am 99985d98: am e74da38e: Merge "Make SoundPool use MediaCodec"
automerge: 6e14c57

* commit '6e14c5705c5acf9d6036060252f257a084cd7578':
  Make SoundPool use MediaCodec
diff --git a/media/java/android/media/SoundPool.java b/media/java/android/media/SoundPool.java
index 4f74bdd..32d5b82 100644
--- a/media/java/android/media/SoundPool.java
+++ b/media/java/android/media/SoundPool.java
@@ -543,11 +543,6 @@
 
         public int load(String path, int priority)
         {
-            // pass network streams to player
-            if (path.startsWith("http:"))
-                return _load(path, priority);
-
-            // try local path
             int id = 0;
             try {
                 File f = new File(path);
@@ -562,6 +557,7 @@
             return id;
         }
 
+        @Override
         public int load(Context context, int resId, int priority) {
             AssetFileDescriptor afd = context.getResources().openRawResourceFd(resId);
             int id = 0;
@@ -576,6 +572,7 @@
             return id;
         }
 
+        @Override
         public int load(AssetFileDescriptor afd, int priority) {
             if (afd != null) {
                 long len = afd.getLength();
@@ -588,16 +585,17 @@
             }
         }
 
+        @Override
         public int load(FileDescriptor fd, long offset, long length, int priority) {
             return _load(fd, offset, length, priority);
         }
 
-        private native final int _load(String uri, int priority);
-
         private native final int _load(FileDescriptor fd, long offset, long length, int priority);
 
+        @Override
         public native final boolean unload(int soundID);
 
+        @Override
         public final int play(int soundID, float leftVolume, float rightVolume,
                 int priority, int loop, float rate) {
             if (isRestricted()) {
@@ -620,16 +618,22 @@
             }
         }
 
+        @Override
         public native final void pause(int streamID);
 
+        @Override
         public native final void resume(int streamID);
 
+        @Override
         public native final void autoPause();
 
+        @Override
         public native final void autoResume();
 
+        @Override
         public native final void stop(int streamID);
 
+        @Override
         public final void setVolume(int streamID, float leftVolume, float rightVolume) {
             if (isRestricted()) {
                 return;
@@ -639,16 +643,21 @@
 
         private native final void _setVolume(int streamID, float leftVolume, float rightVolume);
 
+        @Override
         public void setVolume(int streamID, float volume) {
             setVolume(streamID, volume, volume);
         }
 
+        @Override
         public native final void setPriority(int streamID, int priority);
 
+        @Override
         public native final void setLoop(int streamID, int loop);
 
+        @Override
         public native final void setRate(int streamID, float rate);
 
+        @Override
         public void setOnLoadCompleteListener(SoundPool.OnLoadCompleteListener listener)
         {
             synchronized(mLock) {
@@ -729,52 +738,69 @@
             return 0;
         }
 
+        @Override
         public int load(Context context, int resId, int priority) {
             return 0;
         }
 
+        @Override
         public int load(AssetFileDescriptor afd, int priority) {
             return 0;
         }
 
+        @Override
         public int load(FileDescriptor fd, long offset, long length, int priority) {
             return 0;
         }
 
+        @Override
         public final boolean unload(int soundID) {
             return true;
         }
 
+        @Override
         public final int play(int soundID, float leftVolume, float rightVolume,
                 int priority, int loop, float rate) {
             return 0;
         }
 
+        @Override
         public final void pause(int streamID) { }
 
+        @Override
         public final void resume(int streamID) { }
 
+        @Override
         public final void autoPause() { }
 
+        @Override
         public final void autoResume() { }
 
+        @Override
         public final void stop(int streamID) { }
 
+        @Override
         public final void setVolume(int streamID,
                 float leftVolume, float rightVolume) { }
 
+        @Override
         public void setVolume(int streamID, float volume) {
         }
 
+        @Override
         public final void setPriority(int streamID, int priority) { }
 
+        @Override
         public final void setLoop(int streamID, int loop) { }
 
+        @Override
         public final void setRate(int streamID, float rate) { }
 
+        @Override
         public void setOnLoadCompleteListener(SoundPool.OnLoadCompleteListener listener) {
         }
 
+        @Override
         public final void release() { }
     }
 }
diff --git a/media/jni/soundpool/Android.mk b/media/jni/soundpool/Android.mk
index 3382512..71ab013 100644
--- a/media/jni/soundpool/Android.mk
+++ b/media/jni/soundpool/Android.mk
@@ -2,7 +2,9 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:= \
-    android_media_SoundPool_SoundPoolImpl.cpp
+    android_media_SoundPool_SoundPoolImpl.cpp \
+    SoundPool.cpp \
+    SoundPoolThread.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     liblog \
@@ -10,7 +12,9 @@
     libutils \
     libandroid_runtime \
     libnativehelper \
-    libmedia
+    libmedia \
+    libmediandk \
+    libbinder
 
 LOCAL_MODULE:= libsoundpool
 
diff --git a/media/jni/soundpool/SoundPool.cpp b/media/jni/soundpool/SoundPool.cpp
new file mode 100644
index 0000000..f42f850
--- /dev/null
+++ b/media/jni/soundpool/SoundPool.cpp
@@ -0,0 +1,1018 @@
+/*
+ * Copyright (C) 2007 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoundPool"
+
+#include <inttypes.h>
+
+#include <utils/Log.h>
+
+#define USE_SHARED_MEM_BUFFER
+
+#include <media/AudioTrack.h>
+#include <media/IMediaHTTPService.h>
+#include <media/mediaplayer.h>
+#include <media/stagefright/MediaExtractor.h>
+#include "SoundPool.h"
+#include "SoundPoolThread.h"
+#include <media/AudioPolicyHelper.h>
+#include <ndk/NdkMediaCodec.h>
+#include <ndk/NdkMediaExtractor.h>
+#include <ndk/NdkMediaFormat.h>
+
+namespace android
+{
+
+int kDefaultBufferCount = 4;
+uint32_t kMaxSampleRate = 48000;
+uint32_t kDefaultSampleRate = 44100;
+uint32_t kDefaultFrameCount = 1200;
+size_t kDefaultHeapSize = 1024 * 1024; // 1MB
+
+
+SoundPool::SoundPool(int maxChannels, const audio_attributes_t* pAttributes)
+{
+    ALOGV("SoundPool constructor: maxChannels=%d, attr.usage=%d, attr.flags=0x%x, attr.tags=%s",
+            maxChannels, pAttributes->usage, pAttributes->flags, pAttributes->tags);
+
+    // check limits
+    mMaxChannels = maxChannels;
+    if (mMaxChannels < 1) {
+        mMaxChannels = 1;
+    }
+    else if (mMaxChannels > 32) {
+        mMaxChannels = 32;
+    }
+    ALOGW_IF(maxChannels != mMaxChannels, "App requested %d channels", maxChannels);
+
+    mQuit = false;
+    mDecodeThread = 0;
+    memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
+    mAllocated = 0;
+    mNextSampleID = 0;
+    mNextChannelID = 0;
+
+    mCallback = 0;
+    mUserData = 0;
+
+    mChannelPool = new SoundChannel[mMaxChannels];
+    for (int i = 0; i < mMaxChannels; ++i) {
+        mChannelPool[i].init(this);
+        mChannels.push_back(&mChannelPool[i]);
+    }
+
+    // start decode thread
+    startThreads();
+}
+
+SoundPool::~SoundPool()
+{
+    ALOGV("SoundPool destructor");
+    mDecodeThread->quit();
+    quit();
+
+    Mutex::Autolock lock(&mLock);
+
+    mChannels.clear();
+    if (mChannelPool)
+        delete [] mChannelPool;
+    // clean up samples
+    ALOGV("clear samples");
+    mSamples.clear();
+
+    if (mDecodeThread)
+        delete mDecodeThread;
+}
+
+void SoundPool::addToRestartList(SoundChannel* channel)
+{
+    Mutex::Autolock lock(&mRestartLock);
+    if (!mQuit) {
+        mRestart.push_back(channel);
+        mCondition.signal();
+    }
+}
+
+void SoundPool::addToStopList(SoundChannel* channel)
+{
+    Mutex::Autolock lock(&mRestartLock);
+    if (!mQuit) {
+        mStop.push_back(channel);
+        mCondition.signal();
+    }
+}
+
+int SoundPool::beginThread(void* arg)
+{
+    SoundPool* p = (SoundPool*)arg;
+    return p->run();
+}
+
+int SoundPool::run()
+{
+    mRestartLock.lock();
+    while (!mQuit) {
+        mCondition.wait(mRestartLock);
+        ALOGV("awake");
+        if (mQuit) break;
+
+        while (!mStop.empty()) {
+            SoundChannel* channel;
+            ALOGV("Getting channel from stop list");
+            List<SoundChannel* >::iterator iter = mStop.begin();
+            channel = *iter;
+            mStop.erase(iter);
+            mRestartLock.unlock();
+            if (channel != 0) {
+                Mutex::Autolock lock(&mLock);
+                channel->stop();
+            }
+            mRestartLock.lock();
+            if (mQuit) break;
+        }
+
+        while (!mRestart.empty()) {
+            SoundChannel* channel;
+            ALOGV("Getting channel from list");
+            List<SoundChannel*>::iterator iter = mRestart.begin();
+            channel = *iter;
+            mRestart.erase(iter);
+            mRestartLock.unlock();
+            if (channel != 0) {
+                Mutex::Autolock lock(&mLock);
+                channel->nextEvent();
+            }
+            mRestartLock.lock();
+            if (mQuit) break;
+        }
+    }
+
+    mStop.clear();
+    mRestart.clear();
+    mCondition.signal();
+    mRestartLock.unlock();
+    ALOGV("goodbye");
+    return 0;
+}
+
+void SoundPool::quit()
+{
+    mRestartLock.lock();
+    mQuit = true;
+    mCondition.signal();
+    mCondition.wait(mRestartLock);
+    ALOGV("return from quit");
+    mRestartLock.unlock();
+}
+
+bool SoundPool::startThreads()
+{
+    createThreadEtc(beginThread, this, "SoundPool");
+    if (mDecodeThread == NULL)
+        mDecodeThread = new SoundPoolThread(this);
+    return mDecodeThread != NULL;
+}
+
+SoundChannel* SoundPool::findChannel(int channelID)
+{
+    for (int i = 0; i < mMaxChannels; ++i) {
+        if (mChannelPool[i].channelID() == channelID) {
+            return &mChannelPool[i];
+        }
+    }
+    return NULL;
+}
+
+SoundChannel* SoundPool::findNextChannel(int channelID)
+{
+    for (int i = 0; i < mMaxChannels; ++i) {
+        if (mChannelPool[i].nextChannelID() == channelID) {
+            return &mChannelPool[i];
+        }
+    }
+    return NULL;
+}
+
+int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
+{
+    ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", priority=%d",
+            fd, offset, length, priority);
+    Mutex::Autolock lock(&mLock);
+    sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
+    mSamples.add(sample->sampleID(), sample);
+    doLoad(sample);
+    return sample->sampleID();
+}
+
+void SoundPool::doLoad(sp<Sample>& sample)
+{
+    ALOGV("doLoad: loading sample sampleID=%d", sample->sampleID());
+    sample->startLoad();
+    mDecodeThread->loadSample(sample->sampleID());
+}
+
+bool SoundPool::unload(int sampleID)
+{
+    ALOGV("unload: sampleID=%d", sampleID);
+    Mutex::Autolock lock(&mLock);
+    return mSamples.removeItem(sampleID);
+}
+
+int SoundPool::play(int sampleID, float leftVolume, float rightVolume,
+        int priority, int loop, float rate)
+{
+    ALOGV("play sampleID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f",
+            sampleID, leftVolume, rightVolume, priority, loop, rate);
+    sp<Sample> sample;
+    SoundChannel* channel;
+    int channelID;
+
+    Mutex::Autolock lock(&mLock);
+
+    if (mQuit) {
+        return 0;
+    }
+    // is sample ready?
+    sample = findSample(sampleID);
+    if ((sample == 0) || (sample->state() != Sample::READY)) {
+        ALOGW("  sample %d not READY", sampleID);
+        return 0;
+    }
+
+    dump();
+
+    // allocate a channel
+    channel = allocateChannel_l(priority);
+
+    // no channel allocated - return 0
+    if (!channel) {
+        ALOGV("No channel allocated");
+        return 0;
+    }
+
+    channelID = ++mNextChannelID;
+
+    ALOGV("play channel %p state = %d", channel, channel->state());
+    channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
+    return channelID;
+}
+
+SoundChannel* SoundPool::allocateChannel_l(int priority)
+{
+    List<SoundChannel*>::iterator iter;
+    SoundChannel* channel = NULL;
+
+    // allocate a channel
+    if (!mChannels.empty()) {
+        iter = mChannels.begin();
+        if (priority >= (*iter)->priority()) {
+            channel = *iter;
+            mChannels.erase(iter);
+            ALOGV("Allocated active channel");
+        }
+    }
+
+    // update priority and put it back in the list
+    if (channel) {
+        channel->setPriority(priority);
+        for (iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
+            if (priority < (*iter)->priority()) {
+                break;
+            }
+        }
+        mChannels.insert(iter, channel);
+    }
+    return channel;
+}
+
+// move a channel from its current position to the front of the list
+void SoundPool::moveToFront_l(SoundChannel* channel)
+{
+    for (List<SoundChannel*>::iterator iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
+        if (*iter == channel) {
+            mChannels.erase(iter);
+            mChannels.push_front(channel);
+            break;
+        }
+    }
+}
+
+void SoundPool::pause(int channelID)
+{
+    ALOGV("pause(%d)", channelID);
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->pause();
+    }
+}
+
+void SoundPool::autoPause()
+{
+    ALOGV("autoPause()");
+    Mutex::Autolock lock(&mLock);
+    for (int i = 0; i < mMaxChannels; ++i) {
+        SoundChannel* channel = &mChannelPool[i];
+        channel->autoPause();
+    }
+}
+
+void SoundPool::resume(int channelID)
+{
+    ALOGV("resume(%d)", channelID);
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->resume();
+    }
+}
+
+void SoundPool::autoResume()
+{
+    ALOGV("autoResume()");
+    Mutex::Autolock lock(&mLock);
+    for (int i = 0; i < mMaxChannels; ++i) {
+        SoundChannel* channel = &mChannelPool[i];
+        channel->autoResume();
+    }
+}
+
+void SoundPool::stop(int channelID)
+{
+    ALOGV("stop(%d)", channelID);
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->stop();
+    } else {
+        channel = findNextChannel(channelID);
+        if (channel)
+            channel->clearNextEvent();
+    }
+}
+
+void SoundPool::setVolume(int channelID, float leftVolume, float rightVolume)
+{
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->setVolume(leftVolume, rightVolume);
+    }
+}
+
+void SoundPool::setPriority(int channelID, int priority)
+{
+    ALOGV("setPriority(%d, %d)", channelID, priority);
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->setPriority(priority);
+    }
+}
+
+void SoundPool::setLoop(int channelID, int loop)
+{
+    ALOGV("setLoop(%d, %d)", channelID, loop);
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->setLoop(loop);
+    }
+}
+
+void SoundPool::setRate(int channelID, float rate)
+{
+    ALOGV("setRate(%d, %f)", channelID, rate);
+    Mutex::Autolock lock(&mLock);
+    SoundChannel* channel = findChannel(channelID);
+    if (channel) {
+        channel->setRate(rate);
+    }
+}
+
+// call with lock held
+void SoundPool::done_l(SoundChannel* channel)
+{
+    ALOGV("done_l(%d)", channel->channelID());
+    // if "stolen", play next event
+    if (channel->nextChannelID() != 0) {
+        ALOGV("add to restart list");
+        addToRestartList(channel);
+    }
+
+    // return to idle state
+    else {
+        ALOGV("move to front");
+        moveToFront_l(channel);
+    }
+}
+
+void SoundPool::setCallback(SoundPoolCallback* callback, void* user)
+{
+    Mutex::Autolock lock(&mCallbackLock);
+    mCallback = callback;
+    mUserData = user;
+}
+
+void SoundPool::notify(SoundPoolEvent event)
+{
+    Mutex::Autolock lock(&mCallbackLock);
+    if (mCallback != NULL) {
+        mCallback(event, this, mUserData);
+    }
+}
+
+void SoundPool::dump()
+{
+    for (int i = 0; i < mMaxChannels; ++i) {
+        mChannelPool[i].dump();
+    }
+}
+
+
+Sample::Sample(int sampleID, int fd, int64_t offset, int64_t length)
+{
+    init();
+    mSampleID = sampleID;
+    mFd = dup(fd);
+    mOffset = offset;
+    mLength = length;
+    ALOGV("create sampleID=%d, fd=%d, offset=%" PRId64 " length=%" PRId64,
+        mSampleID, mFd, mLength, mOffset);
+}
+
+void Sample::init()
+{
+    mSize = 0;
+    mRefCount = 0;
+    mSampleID = 0;
+    mState = UNLOADED;
+    mFd = -1;
+    mOffset = 0;
+    mLength = 0;
+}
+
+Sample::~Sample()
+{
+    ALOGV("Sample::destructor sampleID=%d, fd=%d", mSampleID, mFd);
+    if (mFd > 0) {
+        ALOGV("close(%d)", mFd);
+        ::close(mFd);
+    }
+}
+
+static status_t decode(int fd, int64_t offset, int64_t length,
+        uint32_t *rate, int *numChannels, audio_format_t *audioFormat,
+        sp<MemoryHeapBase> heap, size_t *memsize) {
+
+    ALOGV("fd %d, offset %lld, size %lld", fd, offset, length);
+    AMediaExtractor *ex = AMediaExtractor_new();
+    status_t err = AMediaExtractor_setDataSourceFd(ex, fd, offset, length);
+
+    if (err != AMEDIA_OK) {
+        return err;
+    }
+
+    *audioFormat = AUDIO_FORMAT_PCM_16_BIT;
+
+    size_t numTracks = AMediaExtractor_getTrackCount(ex);
+    for (size_t i = 0; i < numTracks; i++) {
+        AMediaFormat *format = AMediaExtractor_getTrackFormat(ex, i);
+        const char *mime;
+        if (!AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime)) {
+            AMediaExtractor_delete(ex);
+            AMediaFormat_delete(format);
+            return UNKNOWN_ERROR;
+        }
+        if (strncmp(mime, "audio/", 6) == 0) {
+
+            AMediaCodec *codec = AMediaCodec_createDecoderByType(mime);
+            if (AMediaCodec_configure(codec, format,
+                    NULL /* window */, NULL /* drm */, 0 /* flags */) != AMEDIA_OK
+                || AMediaCodec_start(codec) != AMEDIA_OK
+                || AMediaExtractor_selectTrack(ex, i) != AMEDIA_OK) {
+                AMediaExtractor_delete(ex);
+                AMediaCodec_delete(codec);
+                AMediaFormat_delete(format);
+                return UNKNOWN_ERROR;
+            }
+
+            bool sawInputEOS = false;
+            bool sawOutputEOS = false;
+            uint8_t* writePos = static_cast<uint8_t*>(heap->getBase());
+            size_t available = heap->getSize();
+            size_t written = 0;
+
+            AMediaFormat_delete(format);
+            format = AMediaCodec_getOutputFormat(codec);
+
+            while (!sawOutputEOS) {
+                if (!sawInputEOS) {
+                    ssize_t bufidx = AMediaCodec_dequeueInputBuffer(codec, 5000);
+                    ALOGV("input buffer %d", bufidx);
+                    if (bufidx >= 0) {
+                        size_t bufsize;
+                        uint8_t *buf = AMediaCodec_getInputBuffer(codec, bufidx, &bufsize);
+                        int sampleSize = AMediaExtractor_readSampleData(ex, buf, bufsize);
+                        ALOGV("read %d", sampleSize);
+                        if (sampleSize < 0) {
+                            sampleSize = 0;
+                            sawInputEOS = true;
+                            ALOGV("EOS");
+                        }
+                        int64_t presentationTimeUs = AMediaExtractor_getSampleTime(ex);
+
+                        AMediaCodec_queueInputBuffer(codec, bufidx,
+                                0 /* offset */, sampleSize, presentationTimeUs,
+                                sawInputEOS ? AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM : 0);
+                        AMediaExtractor_advance(ex);
+                    }
+                }
+
+                AMediaCodecBufferInfo info;
+                int status = AMediaCodec_dequeueOutputBuffer(codec, &info, 1);
+                ALOGV("dequeueoutput returned: %d", status);
+                if (status >= 0) {
+                    if (info.flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) {
+                        ALOGV("output EOS");
+                        sawOutputEOS = true;
+                    }
+                    ALOGV("got decoded buffer size %d", info.size);
+
+                    uint8_t *buf = AMediaCodec_getOutputBuffer(codec, status, NULL /* out_size */);
+                    size_t dataSize = info.size;
+                    if (dataSize > available) {
+                        dataSize = available;
+                    }
+                    memcpy(writePos, buf + info.offset, dataSize);
+                    writePos += dataSize;
+                    written += dataSize;
+                    available -= dataSize;
+                    AMediaCodec_releaseOutputBuffer(codec, status, false /* render */);
+                    if (available == 0) {
+                        // there might be more data, but there's no space for it
+                        sawOutputEOS = true;
+                    }
+                } else if (status == AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED) {
+                    ALOGV("output buffers changed");
+                } else if (status == AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED) {
+                    AMediaFormat_delete(format);
+                    format = AMediaCodec_getOutputFormat(codec);
+                    ALOGV("format changed to: %s", AMediaFormat_toString(format));
+                } else if (status == AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
+                    ALOGV("no output buffer right now");
+                } else {
+                    ALOGV("unexpected info code: %d", status);
+                }
+            }
+
+            AMediaCodec_stop(codec);
+            AMediaCodec_delete(codec);
+            AMediaExtractor_delete(ex);
+            if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, (int32_t*) rate) ||
+                    !AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, numChannels)) {
+                AMediaFormat_delete(format);
+                return UNKNOWN_ERROR;
+            }
+            AMediaFormat_delete(format);
+            *memsize = written;
+            return OK;
+        }
+        AMediaFormat_delete(format);
+    }
+    AMediaExtractor_delete(ex);
+    return UNKNOWN_ERROR;
+}
+
+status_t Sample::doLoad()
+{
+    uint32_t sampleRate;
+    int numChannels;
+    audio_format_t format;
+    status_t status;
+    mHeap = new MemoryHeapBase(kDefaultHeapSize);
+
+    ALOGV("Start decode");
+    status = decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
+                                 mHeap, &mSize);
+    ALOGV("close(%d)", mFd);
+    ::close(mFd);
+    mFd = -1;
+    if (status != NO_ERROR) {
+        ALOGE("Unable to load sample");
+        goto error;
+    }
+    ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d",
+          mHeap->getBase(), mSize, sampleRate, numChannels);
+
+    if (sampleRate > kMaxSampleRate) {
+       ALOGE("Sample rate (%u) out of range", sampleRate);
+       status = BAD_VALUE;
+       goto error;
+    }
+
+    if ((numChannels < 1) || (numChannels > 2)) {
+        ALOGE("Sample channel count (%d) out of range", numChannels);
+        status = BAD_VALUE;
+        goto error;
+    }
+
+    mData = new MemoryBase(mHeap, 0, mSize);
+    mSampleRate = sampleRate;
+    mNumChannels = numChannels;
+    mFormat = format;
+    mState = READY;
+    return NO_ERROR;
+
+error:
+    mHeap.clear();
+    return status;
+}
+
+
+void SoundChannel::init(SoundPool* soundPool)
+{
+    mSoundPool = soundPool;
+}
+
+// call with sound pool lock held
+void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
+        float rightVolume, int priority, int loop, float rate)
+{
+    sp<AudioTrack> oldTrack;
+    sp<AudioTrack> newTrack;
+    status_t status;
+
+    { // scope for the lock
+        Mutex::Autolock lock(&mLock);
+
+        ALOGV("SoundChannel::play %p: sampleID=%d, channelID=%d, leftVolume=%f, rightVolume=%f,"
+                " priority=%d, loop=%d, rate=%f",
+                this, sample->sampleID(), nextChannelID, leftVolume, rightVolume,
+                priority, loop, rate);
+
+        // if not idle, this voice is being stolen
+        if (mState != IDLE) {
+            ALOGV("channel %d stolen - event queued for channel %d", channelID(), nextChannelID);
+            mNextEvent.set(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
+            stop_l();
+            return;
+        }
+
+        // initialize track
+        size_t afFrameCount;
+        uint32_t afSampleRate;
+        audio_stream_type_t streamType = audio_attributes_to_stream_type(mSoundPool->attributes());
+        if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
+            afFrameCount = kDefaultFrameCount;
+        }
+        if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
+            afSampleRate = kDefaultSampleRate;
+        }
+        int numChannels = sample->numChannels();
+        uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5);
+        size_t frameCount = 0;
+
+        if (loop) {
+            frameCount = sample->size()/numChannels/
+                ((sample->format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t));
+        }
+
+#ifndef USE_SHARED_MEM_BUFFER
+        uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate;
+        // Ensure minimum audio buffer size in case of short looped sample
+        if(frameCount < totalFrames) {
+            frameCount = totalFrames;
+        }
+#endif
+
+        // mToggle toggles each time a track is started on a given channel.
+        // The toggle is concatenated with the SoundChannel address and passed to AudioTrack
+        // as callback user data. This enables the detection of callbacks received from the old
+        // audio track while the new one is being started and avoids processing them with
+        // wrong audio audio buffer size  (mAudioBufferSize)
+        unsigned long toggle = mToggle ^ 1;
+        void *userData = (void *)((unsigned long)this | toggle);
+        audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels);
+
+        // do not create a new audio track if current track is compatible with sample parameters
+#ifdef USE_SHARED_MEM_BUFFER
+        newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
+                channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
+#else
+        uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount;
+        newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
+                channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
+                bufferFrames);
+#endif
+        oldTrack = mAudioTrack;
+        status = newTrack->initCheck();
+        if (status != NO_ERROR) {
+            ALOGE("Error creating AudioTrack");
+            goto exit;
+        }
+        ALOGV("setVolume %p", newTrack.get());
+        newTrack->setVolume(leftVolume, rightVolume);
+        newTrack->setLoop(0, frameCount, loop);
+
+        // From now on, AudioTrack callbacks received with previous toggle value will be ignored.
+        mToggle = toggle;
+        mAudioTrack = newTrack;
+        mPos = 0;
+        mSample = sample;
+        mChannelID = nextChannelID;
+        mPriority = priority;
+        mLoop = loop;
+        mLeftVolume = leftVolume;
+        mRightVolume = rightVolume;
+        mNumChannels = numChannels;
+        mRate = rate;
+        clearNextEvent();
+        mState = PLAYING;
+        mAudioTrack->start();
+        mAudioBufferSize = newTrack->frameCount()*newTrack->frameSize();
+    }
+
+exit:
+    ALOGV("delete oldTrack %p", oldTrack.get());
+    if (status != NO_ERROR) {
+        mAudioTrack.clear();
+    }
+}
+
+void SoundChannel::nextEvent()
+{
+    sp<Sample> sample;
+    int nextChannelID;
+    float leftVolume;
+    float rightVolume;
+    int priority;
+    int loop;
+    float rate;
+
+    // check for valid event
+    {
+        Mutex::Autolock lock(&mLock);
+        nextChannelID = mNextEvent.channelID();
+        if (nextChannelID  == 0) {
+            ALOGV("stolen channel has no event");
+            return;
+        }
+
+        sample = mNextEvent.sample();
+        leftVolume = mNextEvent.leftVolume();
+        rightVolume = mNextEvent.rightVolume();
+        priority = mNextEvent.priority();
+        loop = mNextEvent.loop();
+        rate = mNextEvent.rate();
+    }
+
+    ALOGV("Starting stolen channel %d -> %d", channelID(), nextChannelID);
+    play(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
+}
+
+void SoundChannel::callback(int event, void* user, void *info)
+{
+    SoundChannel* channel = static_cast<SoundChannel*>((void *)((unsigned long)user & ~1));
+
+    channel->process(event, info, (unsigned long)user & 1);
+}
+
+void SoundChannel::process(int event, void *info, unsigned long toggle)
+{
+    //ALOGV("process(%d)", mChannelID);
+
+    Mutex::Autolock lock(&mLock);
+
+    AudioTrack::Buffer* b = NULL;
+    if (event == AudioTrack::EVENT_MORE_DATA) {
+       b = static_cast<AudioTrack::Buffer *>(info);
+    }
+
+    if (mToggle != toggle) {
+        ALOGV("process wrong toggle %p channel %d", this, mChannelID);
+        if (b != NULL) {
+            b->size = 0;
+        }
+        return;
+    }
+
+    sp<Sample> sample = mSample;
+
+//    ALOGV("SoundChannel::process event %d", event);
+
+    if (event == AudioTrack::EVENT_MORE_DATA) {
+
+        // check for stop state
+        if (b->size == 0) return;
+
+        if (mState == IDLE) {
+            b->size = 0;
+            return;
+        }
+
+        if (sample != 0) {
+            // fill buffer
+            uint8_t* q = (uint8_t*) b->i8;
+            size_t count = 0;
+
+            if (mPos < (int)sample->size()) {
+                uint8_t* p = sample->data() + mPos;
+                count = sample->size() - mPos;
+                if (count > b->size) {
+                    count = b->size;
+                }
+                memcpy(q, p, count);
+//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size,
+//                      count);
+            } else if (mPos < mAudioBufferSize) {
+                count = mAudioBufferSize - mPos;
+                if (count > b->size) {
+                    count = b->size;
+                }
+                memset(q, 0, count);
+//              ALOGV("fill extra: q=%p, mPos=%u, b->size=%u, count=%d", q, mPos, b->size, count);
+            }
+
+            mPos += count;
+            b->size = count;
+            //ALOGV("buffer=%p, [0]=%d", b->i16, b->i16[0]);
+        }
+    } else if (event == AudioTrack::EVENT_UNDERRUN || event == AudioTrack::EVENT_BUFFER_END) {
+        ALOGV("process %p channel %d event %s",
+              this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" :
+                      "BUFFER_END");
+        mSoundPool->addToStopList(this);
+    } else if (event == AudioTrack::EVENT_LOOP_END) {
+        ALOGV("End loop %p channel %d", this, mChannelID);
+    } else if (event == AudioTrack::EVENT_NEW_IAUDIOTRACK) {
+        ALOGV("process %p channel %d NEW_IAUDIOTRACK", this, mChannelID);
+    } else {
+        ALOGW("SoundChannel::process unexpected event %d", event);
+    }
+}
+
+
+// call with lock held
+bool SoundChannel::doStop_l()
+{
+    if (mState != IDLE) {
+        setVolume_l(0, 0);
+        ALOGV("stop");
+        mAudioTrack->stop();
+        mSample.clear();
+        mState = IDLE;
+        mPriority = IDLE_PRIORITY;
+        return true;
+    }
+    return false;
+}
+
+// call with lock held and sound pool lock held
+void SoundChannel::stop_l()
+{
+    if (doStop_l()) {
+        mSoundPool->done_l(this);
+    }
+}
+
+// call with sound pool lock held
+void SoundChannel::stop()
+{
+    bool stopped;
+    {
+        Mutex::Autolock lock(&mLock);
+        stopped = doStop_l();
+    }
+
+    if (stopped) {
+        mSoundPool->done_l(this);
+    }
+}
+
+//FIXME: Pause is a little broken right now
+void SoundChannel::pause()
+{
+    Mutex::Autolock lock(&mLock);
+    if (mState == PLAYING) {
+        ALOGV("pause track");
+        mState = PAUSED;
+        mAudioTrack->pause();
+    }
+}
+
+void SoundChannel::autoPause()
+{
+    Mutex::Autolock lock(&mLock);
+    if (mState == PLAYING) {
+        ALOGV("pause track");
+        mState = PAUSED;
+        mAutoPaused = true;
+        mAudioTrack->pause();
+    }
+}
+
+void SoundChannel::resume()
+{
+    Mutex::Autolock lock(&mLock);
+    if (mState == PAUSED) {
+        ALOGV("resume track");
+        mState = PLAYING;
+        mAutoPaused = false;
+        mAudioTrack->start();
+    }
+}
+
+void SoundChannel::autoResume()
+{
+    Mutex::Autolock lock(&mLock);
+    if (mAutoPaused && (mState == PAUSED)) {
+        ALOGV("resume track");
+        mState = PLAYING;
+        mAutoPaused = false;
+        mAudioTrack->start();
+    }
+}
+
+void SoundChannel::setRate(float rate)
+{
+    Mutex::Autolock lock(&mLock);
+    if (mAudioTrack != NULL && mSample != 0) {
+        uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5);
+        mAudioTrack->setSampleRate(sampleRate);
+        mRate = rate;
+    }
+}
+
+// call with lock held
+void SoundChannel::setVolume_l(float leftVolume, float rightVolume)
+{
+    mLeftVolume = leftVolume;
+    mRightVolume = rightVolume;
+    if (mAudioTrack != NULL)
+        mAudioTrack->setVolume(leftVolume, rightVolume);
+}
+
+void SoundChannel::setVolume(float leftVolume, float rightVolume)
+{
+    Mutex::Autolock lock(&mLock);
+    setVolume_l(leftVolume, rightVolume);
+}
+
+void SoundChannel::setLoop(int loop)
+{
+    Mutex::Autolock lock(&mLock);
+    if (mAudioTrack != NULL && mSample != 0) {
+        uint32_t loopEnd = mSample->size()/mNumChannels/
+            ((mSample->format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t));
+        mAudioTrack->setLoop(0, loopEnd, loop);
+        mLoop = loop;
+    }
+}
+
+SoundChannel::~SoundChannel()
+{
+    ALOGV("SoundChannel destructor %p", this);
+    {
+        Mutex::Autolock lock(&mLock);
+        clearNextEvent();
+        doStop_l();
+    }
+    // do not call AudioTrack destructor with mLock held as it will wait for the AudioTrack
+    // callback thread to exit which may need to execute process() and acquire the mLock.
+    mAudioTrack.clear();
+}
+
+void SoundChannel::dump()
+{
+    ALOGV("mState = %d mChannelID=%d, mNumChannels=%d, mPos = %d, mPriority=%d, mLoop=%d",
+            mState, mChannelID, mNumChannels, mPos, mPriority, mLoop);
+}
+
+void SoundEvent::set(const sp<Sample>& sample, int channelID, float leftVolume,
+            float rightVolume, int priority, int loop, float rate)
+{
+    mSample = sample;
+    mChannelID = channelID;
+    mLeftVolume = leftVolume;
+    mRightVolume = rightVolume;
+    mPriority = priority;
+    mLoop = loop;
+    mRate =rate;
+}
+
+} // end namespace android
diff --git a/media/jni/soundpool/SoundPool.h b/media/jni/soundpool/SoundPool.h
new file mode 100644
index 0000000..9d9cbdf
--- /dev/null
+++ b/media/jni/soundpool/SoundPool.h
@@ -0,0 +1,232 @@
+/*
+ * Copyright (C) 2007 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.
+ */
+
+#ifndef SOUNDPOOL_H_
+#define SOUNDPOOL_H_
+
+#include <utils/threads.h>
+#include <utils/List.h>
+#include <utils/Vector.h>
+#include <utils/KeyedVector.h>
+#include <media/AudioTrack.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/MemoryBase.h>
+
+namespace android {
+
+static const int IDLE_PRIORITY = -1;
+
+// forward declarations
+class SoundEvent;
+class SoundPoolThread;
+class SoundPool;
+
+// for queued events
+class SoundPoolEvent {
+public:
+    SoundPoolEvent(int msg, int arg1=0, int arg2=0) :
+        mMsg(msg), mArg1(arg1), mArg2(arg2) {}
+    int         mMsg;
+    int         mArg1;
+    int         mArg2;
+    enum MessageType { INVALID, SAMPLE_LOADED };
+};
+
+// callback function prototype
+typedef void SoundPoolCallback(SoundPoolEvent event, SoundPool* soundPool, void* user);
+
+// tracks samples used by application
+class Sample  : public RefBase {
+public:
+    enum sample_state { UNLOADED, LOADING, READY, UNLOADING };
+    Sample(int sampleID, int fd, int64_t offset, int64_t length);
+    ~Sample();
+    int sampleID() { return mSampleID; }
+    int numChannels() { return mNumChannels; }
+    int sampleRate() { return mSampleRate; }
+    audio_format_t format() { return mFormat; }
+    size_t size() { return mSize; }
+    int state() { return mState; }
+    uint8_t* data() { return static_cast<uint8_t*>(mData->pointer()); }
+    status_t doLoad();
+    void startLoad() { mState = LOADING; }
+    sp<IMemory> getIMemory() { return mData; }
+
+private:
+    void init();
+
+    size_t              mSize;
+    volatile int32_t    mRefCount;
+    uint16_t            mSampleID;
+    uint16_t            mSampleRate;
+    uint8_t             mState : 3;
+    uint8_t             mNumChannels : 2;
+    audio_format_t      mFormat;
+    int                 mFd;
+    int64_t             mOffset;
+    int64_t             mLength;
+    sp<IMemory>         mData;
+    sp<MemoryHeapBase>  mHeap;
+};
+
+// stores pending events for stolen channels
+class SoundEvent
+{
+public:
+    SoundEvent() : mChannelID(0), mLeftVolume(0), mRightVolume(0),
+            mPriority(IDLE_PRIORITY), mLoop(0), mRate(0) {}
+    void set(const sp<Sample>& sample, int channelID, float leftVolume,
+            float rightVolume, int priority, int loop, float rate);
+    sp<Sample>      sample() { return mSample; }
+    int             channelID() { return mChannelID; }
+    float           leftVolume() { return mLeftVolume; }
+    float           rightVolume() { return mRightVolume; }
+    int             priority() { return mPriority; }
+    int             loop() { return mLoop; }
+    float           rate() { return mRate; }
+    void            clear() { mChannelID = 0; mSample.clear(); }
+
+protected:
+    sp<Sample>      mSample;
+    int             mChannelID;
+    float           mLeftVolume;
+    float           mRightVolume;
+    int             mPriority;
+    int             mLoop;
+    float           mRate;
+};
+
+// for channels aka AudioTracks
+class SoundChannel : public SoundEvent {
+public:
+    enum state { IDLE, RESUMING, STOPPING, PAUSED, PLAYING };
+    SoundChannel() : mState(IDLE), mNumChannels(1),
+            mPos(0), mToggle(0), mAutoPaused(false) {}
+    ~SoundChannel();
+    void init(SoundPool* soundPool);
+    void play(const sp<Sample>& sample, int channelID, float leftVolume, float rightVolume,
+            int priority, int loop, float rate);
+    void setVolume_l(float leftVolume, float rightVolume);
+    void setVolume(float leftVolume, float rightVolume);
+    void stop_l();
+    void stop();
+    void pause();
+    void autoPause();
+    void resume();
+    void autoResume();
+    void setRate(float rate);
+    int state() { return mState; }
+    void setPriority(int priority) { mPriority = priority; }
+    void setLoop(int loop);
+    int numChannels() { return mNumChannels; }
+    void clearNextEvent() { mNextEvent.clear(); }
+    void nextEvent();
+    int nextChannelID() { return mNextEvent.channelID(); }
+    void dump();
+
+private:
+    static void callback(int event, void* user, void *info);
+    void process(int event, void *info, unsigned long toggle);
+    bool doStop_l();
+
+    SoundPool*          mSoundPool;
+    sp<AudioTrack>      mAudioTrack;
+    SoundEvent          mNextEvent;
+    Mutex               mLock;
+    int                 mState;
+    int                 mNumChannels;
+    int                 mPos;
+    int                 mAudioBufferSize;
+    unsigned long       mToggle;
+    bool                mAutoPaused;
+};
+
+// application object for managing a pool of sounds
+class SoundPool {
+    friend class SoundPoolThread;
+    friend class SoundChannel;
+public:
+    SoundPool(int maxChannels, const audio_attributes_t* pAttributes);
+    ~SoundPool();
+    int load(int fd, int64_t offset, int64_t length, int priority);
+    bool unload(int sampleID);
+    int play(int sampleID, float leftVolume, float rightVolume, int priority,
+            int loop, float rate);
+    void pause(int channelID);
+    void autoPause();
+    void resume(int channelID);
+    void autoResume();
+    void stop(int channelID);
+    void setVolume(int channelID, float leftVolume, float rightVolume);
+    void setPriority(int channelID, int priority);
+    void setLoop(int channelID, int loop);
+    void setRate(int channelID, float rate);
+    const audio_attributes_t* attributes() { return &mAttributes; }
+
+    // called from SoundPoolThread
+    void sampleLoaded(int sampleID);
+
+    // called from AudioTrack thread
+    void done_l(SoundChannel* channel);
+
+    // callback function
+    void setCallback(SoundPoolCallback* callback, void* user);
+    void* getUserData() { return mUserData; }
+
+private:
+    SoundPool() {} // no default constructor
+    bool startThreads();
+    void doLoad(sp<Sample>& sample);
+    sp<Sample> findSample(int sampleID) { return mSamples.valueFor(sampleID); }
+    SoundChannel* findChannel (int channelID);
+    SoundChannel* findNextChannel (int channelID);
+    SoundChannel* allocateChannel_l(int priority);
+    void moveToFront_l(SoundChannel* channel);
+    void notify(SoundPoolEvent event);
+    void dump();
+
+    // restart thread
+    void addToRestartList(SoundChannel* channel);
+    void addToStopList(SoundChannel* channel);
+    static int beginThread(void* arg);
+    int run();
+    void quit();
+
+    Mutex                   mLock;
+    Mutex                   mRestartLock;
+    Condition               mCondition;
+    SoundPoolThread*        mDecodeThread;
+    SoundChannel*           mChannelPool;
+    List<SoundChannel*>     mChannels;
+    List<SoundChannel*>     mRestart;
+    List<SoundChannel*>     mStop;
+    DefaultKeyedVector< int, sp<Sample> >   mSamples;
+    int                     mMaxChannels;
+    audio_attributes_t      mAttributes;
+    int                     mAllocated;
+    int                     mNextSampleID;
+    int                     mNextChannelID;
+    bool                    mQuit;
+
+    // callback
+    Mutex                   mCallbackLock;
+    SoundPoolCallback*      mCallback;
+    void*                   mUserData;
+};
+
+} // end namespace android
+
+#endif /*SOUNDPOOL_H_*/
diff --git a/media/jni/soundpool/SoundPoolThread.cpp b/media/jni/soundpool/SoundPoolThread.cpp
new file mode 100644
index 0000000..ba3b482
--- /dev/null
+++ b/media/jni/soundpool/SoundPoolThread.cpp
@@ -0,0 +1,114 @@
+/*
+ * Copyright (C) 2007 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoundPoolThread"
+#include "utils/Log.h"
+
+#include "SoundPoolThread.h"
+
+namespace android {
+
+void SoundPoolThread::write(SoundPoolMsg msg) {
+    Mutex::Autolock lock(&mLock);
+    while (mMsgQueue.size() >= maxMessages) {
+        mCondition.wait(mLock);
+    }
+
+    // if thread is quitting, don't add to queue
+    if (mRunning) {
+        mMsgQueue.push(msg);
+        mCondition.signal();
+    }
+}
+
+const SoundPoolMsg SoundPoolThread::read() {
+    Mutex::Autolock lock(&mLock);
+    while (mMsgQueue.size() == 0) {
+        mCondition.wait(mLock);
+    }
+    SoundPoolMsg msg = mMsgQueue[0];
+    mMsgQueue.removeAt(0);
+    mCondition.signal();
+    return msg;
+}
+
+void SoundPoolThread::quit() {
+    Mutex::Autolock lock(&mLock);
+    if (mRunning) {
+        mRunning = false;
+        mMsgQueue.clear();
+        mMsgQueue.push(SoundPoolMsg(SoundPoolMsg::KILL, 0));
+        mCondition.signal();
+        mCondition.wait(mLock);
+    }
+    ALOGV("return from quit");
+}
+
+SoundPoolThread::SoundPoolThread(SoundPool* soundPool) :
+    mSoundPool(soundPool)
+{
+    mMsgQueue.setCapacity(maxMessages);
+    if (createThreadEtc(beginThread, this, "SoundPoolThread")) {
+        mRunning = true;
+    }
+}
+
+SoundPoolThread::~SoundPoolThread()
+{
+    quit();
+}
+
+int SoundPoolThread::beginThread(void* arg) {
+    ALOGV("beginThread");
+    SoundPoolThread* soundPoolThread = (SoundPoolThread*)arg;
+    return soundPoolThread->run();
+}
+
+int SoundPoolThread::run() {
+    ALOGV("run");
+    for (;;) {
+        SoundPoolMsg msg = read();
+        ALOGV("Got message m=%d, mData=%d", msg.mMessageType, msg.mData);
+        switch (msg.mMessageType) {
+        case SoundPoolMsg::KILL:
+            ALOGV("goodbye");
+            return NO_ERROR;
+        case SoundPoolMsg::LOAD_SAMPLE:
+            doLoadSample(msg.mData);
+            break;
+        default:
+            ALOGW("run: Unrecognized message %d\n",
+                    msg.mMessageType);
+            break;
+        }
+    }
+}
+
+void SoundPoolThread::loadSample(int sampleID) {
+    write(SoundPoolMsg(SoundPoolMsg::LOAD_SAMPLE, sampleID));
+}
+
+void SoundPoolThread::doLoadSample(int sampleID) {
+    sp <Sample> sample = mSoundPool->findSample(sampleID);
+    status_t status = -1;
+    if (sample != 0) {
+        status = sample->doLoad();
+    }
+    mSoundPool->notify(SoundPoolEvent(SoundPoolEvent::SAMPLE_LOADED, sampleID, status));
+}
+
+} // end namespace android
diff --git a/media/jni/soundpool/SoundPoolThread.h b/media/jni/soundpool/SoundPoolThread.h
new file mode 100644
index 0000000..d388388
--- /dev/null
+++ b/media/jni/soundpool/SoundPoolThread.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2007 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.
+ */
+
+#ifndef SOUNDPOOLTHREAD_H_
+#define SOUNDPOOLTHREAD_H_
+
+#include <utils/threads.h>
+#include <utils/Vector.h>
+#include <media/AudioTrack.h>
+
+#include "SoundPool.h"
+
+namespace android {
+
+class SoundPoolMsg {
+public:
+    enum MessageType { INVALID, KILL, LOAD_SAMPLE };
+    SoundPoolMsg() : mMessageType(INVALID), mData(0) {}
+    SoundPoolMsg(MessageType MessageType, int data) :
+        mMessageType(MessageType), mData(data) {}
+    uint16_t         mMessageType;
+    uint16_t         mData;
+};
+
+/*
+ * This class handles background requests from the SoundPool
+ */
+class SoundPoolThread {
+public:
+    SoundPoolThread(SoundPool* SoundPool);
+    ~SoundPoolThread();
+    void loadSample(int sampleID);
+    void quit();
+    void write(SoundPoolMsg msg);
+
+private:
+    static const size_t maxMessages = 5;
+
+    static int beginThread(void* arg);
+    int run();
+    void doLoadSample(int sampleID);
+    const SoundPoolMsg read();
+
+    Mutex                   mLock;
+    Condition               mCondition;
+    Vector<SoundPoolMsg>    mMsgQueue;
+    SoundPool*              mSoundPool;
+    bool                    mRunning;
+};
+
+} // end namespace android
+
+#endif /*SOUNDPOOLTHREAD_H_*/
diff --git a/media/jni/soundpool/android_media_SoundPool_SoundPoolImpl.cpp b/media/jni/soundpool/android_media_SoundPool_SoundPoolImpl.cpp
index baf61d5..b2333f8 100644
--- a/media/jni/soundpool/android_media_SoundPool_SoundPoolImpl.cpp
+++ b/media/jni/soundpool/android_media_SoundPool_SoundPoolImpl.cpp
@@ -23,7 +23,7 @@
 #include <nativehelper/jni.h>
 #include <nativehelper/JNIHelp.h>
 #include <android_runtime/AndroidRuntime.h>
-#include <media/SoundPool.h>
+#include "SoundPool.h"
 
 using namespace android;
 
@@ -45,20 +45,6 @@
 static audio_attributes_fields_t javaAudioAttrFields;
 
 // ----------------------------------------------------------------------------
-static jint
-android_media_SoundPool_SoundPoolImpl_load_URL(JNIEnv *env, jobject thiz, jstring path, jint priority)
-{
-    ALOGV("android_media_SoundPool_SoundPoolImpl_load_URL");
-    SoundPool *ap = MusterSoundPool(env, thiz);
-    if (path == NULL) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
-        return 0;
-    }
-    const char* s = env->GetStringUTFChars(path, NULL);
-    int id = ap->load(s, priority);
-    env->ReleaseStringUTFChars(path, s);
-    return (jint) id;
-}
 
 static jint
 android_media_SoundPool_SoundPoolImpl_load_FD(JNIEnv *env, jobject thiz, jobject fileDescriptor,
@@ -249,10 +235,6 @@
 // Dalvik VM type signatures
 static JNINativeMethod gMethods[] = {
     {   "_load",
-        "(Ljava/lang/String;I)I",
-        (void *)android_media_SoundPool_SoundPoolImpl_load_URL
-    },
-    {   "_load",
         "(Ljava/io/FileDescriptor;JJI)I",
         (void *)android_media_SoundPool_SoundPoolImpl_load_FD
     },