *pSles, double *pSamples, long maxSamples) {
+    //int status = SLES_FAIL;
+
+    SLES_PRINTF("slesProcessNext: pSles = %p, currentSample: %p,  maxSamples = %ld", pSles,
+            pSamples, maxSamples);
+
+    int samplesRead = 0;
+
+    int currentSample = 0;
+    double *pCurrentSample = pSamples;
+    int maxValue = 32768;
+
+    if (pSles == NULL) {
+        return samplesRead;
+    }
+
+    SLresult result;
+    for (int i = 0; i < 10; i++) {
+        usleep(100000);
+        if (pSles->fifo2Buffer != NULL) {
+            for (;;) {
+                short buffer[pSles->bufSizeInFrames * pSles->channels];
+                ssize_t actual = audio_utils_fifo_read(&(pSles->fifo2), buffer,
+                        pSles->bufSizeInFrames);
+                if (actual <= 0)
+                    break;
+                {
+                    for (int jj =0; jj<actual && currentSample < maxSamples; jj++) {
+                        *(pCurrentSample++) = ((double)buffer[jj])/maxValue;
+                        currentSample++;
+                    }
+                }
+                samplesRead +=actual;
+            }
+        }
+        if (pSles->injectImpulse > 0) {
+            if (pSles->injectImpulse <= 100) {
+                pSles->injectImpulse = -1;
+                write(1, "I", 1);
+            } else {
+                if ((pSles->injectImpulse % 1000) < 100) {
+                    write(1, "i", 1);
+                }
+                pSles->injectImpulse -= 100;
+            }
+        } else if (i == 9) {
+            write(1, ".", 1);
+        }
+    }
+    SLBufferQueueState playerBQState;
+    result = (*(pSles->playerBufferQueue))->GetState(pSles->playerBufferQueue,
+            &playerBQState);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    SLAndroidSimpleBufferQueueState recorderBQState;
+    result = (*(pSles->recorderBufferQueue))->GetState(pSles->recorderBufferQueue,
+            &recorderBQState);
+    ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+    SLES_PRINTF("End of slesProcessNext: pSles = %p, samplesRead = %d, maxSamples= %ld", pSles,
+            samplesRead, maxSamples);
+
+    return samplesRead;
+}
+
+int slesDestroyServer(sles_data *pSles) {
+    int status = SLES_FAIL;
+
+    SLES_PRINTF("Start slesDestroyServer: pSles = %p", pSles);
+    if (pSles == NULL) {
+        return status;
+    }
+
+    if (NULL != pSles->playerObject) {
+
+        SLES_PRINTF("stopping player...");
+        SLPlayItf playerPlay;
+        SLresult result = (*(pSles->playerObject))->GetInterface(pSles->playerObject,
+                SL_IID_PLAY, &playerPlay);
+
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+        //stop player and recorder if they exist
+        result = (*playerPlay)->SetPlayState(playerPlay, SL_PLAYSTATE_STOPPED);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    }
+
+    if (NULL != pSles->recorderObject) {
+        SLES_PRINTF("stopping recorder...");
+        SLRecordItf recorderRecord;
+        SLresult result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject,
+                SL_IID_RECORD, &recorderRecord);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+
+        result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+        ASSERT_EQ(SL_RESULT_SUCCESS, result);
+    }
+
+    usleep(1000);
+
+    audio_utils_fifo_deinit(&(pSles->fifo));
+    delete[] pSles->fifoBuffer;
+
+    SLES_PRINTF("slesDestroyServer 2");
+
+    //        if (sndfile != NULL) {
+    audio_utils_fifo_deinit(&(pSles->fifo2));
+    delete[] pSles->fifo2Buffer;
+
+    SLES_PRINTF("slesDestroyServer 3");
+
+    //            sf_close(sndfile);
+    //        }
+    if (NULL != pSles->playerObject) {
+        (*(pSles->playerObject))->Destroy(pSles->playerObject);
+    }
+
+    SLES_PRINTF("slesDestroyServer 4");
+
+    if (NULL != pSles->recorderObject) {
+        (*(pSles->recorderObject))->Destroy(pSles->recorderObject);
+    }
+
+    SLES_PRINTF("slesDestroyServer 5");
+
+    (*(pSles->outputmixObject))->Destroy(pSles->outputmixObject);
+    SLES_PRINTF("slesDestroyServer 6");
+    (*(pSles->engineObject))->Destroy(pSles->engineObject);
+    SLES_PRINTF("slesDestroyServer 7");
+
+    //        free(pSles);
+    //        pSles=NULL;
+
+    status = SLES_SUCCESS;
+
+    SLES_PRINTF("End slesDestroyServer: status = %d", status);
+    return status;
+}
+
diff --git a/apps/CtsVerifier/jni/audio_loopback/sles.h b/apps/CtsVerifier/jni/audio_loopback/sles.h
new file mode 100644
index 0000000..2550b81
--- /dev/null
+++ b/apps/CtsVerifier/jni/audio_loopback/sles.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <pthread.h>
+#include <android/log.h>
+
+#ifndef _Included_org_drrickorang_loopback_sles
+#define _Included_org_drrickorang_loopback_sles
+
+//struct audio_utils_fifo;
+#define SLES_PRINTF(...)  __android_log_print(ANDROID_LOG_INFO, "sles_jni", __VA_ARGS__);
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+#include <audio_utils/fifo.h>
+
+typedef struct {
+    SLuint32 rxBufCount;     // -r#
+    SLuint32 txBufCount;     // -t#
+    SLuint32 bufSizeInFrames;  // -f#
+    SLuint32 channels;       // -c#
+    SLuint32 sampleRate; // -s#
+    SLuint32 exitAfterSeconds; // -e#
+    SLuint32 freeBufCount;   // calculated
+    SLuint32 bufSizeInBytes; // calculated
+    int injectImpulse; // -i#i
+
+    // Storage area for the buffer queues
+    char **rxBuffers;
+    char **txBuffers;
+    char **freeBuffers;
+
+    // Buffer indices
+    SLuint32 rxFront;    // oldest recording
+    SLuint32 rxRear;     // next to be recorded
+    SLuint32 txFront;    // oldest playing
+    SLuint32 txRear;     // next to be played
+    SLuint32 freeFront;  // oldest free
+    SLuint32 freeRear;   // next to be freed
+
+    struct audio_utils_fifo fifo; //(*)
+    struct audio_utils_fifo fifo2;
+    short *fifo2Buffer;
+    short *fifoBuffer;
+    SLAndroidSimpleBufferQueueItf recorderBufferQueue;
+    SLBufferQueueItf playerBufferQueue;
+
+    pthread_mutex_t mutex;// = PTHREAD_MUTEX_INITIALIZER;
+
+    //other things that belong here
+    SLObjectItf playerObject;
+    SLObjectItf recorderObject;
+    SLObjectItf outputmixObject;
+    SLObjectItf engineObject;
+} sles_data;
+
+enum {
+    SLES_SUCCESS = 0,
+    SLES_FAIL = 1,
+} SLES_STATUS_ENUM;
+
+int slesInit(sles_data ** ppSles, int samplingRate, int frameCount, int micSource);
+
+//note the double pointer to properly free the memory of the structure
+int slesDestroy(sles_data ** ppSles);
+
+///full
+int slesFull(sles_data *pSles);
+
+int slesCreateServer(sles_data *pSles, int samplingRate, int frameCount, int micSource);
+int slesProcessNext(sles_data *pSles, double *pSamples, long maxSamples);
+int slesDestroyServer(sles_data *pSles);
+
+#ifdef __cplusplus
+}
+#endif
+#endif //_Included_org_drrickorang_loopback_sles
diff --git a/apps/CtsVerifier/res/layout/audio_loopback_activity.xml b/apps/CtsVerifier/res/layout/audio_loopback_activity.xml
new file mode 100644
index 0000000..626ac4f
--- /dev/null
+++ b/apps/CtsVerifier/res/layout/audio_loopback_activity.xml
@@ -0,0 +1,83 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2015 The Android Open Source Project
+
+     Licensed under the Apache License, Version 2.0 (the "License");
+     you may not use this file except in compliance with the License.
+     You may obtain a copy of the License at
+
+          http://www.apache.org/licenses/LICENSE-2.0
+
+     Unless required by applicable law or agreed to in writing, software
+     distributed under the License is distributed on an "AS IS" BASIS,
+     WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+     See the License for the specific language governing permissions and
+     limitations under the License.
+-->
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+        android:layout_width="match_parent"
+        android:layout_height="wrap_content"
+        android:padding="10dip"
+        android:orientation="vertical">
+
+  <TextView
+      android:layout_width="match_parent"
+      android:layout_height="wrap_content"
+      android:scrollbars="vertical"
+      android:gravity="bottom"
+      android:id="@+id/info_text"
+      android:text="@string/audio_loopback_instructions" />
+  <LinearLayout
+      android:layout_width="match_parent"
+      android:layout_height="wrap_content"
+      android:orientation="vertical">
+      <Button
+          android:layout_width="match_parent"
+          android:layout_height="wrap_content"
+          android:id="@+id/audio_loopback_plug_ready_btn"
+          android:text="@string/audio_loopback_plug_ready_btn"/>
+
+    <LinearLayout
+        android:orientation="vertical"
+        android:layout_width="match_parent"
+        android:layout_height="match_parent"
+        android:id="@+id/audio_loopback_layout">
+
+        <TextView
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="@string/audio_loopback_instructions2"
+            android:id="@+id/audio_loopback_instructions2"/>
+
+        <SeekBar
+            android:layout_width="match_parent"
+            android:layout_height="wrap_content"
+            android:id="@+id/audio_loopback_level_seekbar"/>
+
+        <TextView
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="@string/audio_loopback_level_text"
+            android:id="@+id/audio_loopback_level_text"/>
+
+        <Button
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="@string/audio_loopback_test_btn"
+            android:id="@+id/audio_loopback_test_btn"/>
+
+        <ProgressBar
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:id="@+id/audio_loopback_progress_bar"/>
+
+        <TextView
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="@string/audio_loopback_results_text"
+            android:id="@+id/audio_loopback_results_text"/>
+    </LinearLayout>
+    </LinearLayout>
+
+  <include layout="@layout/pass_fail_buttons" />
+
+</LinearLayout>
\ No newline at end of file
diff --git a/apps/CtsVerifier/res/values/strings.xml b/apps/CtsVerifier/res/values/strings.xml
index 52cdd10..b72e83f 100644
--- a/apps/CtsVerifier/res/values/strings.xml
+++ b/apps/CtsVerifier/res/values/strings.xml
@@ -1804,4 +1804,27 @@
     <string name="audio_routingnotification_recHeader">AudioRecord Routing Notifications</string>
     <string name="audio_routingnotification_trackRoutingMsg">AudioTrack rerouting</string>
     <string name="audio_routingnotification_recordRoutingMsg">AudioRecord rerouting</string>
+
+    <!-- Audio Loopback Latency Test -->
+    <string name="audio_loopback_test">Audio Loopback Latency Test</string>
+     <string name="audio_loopback_info">
+          This test requires the Loopback Plug. Please connect a Loopback Plug on the headset
+          connector, and proceed with the instructions on the screen.
+          The system will measure the input-output audio latency by injecting a pulse on the output,
+          and computing the distance between replicas of the pulse.
+          You can vary the Audio Level slider to ensure the pulse will feed back at adecuate levels.
+          Repeat until a confidence level >= 0.6 is achieved.
+    </string>
+    <string name="audio_loopback_instructions">
+          Please connect a "Loopback Plug" and press "Loopback Plug Ready".
+    </string>
+    <string name="audio_loopback_plug_ready_btn">Loopback Plug Ready</string>
+    <string name="audio_loopback_instructions2">
+          Set the audio level to a suitable value, then press Test button.
+          It might require multiple tries until a confidence >= 0.6 is achieved.
+    </string>
+    <string name="audio_loopback_level_text">Audio Level</string>
+    <string name="audio_loopback_test_btn">Test</string>
+    <string name="audio_loopback_results_text">Results...</string>
+
 </resources>
diff --git a/apps/CtsVerifier/src/com/android/cts/verifier/audio/AudioLoopbackActivity.java b/apps/CtsVerifier/src/com/android/cts/verifier/audio/AudioLoopbackActivity.java
new file mode 100644
index 0000000..e603a69
--- /dev/null
+++ b/apps/CtsVerifier/src/com/android/cts/verifier/audio/AudioLoopbackActivity.java
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.cts.verifier.audio;
+
+import com.android.cts.verifier.PassFailButtons;
+import com.android.cts.verifier.R;
+import com.android.compatibility.common.util.ReportLog;
+import com.android.compatibility.common.util.ResultType;
+import com.android.compatibility.common.util.ResultUnit;
+import android.content.Context;
+
+import android.media.AudioDeviceCallback;
+import android.media.AudioDeviceInfo;
+import android.media.AudioManager;
+import android.media.AudioTrack;
+
+import android.os.Bundle;
+import android.os.Handler;
+import android.os.Message;
+
+import android.util.Log;
+
+import android.view.View;
+import android.view.View.OnClickListener;
+
+import android.widget.Button;
+import android.widget.TextView;
+import android.widget.SeekBar;
+import android.widget.LinearLayout;
+import android.widget.ProgressBar;
+
+/**
+ * Tests Audio Device roundtrip latency by using a loopback plug.
+ */
+public class AudioLoopbackActivity extends PassFailButtons.Activity {
+    private static final String TAG = "AudioLoopbackActivity";
+
+    public static final int BYTES_PER_FRAME = 2;
+
+    NativeAudioThread nativeAudioThread = null;
+
+    private int mSamplingRate = 44100;
+    private int mMinBufferSizeInFrames = 0;
+    private static final double CONFIDENCE_THRESHOLD = 0.6;
+    private Correlation mCorrelation = new Correlation();
+
+    OnBtnClickListener mBtnClickListener = new OnBtnClickListener();
+    Context mContext;
+
+    Button mLoopbackPlugReady;
+    TextView mAudioLevelText;
+    SeekBar mAudioLevelSeekbar;
+    LinearLayout mLinearLayout;
+    Button mTestButton;
+    TextView mResultText;
+    ProgressBar mProgressBar;
+
+    int mMaxLevel;
+    private class OnBtnClickListener implements OnClickListener {
+        @Override
+        public void onClick(View v) {
+            switch (v.getId()) {
+                case R.id.audio_loopback_plug_ready_btn:
+                    Log.i(TAG, "audio loopback plug ready");
+                    //enable all the other views.
+                    enableLayout(true);
+                    break;
+                case R.id.audio_loopback_test_btn:
+                    Log.i(TAG, "audio loopback test");
+                    startAudioTest();
+                    break;
+
+            }
+        }
+    }
+
+    @Override
+    protected void onCreate(Bundle savedInstanceState) {
+        super.onCreate(savedInstanceState);
+        setContentView(R.layout.audio_loopback_activity);
+
+        mContext = this;
+
+        mLoopbackPlugReady = (Button)findViewById(R.id.audio_loopback_plug_ready_btn);
+        mLoopbackPlugReady.setOnClickListener(mBtnClickListener);
+        mLinearLayout = (LinearLayout)findViewById(R.id.audio_loopback_layout);
+        mAudioLevelText = (TextView)findViewById(R.id.audio_loopback_level_text);
+        mAudioLevelSeekbar = (SeekBar)findViewById(R.id.audio_loopback_level_seekbar);
+        mTestButton =(Button)findViewById(R.id.audio_loopback_test_btn);
+        mTestButton.setOnClickListener(mBtnClickListener);
+        mResultText = (TextView)findViewById(R.id.audio_loopback_results_text);
+        mProgressBar = (ProgressBar)findViewById(R.id.audio_loopback_progress_bar);
+        showWait(false);
+
+        enableLayout(false);         //disabled all content
+        AudioManager am = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
+        mMaxLevel = am.getStreamMaxVolume(AudioManager.STREAM_MUSIC);
+        mAudioLevelSeekbar.setMax(mMaxLevel);
+        am.setStreamVolume(AudioManager.STREAM_MUSIC, (int)(0.7 * mMaxLevel), 0);
+        refreshLevel();
+
+        mAudioLevelSeekbar.setOnSeekBarChangeListener(new SeekBar.OnSeekBarChangeListener() {
+            @Override
+            public void onStopTrackingTouch(SeekBar seekBar) {
+            }
+
+            @Override
+            public void onStartTrackingTouch(SeekBar seekBar) {
+            }
+
+            @Override
+            public void onProgressChanged(SeekBar seekBar, int progress, boolean fromUser) {
+
+                AudioManager am = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
+                am.setStreamVolume(AudioManager.STREAM_MUSIC,
+                        progress, 0);
+                refreshLevel();
+                Log.i(TAG,"Changed stream volume to: " + progress);
+            }
+        });
+
+        setPassFailButtonClickListeners();
+        getPassButton().setEnabled(false);
+        setInfoResources(R.string.sample_test, R.string.audio_loopback_info, -1);
+    }
+
+    /**
+     * refresh Audio Level seekbar and text
+     */
+    private void refreshLevel() {
+        AudioManager am = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
+
+        int currentLevel = am.getStreamVolume(AudioManager.STREAM_MUSIC);
+        mAudioLevelSeekbar.setProgress(currentLevel);
+
+        String levelText = String.format("%s: %d/%d",
+                getResources().getString(R.string.audio_loopback_level_text),
+                currentLevel, mMaxLevel);
+        mAudioLevelText.setText(levelText);
+    }
+
+    /**
+     * enable test ui elements
+     */
+    private void enableLayout(boolean enable) {
+        for (int i = 0; i<mLinearLayout.getChildCount(); i++) {
+            View view = mLinearLayout.getChildAt(i);
+            view.setEnabled(enable);
+        }
+    }
+
+    /**
+     * show active progress bar
+     */
+    private void showWait(boolean show) {
+        if (show) {
+            mProgressBar.setVisibility(View.VISIBLE) ;
+        } else {
+            mProgressBar.setVisibility(View.INVISIBLE) ;
+        }
+    }
+
+    /**
+     *  Start the loopback audio test
+     */
+    private void startAudioTest() {
+        getPassButton().setEnabled(false);
+
+        //get system defaults for sampling rate, buffers.
+        AudioManager am = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
+        String value = am.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+        mMinBufferSizeInFrames = Integer.parseInt(value);
+
+        int minBufferSizeInBytes = BYTES_PER_FRAME * mMinBufferSizeInFrames;
+
+        mSamplingRate = AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_MUSIC);
+
+        Log.i(TAG, String.format("startAudioTest sr:%d , buffer:%d frames",
+                mSamplingRate, mMinBufferSizeInFrames));
+
+        nativeAudioThread = new NativeAudioThread();
+        if (nativeAudioThread != null) {
+            nativeAudioThread.setMessageHandler(mMessageHandler);
+            nativeAudioThread.mSessionId = 0;
+            nativeAudioThread.setParams(mSamplingRate,
+                    minBufferSizeInBytes,
+                    minBufferSizeInBytes,
+                    0x03 /*voice recognition*/);
+            nativeAudioThread.start();
+
+            try {
+                Thread.sleep(200);
+            } catch (InterruptedException e) {
+                e.printStackTrace();
+            }
+
+            nativeAudioThread.runTest();
+
+        }
+    }
+
+    /**
+     * handler for messages from audio thread
+     */
+    private Handler mMessageHandler = new Handler() {
+        public void handleMessage(Message msg) {
+            super.handleMessage(msg);
+            switch(msg.what) {
+                case NativeAudioThread.NATIVE_AUDIO_THREAD_MESSAGE_REC_STARTED:
+                    Log.v(TAG,"got message native rec started!!");
+                    showWait(true);
+                    mResultText.setText("Test Running...");
+                    break;
+                case NativeAudioThread.NATIVE_AUDIO_THREAD_MESSAGE_REC_ERROR:
+                    Log.v(TAG,"got message native rec can't start!!");
+                    showWait(false);
+                    mResultText.setText("Test Error.");
+                    break;
+                case NativeAudioThread.NATIVE_AUDIO_THREAD_MESSAGE_REC_COMPLETE:
+                case NativeAudioThread.NATIVE_AUDIO_THREAD_MESSAGE_REC_COMPLETE_ERRORS:
+                    if (nativeAudioThread != null) {
+                        Log.v(TAG,"Finished recording.");
+                        double [] waveData = nativeAudioThread.getWaveData();
+                        mCorrelation.computeCorrelation(waveData, mSamplingRate);
+                        mResultText.setText(String.format(
+                                "Test Finished\nLatency:%.2f ms\nConfidence: %.2f",
+                                mCorrelation.mEstimatedLatencyMs,
+                                mCorrelation.mEstimatedLatencyConfidence));
+
+                        recordTestResults();
+                        if (mCorrelation.mEstimatedLatencyConfidence >= CONFIDENCE_THRESHOLD) {
+                            getPassButton().setEnabled(true);
+                        }
+
+                        //close
+                        if (nativeAudioThread != null) {
+                            nativeAudioThread.isRunning = false;
+                            try {
+                                nativeAudioThread.finish();
+                                nativeAudioThread.join();
+                            } catch (InterruptedException e) {
+                                e.printStackTrace();
+                            }
+                            nativeAudioThread = null;
+                        }
+                        showWait(false);
+                    }
+                    break;
+                default:
+                    break;
+            }
+        }
+    };
+
+    /**
+     * Store test results in log
+     */
+    private void recordTestResults() {
+
+        getReportLog().addValue(
+                "Estimated Latency",
+                mCorrelation.mEstimatedLatencyMs,
+                ResultType.LOWER_BETTER,
+                ResultUnit.MS);
+
+        getReportLog().addValue(
+                "Confidence",
+                mCorrelation.mEstimatedLatencyConfidence,
+                ResultType.HIGHER_BETTER,
+                ResultUnit.NONE);
+
+        int audioLevel = mAudioLevelSeekbar.getProgress();
+        getReportLog().addValue(
+                "Audio Level",
+                audioLevel,
+                ResultType.NEUTRAL,
+                ResultUnit.NONE);
+
+        getReportLog().addValue(
+                "Frames Buffer Size",
+                mMinBufferSizeInFrames,
+                ResultType.NEUTRAL,
+                ResultUnit.NONE);
+
+        getReportLog().addValue(
+                "Sampling Rate",
+                mSamplingRate,
+                ResultType.NEUTRAL,
+                ResultUnit.NONE);
+
+        Log.v(TAG,"Results Recorded");
+    }
+}
diff --git a/apps/CtsVerifier/src/com/android/cts/verifier/audio/Correlation.java b/apps/CtsVerifier/src/com/android/cts/verifier/audio/Correlation.java
new file mode 100644
index 0000000..75b04eb
--- /dev/null
+++ b/apps/CtsVerifier/src/com/android/cts/verifier/audio/Correlation.java
@@ -0,0 +1,156 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.cts.verifier.audio;
+
+import android.util.Log;
+
+
+public class Correlation {
+
+    private int mBlockSize = 4096;
+    private int mSamplingRate = 44100;
+    private double [] mDataDownsampled = new double [mBlockSize];
+    private double [] mDataAutocorrelated = new double[mBlockSize];
+
+    public double mEstimatedLatencySamples = 0;
+    public double mEstimatedLatencyMs = 0;
+    public double mEstimatedLatencyConfidence = 0.0;
+
+    public void init(int blockSize, int samplingRate) {
+        mBlockSize = blockSize;
+        mSamplingRate = samplingRate;
+    }
+
+    public boolean computeCorrelation(double [] data, int samplingRate) {
+        boolean status = false;
+        log("Started Auto Correlation for data with " + data.length + " points");
+        mSamplingRate = samplingRate;
+
+        downsampleData(data, mDataDownsampled);
+
+        //correlation vector
+        autocorrelation(mDataDownsampled, mDataAutocorrelated);
+
+        int N = data.length; //all samples available
+        double groupSize =  (double) N / mBlockSize;  //samples per downsample point.
+
+        double maxValue = 0;
+        int maxIndex = -1;
+
+        double minLatencyMs = 8; //min latency expected. This algorithm should be improved.
+        int minIndex = (int)(0.5 + minLatencyMs * mSamplingRate / (groupSize*1000));
+
+        double average = 0;
+        double rms = 0;
+        //find max
+        for (int i=minIndex; i<mDataAutocorrelated.length; i++) {
+            average += mDataAutocorrelated[i];
+            rms += mDataAutocorrelated[i]*mDataAutocorrelated[i];
+            if (mDataAutocorrelated[i] > maxValue) {
+                maxValue = mDataAutocorrelated[i];
+                maxIndex = i;
+            }
+        }
+
+        rms = Math.sqrt(rms/mDataAutocorrelated.length);
+        average = average/mDataAutocorrelated.length;
+        log(String.format(" Maxvalue %f, max Index : %d/%d (%d)  minIndex=%d",maxValue, maxIndex,
+                mDataAutocorrelated.length, data.length, minIndex));
+
+        log(String.format("  average : %.3f  rms: %.3f", average, rms));
+
+        mEstimatedLatencyConfidence = 0.0;
+        if (average>0) {
+            double factor = 3.0;
+
+            double raw = (rms-average) /(factor*average);
+            log(String.format("Raw: %.3f",raw));
+            mEstimatedLatencyConfidence = Math.max(Math.min(raw, 1.0),0.0);
+        }
+
+        log(String.format(" ****Confidence: %.2f",mEstimatedLatencyConfidence));
+
+        mEstimatedLatencySamples = maxIndex*groupSize;
+
+        mEstimatedLatencyMs = mEstimatedLatencySamples *1000/mSamplingRate;
+
+        log(String.format(" latencySamples: %.2f  %.2f ms", mEstimatedLatencySamples,
+                mEstimatedLatencyMs));
+
+        status = true;
+        return status;
+    }
+
+    private boolean downsampleData(double [] data, double [] dataDownsampled) {
+
+        boolean status = false;
+        // mDataDownsampled = new double[mBlockSize];
+        for (int i=0; i<mBlockSize; i++) {
+            dataDownsampled[i] = 0;
+        }
+
+        int N = data.length; //all samples available
+        double groupSize =  (double) N / mBlockSize;
+
+        int currentIndex = 0;
+        double nextGroup = groupSize;
+        for (int i = 0; i<N && currentIndex<mBlockSize; i++) {
+
+            if (i> nextGroup) { //advanced to next group.
+                currentIndex++;
+                nextGroup += groupSize;
+            }
+
+            if (currentIndex>=mBlockSize) {
+                break;
+            }
+            dataDownsampled[currentIndex] += Math.abs(data[i]);
+        }
+
+        status = true;
+        return status;
+    }
+
+    private boolean autocorrelation(double [] data, double [] dataOut) {
+        boolean status = false;
+
+        double sumsquared = 0;
+        int N = data.length;
+        for (int i=0; i<N; i++) {
+            double value = data[i];
+            sumsquared += value*value;
+        }
+
+        if (sumsquared>0) {
+            for (int i = 0; i < N; i++) {
+                dataOut[i] = 0;
+                for (int j = 0; j < N - i; j++) {
+
+                    dataOut[i] += data[j] * data[i + j];
+                }
+                dataOut[i] = dataOut[i] / sumsquared;
+            }
+            status = true;
+        }
+
+        return status;
+    }
+
+    private static void log(String msg) {
+        Log.v("Recorder", msg);
+    }
+}
diff --git a/apps/CtsVerifier/src/com/android/cts/verifier/audio/NativeAudioThread.java b/apps/CtsVerifier/src/com/android/cts/verifier/audio/NativeAudioThread.java
new file mode 100644
index 0000000..224d4c8
--- /dev/null
+++ b/apps/CtsVerifier/src/com/android/cts/verifier/audio/NativeAudioThread.java
@@ -0,0 +1,273 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//package org.drrickorang.loopback;
+
+package com.android.cts.verifier.audio;
+
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioTrack;
+//import android.media.MediaPlayer;
+import android.media.AudioRecord;
+import android.media.MediaRecorder;
+import android.util.Log;
+
+import android.os.Handler;
+import  android.os.Message;
+
+/**
+ * A thread/audio track based audio synth.
+ */
+public class NativeAudioThread extends Thread {
+
+    public boolean isRunning = false;
+    double twoPi = 6.28318530718;
+
+    public int mSessionId;
+
+    public double[] mvSamples; //captured samples
+    int mSamplesIndex;
+
+    private final int mSecondsToRun = 2;
+    public int mSamplingRate = 48000;
+    private int mChannelConfigIn = AudioFormat.CHANNEL_IN_MONO;
+    private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT;
+
+    int mMinPlayBufferSizeInBytes = 0;
+    int mMinRecordBuffSizeInBytes = 0;
+    private int mChannelConfigOut = AudioFormat.CHANNEL_OUT_MONO;
+
+    int mMicSource = 0;
+
+//    private double [] samples = new double[50000];
+
+    boolean isPlaying = false;
+    private Handler mMessageHandler;
+    boolean isDestroying = false;
+    boolean hasDestroyingErrors = false;
+
+    static final int NATIVE_AUDIO_THREAD_MESSAGE_REC_STARTED = 892;
+    static final int NATIVE_AUDIO_THREAD_MESSAGE_REC_ERROR = 893;
+    static final int NATIVE_AUDIO_THREAD_MESSAGE_REC_COMPLETE = 894;
+    static final int NATIVE_AUDIO_THREAD_MESSAGE_REC_COMPLETE_ERRORS = 895;
+
+    public void setParams(int samplingRate, int playBufferInBytes, int recBufferInBytes,
+            int micSource) {
+        mSamplingRate = samplingRate;
+        mMinPlayBufferSizeInBytes = playBufferInBytes;
+        mMinRecordBuffSizeInBytes = recBufferInBytes;
+        mMicSource = micSource;
+    }
+
+    //JNI load
+    static {
+        try {
+            System.loadLibrary("audioloopback_jni");
+        } catch (UnsatisfiedLinkError e) {
+            log("Error loading loopback JNI library");
+            e.printStackTrace();
+        }
+
+        /* TODO: gracefully fail/notify if the library can't be loaded */
+    }
+
+    //jni calls
+    public native long slesInit(int samplingRate, int frameCount, int micSource);
+    public native int slesProcessNext(long sles_data, double[] samples, long offset);
+    public native int slesDestroy(long sles_data);
+
+    public void run() {
+
+        setPriority(Thread.MAX_PRIORITY);
+        isRunning = true;
+
+        //erase output buffer
+        if (mvSamples != null)
+            mvSamples = null;
+
+        //resize
+        int nNewSize = (int)(1.1* mSamplingRate * mSecondsToRun ); //10% more just in case
+        mvSamples = new double[nNewSize];
+        mSamplesIndex = 0; //reset index
+
+        //clear samples
+        for (int i=0; i<nNewSize; i++) {
+            mvSamples[i] = 0;
+        }
+
+        //start playing
+        isPlaying = true;
+
+
+        log(" Started capture test");
+        if (mMessageHandler != null) {
+            Message msg = Message.obtain();
+            msg.what = NATIVE_AUDIO_THREAD_MESSAGE_REC_STARTED;
+            mMessageHandler.sendMessage(msg);
+        }
+
+
+
+        log(String.format("about to init, sampling rate: %d, buffer:%d", mSamplingRate,
+                mMinPlayBufferSizeInBytes/2 ));
+        long sles_data = slesInit(mSamplingRate, mMinPlayBufferSizeInBytes/2, mMicSource);
+        log(String.format("sles_data = 0x%X",sles_data));
+
+        if (sles_data == 0 ) {
+            log(" ERROR at JNI initialization");
+            if (mMessageHandler != null) {
+                Message msg = Message.obtain();
+                msg.what = NATIVE_AUDIO_THREAD_MESSAGE_REC_ERROR;
+                mMessageHandler.sendMessage(msg);
+            }
+        }  else {
+
+            //wait a little bit...
+            try {
+                sleep(10); //just to let it start properly?
+            } catch (InterruptedException e) {
+                e.printStackTrace();
+            }
+
+
+
+            mSamplesIndex = 0;
+            int totalSamplesRead = 0;
+            long offset = 0;
+            for (int ii = 0; ii < mSecondsToRun; ii++) {
+                log(String.format("block %d...", ii));
+                int samplesRead = slesProcessNext(sles_data, mvSamples,offset);
+                totalSamplesRead += samplesRead;
+
+                offset += samplesRead;
+                log(" [" + ii + "] jni samples read:" + samplesRead + "  currentOffset:" + offset);
+            }
+
+            log(String.format(" samplesRead: %d, sampleOffset:%d", totalSamplesRead, offset));
+            log(String.format("about to destroy..."));
+
+            runDestroy(sles_data);
+
+            int maxTry = 20;
+            int tryCount = 0;
+            //isDestroying = true;
+            while (isDestroying) {
+
+                try {
+                    sleep(40);
+                } catch (InterruptedException e) {
+                    e.printStackTrace();
+                }
+
+                tryCount++;
+
+                log("destroy try: " + tryCount);
+
+                if (tryCount >= maxTry) {
+                    hasDestroyingErrors = true;
+                    log("WARNING: waited for max time to properly destroy JNI.");
+                    break;
+                }
+            }
+            log(String.format("after destroying. TotalSamplesRead = %d", totalSamplesRead));
+
+            if (totalSamplesRead==0)
+            {
+                hasDestroyingErrors = true;
+            }
+
+            endTest();
+        }
+    }
+
+    public void setMessageHandler(Handler messageHandler) {
+        mMessageHandler = messageHandler;
+    }
+
+    private void runDestroy(final long sles_data ) {
+        isDestroying = true;
+
+        //start thread
+
+        final long local_sles_data = sles_data;
+        ////
+        Thread thread = new Thread(new Runnable() {
+            public void run() {
+                isDestroying = true;
+                log("**Start runnable destroy");
+
+                int status = slesDestroy(local_sles_data);
+                log(String.format("**End runnable destroy sles delete status: %d", status));
+                isDestroying = false;
+            }
+        });
+
+        thread.start();
+
+
+
+        log("end of runDestroy()");
+
+
+    }
+
+    public void togglePlay() {
+
+    }
+
+    public void runTest() {
+
+
+    }
+
+   public void endTest() {
+       log("--Ending capture test--");
+       isPlaying = false;
+
+
+       if (mMessageHandler != null) {
+           Message msg = Message.obtain();
+           if (hasDestroyingErrors)
+               msg.what = NATIVE_AUDIO_THREAD_MESSAGE_REC_COMPLETE_ERRORS;
+           else
+               msg.what = NATIVE_AUDIO_THREAD_MESSAGE_REC_COMPLETE;
+           mMessageHandler.sendMessage(msg);
+       }
+
+   }
+
+    public void finish() {
+
+        if (isRunning) {
+            isRunning = false;
+            try {
+                sleep(20);
+            } catch (InterruptedException e) {
+                e.printStackTrace();
+            }
+        }
+    }
+
+    private static void log(String msg) {
+        Log.v("Loopback", msg);
+    }
+
+    double [] getWaveData () {
+        return mvSamples;
+    }
+
+}  //end thread.