diff --git a/Android.mk b/Android.mk
index 8234410..83a17ed 100644
--- a/Android.mk
+++ b/Android.mk
@@ -2,7 +2,6 @@
 # Quake game
 # This makefile builds both an activity and a shared library.
 #########################################################################
-ifneq ($(TARGET_SIMULATOR),true) # not 64 bit clean
 
 TOP_LOCAL_PATH:= $(call my-dir)
 
@@ -91,18 +90,17 @@
   world.cpp \
   zone.cpp
 
+LOCAL_C_INCLUDES:= \
+	system/media/wilhelm/include
+
 LOCAL_SHARED_LIBRARIES := \
 	libutils \
-	libmedia \
 	libEGL \
-	libGLESv1_CM
+	libGLESv1_CM \
+	libOpenSLES
 
 LOCAL_MODULE := libquake
 
 LOCAL_ARM_MODE := arm
 
-LOCAL_PRELINK_MODULE := false
-
 include $(BUILD_SHARED_LIBRARY)
-
-endif # TARGET_SIMULATOR
diff --git a/quake/src/WinQuake/snd_android.cpp b/quake/src/WinQuake/snd_android.cpp
index f02f900..1327023 100644
--- a/quake/src/WinQuake/snd_android.cpp
+++ b/quake/src/WinQuake/snd_android.cpp
@@ -10,12 +10,399 @@
 #include <time.h>
 #include <math.h>
 #include <stdlib.h>
-#include <utils/Log.h>
-#include <media/AudioTrack.h>
+#include <unistd.h>
 
-using namespace android;
+#include <android/log.h>
+#include <SLES/OpenSLES.h>
 
-static AudioTrack gAudioTrack;
+#define LOG_TAG "Quake snd_android"
+#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
+#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
+
+const size_t SAMPLE_RATE = 11025;
+
+
+const size_t BYTES_PER_SAMPLE = 2;
+const size_t CHANNEL_COUNT = 2;
+const size_t BITS_PER_SAMPLE = 8 * BYTES_PER_SAMPLE;
+
+const size_t TOTAL_BUFFER_SIZE = 4 * 1024;
+
+#define MAX_NUMBER_INTERFACES 3
+
+/* Local storage for Audio data in 16 bit words */
+#define AUDIO_DATA_STORAGE_SIZE (TOTAL_BUFFER_SIZE / 2)
+/* Audio data buffer size in 16 bit words. 8 data segments are used in
+this simple example */
+#define AUDIO_DATA_BUFFER_SIZE (4096/8)
+
+const size_t NUMBER_OF_BUFFERS = AUDIO_DATA_STORAGE_SIZE / AUDIO_DATA_BUFFER_SIZE;
+
+/* Checks for error. If any errors exit the application! */
+void CheckErr( SLresult res )
+{
+    if ( res != SL_RESULT_SUCCESS )
+        {
+            fprintf(stdout, "%u SL failure, exiting\n", res);
+            exit(EXIT_FAILURE);
+        }
+    else {
+        //fprintf(stdout, "%d SL success, proceeding...\n", res);
+    }
+}
+
+/* Structure for passing information to callback function */
+typedef struct CallbackCntxt_ {
+    SLPlayItf  playItf;
+    SLint16*   pDataBase;    // Base adress of local audio data storage
+    SLint16*   pData;        // Current adress of local audio data storage
+    SLuint32   size;
+} CallbackCntxt;
+
+/* Local storage for Audio data */
+SLint16 pcmData[AUDIO_DATA_STORAGE_SIZE];
+
+/* Callback for Buffer Queue events */
+void BufferQueueCallback(
+        SLBufferQueueItf queueItf,
+        void *pContext)
+{
+    //fprintf(stdout, "BufferQueueCallback called\n");
+    SLresult res;
+    //fprintf(stdout, " pContext=%p\n", pContext);
+    CallbackCntxt *pCntxt = (CallbackCntxt*)pContext;
+
+    if (pCntxt->pData >= (pCntxt->pDataBase + pCntxt->size)) {
+        pCntxt->pData = pCntxt->pDataBase;
+    }
+    {
+        //fprintf(stdout, "callback: before enqueue\n");
+        res = (*queueItf)->Enqueue(queueItf, (void*) pCntxt->pData,
+                2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */
+        CheckErr(res);
+        /* Increase data pointer by buffer size */
+        pCntxt->pData += AUDIO_DATA_BUFFER_SIZE;
+    }
+    //fprintf(stdout, "end of BufferQueueCallback()\n");
+}
+
+SLEngineItf                EngineItf;
+
+SLint32                    numOutputs = 0;
+SLuint32                   deviceID = 0;
+
+
+SLDataSource               audioSource;
+SLDataLocator_BufferQueue  bufferQueue;
+SLDataFormat_PCM           pcm;
+
+SLDataSink                 audioSink;
+SLDataLocator_OutputMix    locator_outputmix;
+
+
+SLVolumeItf                volumeItf;
+
+
+SLboolean required[MAX_NUMBER_INTERFACES];
+SLInterfaceID iidArray[MAX_NUMBER_INTERFACES];
+
+/* Callback context for the buffer queue callback function */
+CallbackCntxt cntxt;
+
+static SLObjectItf                OutputMix;
+static SLPlayItf                  playItf;
+static SLObjectItf                player;
+static SLBufferQueueItf           bufferQueueItf;
+static SLBufferQueueState         state;
+
+/* Play some audio from a buffer queue  */
+void TestPlaySawtoothBufferQueue( SLObjectItf sl )
+{
+    SLresult                   res;
+    int                        i;
+
+    /* Get the SL Engine Interface which is implicit */
+    res = (*sl)->GetInterface(sl, SL_IID_ENGINE, (void*)&EngineItf);
+    CheckErr(res);
+
+    /* Initialize arrays required[] and iidArray[] */
+    for (i=0;i<MAX_NUMBER_INTERFACES;i++)
+        {
+            required[i] = SL_BOOLEAN_FALSE;
+            iidArray[i] = SL_IID_NULL;
+        }
+
+    // Set arrays required[] and iidArray[] for VOLUME interface
+    required[0] = SL_BOOLEAN_TRUE;
+    iidArray[0] = SL_IID_VOLUME;
+    // Create Output Mix object to be used by player
+    res = (*EngineItf)->CreateOutputMix(EngineItf, &OutputMix, 0,
+            iidArray, required); CheckErr(res);
+
+    // Realizing the Output Mix object in synchronous mode.
+    res = (*OutputMix)->Realize(OutputMix, SL_BOOLEAN_FALSE);
+    CheckErr(res);
+
+#if 0
+    res = (*OutputMix)->GetInterface(OutputMix, SL_IID_VOLUME,
+            (void*)&volumeItf); CheckErr(res);
+#endif
+
+    /* Setup the data source structure for the buffer queue */
+    bufferQueue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
+    bufferQueue.numBuffers = 4;  /* Four buffers in our buffer queue */
+
+    /* Setup the format of the content in the buffer queue */
+    pcm.formatType = SL_DATAFORMAT_PCM;
+    pcm.numChannels = 2;
+    pcm.samplesPerSec = SL_SAMPLINGRATE_11_025;
+    pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+    pcm.containerSize = 16;
+    pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+    pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+
+    audioSource.pFormat      = (void *)&pcm;
+    audioSource.pLocator     = (void *)&bufferQueue;
+
+    /* Setup the data sink structure */
+    locator_outputmix.locatorType   = SL_DATALOCATOR_OUTPUTMIX;
+    locator_outputmix.outputMix    = OutputMix;
+    audioSink.pLocator           = (void *)&locator_outputmix;
+    audioSink.pFormat            = NULL;
+
+    /* Initialize the audio data to silence */
+    memset(pcmData, 0, sizeof(pcmData));
+
+    /* Initialize the context for Buffer queue callbacks */
+    cntxt.pDataBase = /*(void*)&*/pcmData;
+    cntxt.pData = cntxt.pDataBase;
+    cntxt.size = sizeof(pcmData) / 2;
+
+    /* Set arrays required[] and iidArray[] for SEEK interface
+          (PlayItf is implicit) */
+    required[0] = SL_BOOLEAN_TRUE;
+    iidArray[0] = SL_IID_BUFFERQUEUE;
+
+    /* Create the music player */
+    res = (*EngineItf)->CreateAudioPlayer(EngineItf, &player,
+            &audioSource, &audioSink, 1, iidArray, required); CheckErr(res);
+    fprintf(stdout, "bufferQueue example: after CreateAudioPlayer\n");
+
+    /* Realizing the player in synchronous mode. */
+    res = (*player)->Realize(player, SL_BOOLEAN_FALSE); CheckErr(res);
+    fprintf(stdout, "bufferQueue example: after Realize\n");
+
+    /* Get seek and play interfaces */
+    res = (*player)->GetInterface(player, SL_IID_PLAY, (void*)&playItf);
+    CheckErr(res);
+    fprintf(stdout, "bufferQueue example: after GetInterface(PLAY)\n");
+
+    res = (*player)->GetInterface(player, SL_IID_BUFFERQUEUE,
+            (void*)&bufferQueueItf); CheckErr(res);
+
+    /* Setup to receive buffer queue event callbacks */
+    res = (*bufferQueueItf)->RegisterCallback(bufferQueueItf,
+            BufferQueueCallback, &cntxt); CheckErr(res);
+
+#if 0
+    /* Before we start set volume to -3dB (-300mB) */
+    res = (*volumeItf)->SetVolumeLevel(volumeItf, -300); CheckErr(res);
+#endif
+
+    /* Enqueue a few buffers to get the ball rolling */
+    res = (*bufferQueueItf)->Enqueue(bufferQueueItf, cntxt.pData,
+            2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */
+    CheckErr(res);
+    cntxt.pData += AUDIO_DATA_BUFFER_SIZE;
+
+    res = (*bufferQueueItf)->Enqueue(bufferQueueItf, cntxt.pData,
+            2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */
+    CheckErr(res);
+    cntxt.pData += AUDIO_DATA_BUFFER_SIZE;
+
+    res = (*bufferQueueItf)->Enqueue(bufferQueueItf, cntxt.pData,
+            2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */
+    CheckErr(res);
+    cntxt.pData += AUDIO_DATA_BUFFER_SIZE;
+
+    /* Play the PCM samples using a buffer queue */
+    fprintf(stdout, "bufferQueue example: starting to play\n");
+    res = (*playItf)->SetPlayState( playItf, SL_PLAYSTATE_PLAYING );
+    CheckErr(res);
+
+    /* Wait until the PCM data is done playing, the buffer queue callback
+           will continue to queue buffers until the entire PCM data has been
+           played. This is indicated by waiting for the count member of the
+           SLBufferQueueState to go to zero.
+     */
+    res = (*bufferQueueItf)->GetState(bufferQueueItf, &state);
+    CheckErr(res);
+
+#if 0
+    // while (state.playIndex < 100) {
+    while (state.count) {
+        usleep(10000);
+        (*bufferQueueItf)->GetState(bufferQueueItf, &state);
+    }
+
+ #endif
+}
+
+SLObjectItf gSoundEngine;
+
+int startAndroidSound()
+{
+    SLresult    res;
+
+    SLEngineOption EngineOption[] = {
+            {(SLuint32) SL_ENGINEOPTION_THREADSAFE,
+            (SLuint32) SL_BOOLEAN_TRUE}};
+
+    res = slCreateEngine( &gSoundEngine, 1, EngineOption, 0, NULL, NULL);
+    CheckErr(res);
+    /* Realizing the SL Engine in synchronous mode. */
+    res = (*gSoundEngine)->Realize(gSoundEngine, SL_BOOLEAN_FALSE); CheckErr(res);
+
+    /* Run the test */
+    TestPlaySawtoothBufferQueue(gSoundEngine);
+    return EXIT_SUCCESS;
+}
+
+void finishAndroidSound()
+{
+    SLresult                   res;
+
+    if (gSoundEngine == NULL) {
+        return;
+    }
+
+    /* Make sure player is stopped */
+    if (playItf != NULL) {
+         res = (*playItf)->SetPlayState(playItf, SL_PLAYSTATE_STOPPED);
+         CheckErr(res);
+         playItf = NULL;
+    }
+
+    if (player != NULL) {
+         /* Destroy the player */
+         (*player)->Destroy(player);
+         player = NULL;
+    }
+
+    if (OutputMix != NULL) {
+        /* Destroy Output Mix object */
+        (*OutputMix)->Destroy(OutputMix);
+        OutputMix = NULL;
+    }
+
+    /* Shutdown OpenSL ES */
+    (*gSoundEngine)->Destroy(gSoundEngine);
+    gSoundEngine = NULL;
+}
+
+#if 1
+
+/*
+==================
+SNDDMA_Init
+
+Try to find a sound device to mix for.
+Returns false if nothing is found.
+==================
+*/
+qboolean SNDDMA_Init(void)
+{
+    // Initialize Quake's idea of a DMA buffer.
+
+    shm = &sn;
+    memset((void*)&sn, 0, sizeof(sn));
+
+    shm->splitbuffer = false;	// Not used.
+    shm->samplebits = 16;
+    shm->speed = 11025;
+    shm->channels = 2;
+    shm->samples = TOTAL_BUFFER_SIZE / BYTES_PER_SAMPLE;
+    shm->samplepos = 0; // Not used.
+    shm->buffer = (unsigned char*) pcmData;
+    shm->submission_chunk = 1; // Not used.
+
+    shm->soundalive = true;
+
+    if ( (shm->samples & 0x1ff) != 0 ) {
+      LOGE("SNDDDMA_Init: samples must be power of two.");
+      return false;
+    }
+
+    if ( shm->buffer == 0 ) {
+      LOGE("SNDDDMA_Init: Could not allocate sound buffer.");
+      return false;
+    }
+
+    int result = startAndroidSound();
+    return result == EXIT_SUCCESS;
+}
+
+/*
+==============
+SNDDMA_GetDMAPos
+
+return the current sample position (in mono samples read)
+inside the recirculating dma buffer, so the mixing code will know
+how many sample are required to fill it up.
+===============
+*/
+int SNDDMA_GetDMAPos(void)
+{
+    SLresult                   res;
+    if (bufferQueueItf != NULL) {
+        res = (*bufferQueueItf)->GetState(bufferQueueItf, &state);
+        CheckErr(res);
+        // Index of the currently playing or filling buffer.
+        SLuint32 playIndex = state.playIndex;
+        int ringIndex = playIndex % NUMBER_OF_BUFFERS;
+        return ringIndex * AUDIO_DATA_BUFFER_SIZE;
+    }
+    return 0;
+}
+
+/*
+===============
+SNDDMA_ReportWrite
+
+Report valid data being written into the DMA buffer by the sound mixing code.
+This is an Android specific API.
+================
+*/
+void SNDDMA_ReportWrite(size_t lengthBytes) {
+    // TODO: keep track of how much of the sound ring buffer has sound in it,
+    // detect starvation, and mix silence when we are starving.
+}
+
+/*
+==============
+SNDDMA_Submit
+
+Send sound to device if buffer isn't really the dma buffer
+===============
+*/
+void SNDDMA_Submit(void)
+{
+}
+
+/*
+==============
+SNDDMA_Shutdown
+
+Reset the sound device for exiting
+===============
+*/
+void SNDDMA_Shutdown(void)
+{
+    finishAndroidSound();
+}
+
+
+#else
 
 // Written by the callback function running in an audio thread.
 // index in bytes of where we last read.
@@ -239,10 +626,10 @@
   // Initialize the AudioTrack.
 
   status_t result = gAudioTrack.set(
-    AudioSystem::DEFAULT, // stream type
+    AUDIO_STREAM_DEFAULT, // stream type
     SAMPLE_RATE,   // sample rate
-    BITS_PER_SAMPLE == 16 ? AudioSystem::PCM_16_BIT : AudioSystem::PCM_8_BIT,      // format (8 or 16)
-    (CHANNEL_COUNT > 1) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,       // channel mask
+    BITS_PER_SAMPLE == 16 ? AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_8_BIT,      // format (8 or 16)
+    (CHANNEL_COUNT > 1) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,       // channel mask
     0,       // default buffer size
     0, // flags
     AndroidQuakeSoundCallback, // callback
@@ -253,7 +640,7 @@
 
   if ( result == NO_ERROR ) {
     LOGI("AudioTrack latency = %u ms\n", gAudioTrack.latency());
-    LOGI("AudioTrack format = %u bits\n", gAudioTrack.format() == AudioSystem::PCM_16_BIT ? 16 : 8);
+    LOGI("AudioTrack format = %u bits\n", gAudioTrack.format() == AUDIO_FORMAT_PCM_16_BIT ? 16 : 8);
     LOGI("AudioTrack sample rate = %u Hz\n", gAudioTrack.getSampleRate());
     LOGI("AudioTrack frame count = %d\n", int(gAudioTrack.frameCount()));
     LOGI("AudioTrack channel count = %d\n", gAudioTrack.channelCount());
@@ -264,7 +651,7 @@
     memset((void*)&sn, 0, sizeof(sn));
 
     shm->splitbuffer = false;	// Not used.
-    shm->samplebits = gAudioTrack.format() == AudioSystem::PCM_16_BIT ? 16 : 8;
+    shm->samplebits = gAudioTrack.format() == AUDIO_FORMAT_PCM_16_BIT ? 16 : 8;
     shm->speed = gAudioTrack.getSampleRate();
     shm->channels = gAudioTrack.channelCount();
     shm->samples = TOTAL_BUFFER_SIZE / BYTES_PER_SAMPLE;
@@ -347,3 +734,5 @@
 {
   gAudioTrack.stop();
 }
+
+#endif