Merge "Remove MediaPlayerService::decode()"

14 files changed, 13 insertions, 1920 deletions

diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index d7e584a..67b599a 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -51,17 +51,6 @@ public:
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
     virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
 
-    virtual status_t         decode(
-            const sp<IMediaHTTPService> &httpService,
-            const char* url,
-            uint32_t *pSampleRate,
-            int* pNumChannels,
-            audio_format_t* pFormat,
-            const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
-
-    virtual status_t         decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
-                                    int* pNumChannels, audio_format_t* pFormat,
-                                    const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
     virtual sp<IOMX>            getOMX() = 0;
     virtual sp<ICrypto>         makeCrypto() = 0;
     virtual sp<IDrm>            makeDrm() = 0;
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 0f38c16..b980421 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -88,7 +88,6 @@ public:
 
         virtual             ~AudioSink() {}
         virtual bool        ready() const = 0; // audio output is open and ready
-        virtual bool        realtime() const = 0; // audio output is real-time output
         virtual ssize_t     bufferSize() const = 0;
         virtual ssize_t     frameCount() const = 0;
         virtual ssize_t     channelCount() const = 0;
diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h
deleted file mode 100644
index 5830475..0000000
--- a/include/media/SoundPool.h
+++ /dev/null
@@ -1,241 +0,0 @@
-/*
- * 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, const char* url);
-    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; }
-
-    // hack
-    void init(int numChannels, int sampleRate, audio_format_t format, size_t size,
-            sp<IMemory> data ) {
-        mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size;
-            mData = data; }
-
-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;
-    char*               mUrl;
-    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(const char* url, int priority);
-    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/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 9cc208e..5830933 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -232,17 +232,6 @@ public:
             bool            isLooping();
             status_t        setVolume(float leftVolume, float rightVolume);
             void            notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
-    static  status_t        decode(
-            const sp<IMediaHTTPService> &httpService,
-            const char* url,
-            uint32_t *pSampleRate,
-            int* pNumChannels,
-            audio_format_t* pFormat,
-            const sp<IMemoryHeap>& heap,
-            size_t *pSize);
-    static  status_t        decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
-                                   int* pNumChannels, audio_format_t* pFormat,
-                                   const sp<IMemoryHeap>& heap, size_t *pSize);
             status_t        invoke(const Parcel& request, Parcel *reply);
             status_t        setMetadataFilter(const Parcel& filter);
             status_t        getMetadata(bool update_only, bool apply_filter, Parcel *metadata);
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index df6eb57..b91b7d1 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -58,8 +58,6 @@ LOCAL_SRC_FILES:= \
     AudioEffect.cpp \
     Visualizer.cpp \
     MemoryLeakTrackUtil.cpp \
-    SoundPool.cpp \
-    SoundPoolThread.cpp \
     StringArray.cpp
 
 LOCAL_SRC_FILES += ../libnbaio/roundup.c
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 4d33d40..feea267 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -39,8 +39,6 @@ namespace android {
 
 enum {
     CREATE = IBinder::FIRST_CALL_TRANSACTION,
-    DECODE_URL,
-    DECODE_FD,
     CREATE_MEDIA_RECORDER,
     CREATE_METADATA_RETRIEVER,
     GET_OMX,
@@ -88,59 +86,6 @@ public:
         return interface_cast<IMediaRecorder>(reply.readStrongBinder());
     }
 
-    virtual status_t decode(
-            const sp<IMediaHTTPService> &httpService,
-            const char* url,
-            uint32_t *pSampleRate,
-            int* pNumChannels,
-            audio_format_t* pFormat,
-            const sp<IMemoryHeap>& heap,
-            size_t *pSize)
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
-        data.writeInt32(httpService != NULL);
-        if (httpService != NULL) {
-            data.writeStrongBinder(IInterface::asBinder(httpService));
-        }
-        data.writeCString(url);
-        data.writeStrongBinder(IInterface::asBinder(heap));
-        status_t status = remote()->transact(DECODE_URL, data, &reply);
-        if (status == NO_ERROR) {
-            status = (status_t)reply.readInt32();
-            if (status == NO_ERROR) {
-                *pSampleRate = uint32_t(reply.readInt32());
-                *pNumChannels = reply.readInt32();
-                *pFormat = (audio_format_t)reply.readInt32();
-                *pSize = (size_t)reply.readInt32();
-            }
-        }
-        return status;
-    }
-
-    virtual status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
-                               int* pNumChannels, audio_format_t* pFormat,
-                               const sp<IMemoryHeap>& heap, size_t *pSize)
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
-        data.writeFileDescriptor(fd);
-        data.writeInt64(offset);
-        data.writeInt64(length);
-        data.writeStrongBinder(IInterface::asBinder(heap));
-        status_t status = remote()->transact(DECODE_FD, data, &reply);
-        if (status == NO_ERROR) {
-            status = (status_t)reply.readInt32();
-            if (status == NO_ERROR) {
-                *pSampleRate = uint32_t(reply.readInt32());
-                *pNumChannels = reply.readInt32();
-                *pFormat = (audio_format_t)reply.readInt32();
-                *pSize = (size_t)reply.readInt32();
-            }
-        }
-        return status;
-    }
-
     virtual sp<IOMX> getOMX() {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -219,58 +164,6 @@ status_t BnMediaPlayerService::onTransact(
             reply->writeStrongBinder(IInterface::asBinder(player));
             return NO_ERROR;
         } break;
-        case DECODE_URL: {
-            CHECK_INTERFACE(IMediaPlayerService, data, reply);
-            sp<IMediaHTTPService> httpService;
-            if (data.readInt32()) {
-                httpService =
-                    interface_cast<IMediaHTTPService>(data.readStrongBinder());
-            }
-            const char* url = data.readCString();
-            sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
-            uint32_t sampleRate;
-            int numChannels;
-            audio_format_t format;
-            size_t size;
-            status_t status =
-                decode(httpService,
-                       url,
-                       &sampleRate,
-                       &numChannels,
-                       &format,
-                       heap,
-                       &size);
-            reply->writeInt32(status);
-            if (status == NO_ERROR) {
-                reply->writeInt32(sampleRate);
-                reply->writeInt32(numChannels);
-                reply->writeInt32((int32_t)format);
-                reply->writeInt32((int32_t)size);
-            }
-            return NO_ERROR;
-        } break;
-        case DECODE_FD: {
-            CHECK_INTERFACE(IMediaPlayerService, data, reply);
-            int fd = dup(data.readFileDescriptor());
-            int64_t offset = data.readInt64();
-            int64_t length = data.readInt64();
-            sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
-            uint32_t sampleRate;
-            int numChannels;
-            audio_format_t format;
-            size_t size;
-            status_t status = decode(fd, offset, length, &sampleRate, &numChannels, &format,
-                                     heap, &size);
-            ::close(fd);
-            reply->writeInt32(status);
-            if (status == NO_ERROR) {
-                reply->writeInt32(sampleRate);
-                reply->writeInt32(numChannels);
-                reply->writeInt32((int32_t)format);
-                reply->writeInt32((int32_t)size);
-            }
-            return NO_ERROR;
-        } break;
         case CREATE_MEDIA_RECORDER: {
             CHECK_INTERFACE(IMediaPlayerService, data, reply);
             sp<IMediaRecorder> recorder = createMediaRecorder();
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
deleted file mode 100644
index d2e381b..0000000
--- a/media/libmedia/SoundPool.cpp
+++ /dev/null
@@ -1,921 +0,0 @@
-/*
- * 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/SoundPool.h>
-#include "SoundPoolThread.h"
-#include <media/AudioPolicyHelper.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(const char* path, int priority __unused)
-{
-    ALOGV("load: path=%s, priority=%d", path, priority);
-    Mutex::Autolock lock(&mLock);
-    sp<Sample> sample = new Sample(++mNextSampleID, path);
-    mSamples.add(sample->sampleID(), sample);
-    doLoad(sample);
-    return sample->sampleID();
-}
-
-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, const char* url)
-{
-    init();
-    mSampleID = sampleID;
-    mUrl = strdup(url);
-    ALOGV("create sampleID=%d, url=%s", mSampleID, mUrl);
-}
-
-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;
-    mUrl = 0;
-}
-
-Sample::~Sample()
-{
-    ALOGV("Sample::destructor sampleID=%d, fd=%d", mSampleID, mFd);
-    if (mFd > 0) {
-        ALOGV("close(%d)", mFd);
-        ::close(mFd);
-    }
-    free(mUrl);
-}
-
-status_t Sample::doLoad()
-{
-    uint32_t sampleRate;
-    int numChannels;
-    audio_format_t format;
-    status_t status;
-    mHeap = new MemoryHeapBase(kDefaultHeapSize);
-
-    ALOGV("Start decode");
-    if (mUrl) {
-        status = MediaPlayer::decode(
-                NULL /* httpService */,
-                mUrl,
-                &sampleRate,
-                &numChannels,
-                &format,
-                mHeap,
-                &mSize);
-    } else {
-        status = MediaPlayer::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: %s", mUrl);
-        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);
-        uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate;
-        uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount;
-        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
-        // 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
-        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 ||
-            event == AudioTrack::EVENT_NEW_IAUDIOTRACK) {
-        ALOGV("process %p channel %d event %s",
-              this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" :
-                      (event == AudioTrack::EVENT_BUFFER_END) ? "BUFFER_END" : "NEW_IAUDIOTRACK");
-        mSoundPool->addToStopList(this);
-    } else if (event == AudioTrack::EVENT_LOOP_END) {
-        ALOGV("End loop %p channel %d", 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/libmedia/SoundPoolThread.cpp b/media/libmedia/SoundPoolThread.cpp
deleted file mode 100644
index ba3b482..0000000
--- a/media/libmedia/SoundPoolThread.cpp
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * 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/libmedia/SoundPoolThread.h b/media/libmedia/SoundPoolThread.h
deleted file mode 100644
index 7e96900..0000000
--- a/media/libmedia/SoundPoolThread.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * 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 <media/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/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 9611ac7..05c89ed 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -835,53 +835,12 @@ void MediaPlayer::notify(int msg, int ext1, int ext2, const Parcel *obj)
     }
 }
 
-/*static*/ status_t MediaPlayer::decode(
-        const sp<IMediaHTTPService> &httpService,
-        const char* url,
-        uint32_t *pSampleRate,
-        int* pNumChannels,
-        audio_format_t* pFormat,
-        const sp<IMemoryHeap>& heap,
-        size_t *pSize)
-{
-    ALOGV("decode(%s)", url);
-    status_t status;
-    const sp<IMediaPlayerService>& service = getMediaPlayerService();
-    if (service != 0) {
-        status = service->decode(httpService, url, pSampleRate, pNumChannels, pFormat, heap, pSize);
-    } else {
-        ALOGE("Unable to locate media service");
-        status = DEAD_OBJECT;
-    }
-    return status;
-
-}
-
 void MediaPlayer::died()
 {
     ALOGV("died");
     notify(MEDIA_ERROR, MEDIA_ERROR_SERVER_DIED, 0);
 }
 
-/*static*/ status_t MediaPlayer::decode(int fd, int64_t offset, int64_t length,
-                                        uint32_t *pSampleRate, int* pNumChannels,
-                                        audio_format_t* pFormat,
-                                        const sp<IMemoryHeap>& heap, size_t *pSize)
-{
-    ALOGV("decode(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
-    status_t status;
-    const sp<IMediaPlayerService>& service = getMediaPlayerService();
-    if (service != 0) {
-        status = service->decode(fd, offset, length, pSampleRate,
-                                 pNumChannels, pFormat, heap, pSize);
-    } else {
-        ALOGE("Unable to locate media service");
-        status = DEAD_OBJECT;
-    }
-    return status;
-
-}
-
 status_t MediaPlayer::setNextMediaPlayer(const sp<MediaPlayer>& next) {
     if (mPlayer == NULL) {
         return NO_INIT;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 4d88a7f..1936218 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -384,28 +384,6 @@ sp<IRemoteDisplay> MediaPlayerService::listenForRemoteDisplay(
     return new RemoteDisplay(client, iface.string());
 }
 
-status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& /*args*/) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    result.append(" AudioCache\n");
-    if (mHeap != 0) {
-        snprintf(buffer, 255, "  heap base(%p), size(%zu), flags(%d)\n",
-                mHeap->getBase(), mHeap->getSize(), mHeap->getFlags());
-        result.append(buffer);
-    }
-    snprintf(buffer, 255, "  msec per frame(%f), channel count(%d), format(%d), frame count(%zd)\n",
-            mMsecsPerFrame, mChannelCount, mFormat, mFrameCount);
-    result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%d), size(%d), error(%d), command complete(%s)\n",
-            mSampleRate, mSize, mError, mCommandComplete?"true":"false");
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
 status_t MediaPlayerService::AudioOutput::dump(int fd, const Vector<String16>& args) const
 {
     const size_t SIZE = 256;
@@ -1279,128 +1257,6 @@ int Antagonizer::callbackThread(void* user)
 }
 #endif
 
-status_t MediaPlayerService::decode(
-        const sp<IMediaHTTPService> &httpService,
-        const char* url,
-        uint32_t *pSampleRate,
-        int* pNumChannels,
-        audio_format_t* pFormat,
-        const sp<IMemoryHeap>& heap,
-        size_t *pSize)
-{
-    ALOGV("decode(%s)", url);
-    sp<MediaPlayerBase> player;
-    status_t status = BAD_VALUE;
-
-    // Protect our precious, precious DRMd ringtones by only allowing
-    // decoding of http, but not filesystem paths or content Uris.
-    // If the application wants to decode those, it should open a
-    // filedescriptor for them and use that.
-    if (url != NULL && strncmp(url, "http://", 7) != 0) {
-        ALOGD("Can't decode %s by path, use filedescriptor instead", url);
-        return BAD_VALUE;
-    }
-
-    player_type playerType =
-        MediaPlayerFactory::getPlayerType(NULL /* client */, url);
-    ALOGV("player type = %d", playerType);
-
-    // create the right type of player
-    sp<AudioCache> cache = new AudioCache(heap);
-    player = MediaPlayerFactory::createPlayer(playerType, cache.get(), cache->notify);
-    if (player == NULL) goto Exit;
-    if (player->hardwareOutput()) goto Exit;
-
-    static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
-
-    // set data source
-    if (player->setDataSource(httpService, url) != NO_ERROR) goto Exit;
-
-    ALOGV("prepare");
-    player->prepareAsync();
-
-    ALOGV("wait for prepare");
-    if (cache->wait() != NO_ERROR) goto Exit;
-
-    ALOGV("start");
-    player->start();
-
-    ALOGV("wait for playback complete");
-    cache->wait();
-    // in case of error, return what was successfully decoded.
-    if (cache->size() == 0) {
-        goto Exit;
-    }
-
-    *pSize = cache->size();
-    *pSampleRate = cache->sampleRate();
-    *pNumChannels = cache->channelCount();
-    *pFormat = cache->format();
-    ALOGV("return size %d sampleRate=%u, channelCount = %d, format = %d",
-          *pSize, *pSampleRate, *pNumChannels, *pFormat);
-    status = NO_ERROR;
-
-Exit:
-    if (player != 0) player->reset();
-    return status;
-}
-
-status_t MediaPlayerService::decode(int fd, int64_t offset, int64_t length,
-                                       uint32_t *pSampleRate, int* pNumChannels,
-                                       audio_format_t* pFormat,
-                                       const sp<IMemoryHeap>& heap, size_t *pSize)
-{
-    ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
-    sp<MediaPlayerBase> player;
-    status_t status = BAD_VALUE;
-
-    player_type playerType = MediaPlayerFactory::getPlayerType(NULL /* client */,
-                                                               fd,
-                                                               offset,
-                                                               length);
-    ALOGV("player type = %d", playerType);
-
-    // create the right type of player
-    sp<AudioCache> cache = new AudioCache(heap);
-    player = MediaPlayerFactory::createPlayer(playerType, cache.get(), cache->notify);
-    if (player == NULL) goto Exit;
-    if (player->hardwareOutput()) goto Exit;
-
-    static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
-
-    // set data source
-    if (player->setDataSource(fd, offset, length) != NO_ERROR) goto Exit;
-
-    ALOGV("prepare");
-    player->prepareAsync();
-
-    ALOGV("wait for prepare");
-    if (cache->wait() != NO_ERROR) goto Exit;
-
-    ALOGV("start");
-    player->start();
-
-    ALOGV("wait for playback complete");
-    cache->wait();
-    // in case of error, return what was successfully decoded.
-    if (cache->size() == 0) {
-        goto Exit;
-    }
-
-    *pSize = cache->size();
-    *pSampleRate = cache->sampleRate();
-    *pNumChannels = cache->channelCount();
-    *pFormat = cache->format();
-    ALOGV("return size %d, sampleRate=%u, channelCount = %d, format = %d",
-          *pSize, *pSampleRate, *pNumChannels, *pFormat);
-    status = NO_ERROR;
-
-Exit:
-    if (player != 0) player->reset();
-    return status;
-}
-
-
 #undef LOG_TAG
 #define LOG_TAG "AudioSink"
 MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid,
@@ -1950,47 +1806,6 @@ uint32_t MediaPlayerService::AudioOutput::getSampleRate() const
     return mTrack->getSampleRate();
 }
 
-#undef LOG_TAG
-#define LOG_TAG "AudioCache"
-MediaPlayerService::AudioCache::AudioCache(const sp<IMemoryHeap>& heap) :
-    mHeap(heap), mChannelCount(0), mFrameCount(1024), mSampleRate(0), mSize(0),
-    mFrameSize(1), mError(NO_ERROR),  mCommandComplete(false)
-{
-}
-
-uint32_t MediaPlayerService::AudioCache::latency () const
-{
-    return 0;
-}
-
-float MediaPlayerService::AudioCache::msecsPerFrame() const
-{
-    return mMsecsPerFrame;
-}
-
-status_t MediaPlayerService::AudioCache::getPosition(uint32_t *position) const
-{
-    if (position == 0) return BAD_VALUE;
-    *position = mSize / mFrameSize;
-    return NO_ERROR;
-}
-
-status_t MediaPlayerService::AudioCache::getTimestamp(AudioTimestamp &ts) const
-{
-    ts.mPosition = mSize / mFrameSize;
-    nsecs_t now = systemTime(SYSTEM_TIME_MONOTONIC);
-    ts.mTime.tv_sec = now / 1000000000LL;
-    ts.mTime.tv_nsec = now - (1000000000LL * ts.mTime.tv_sec);
-    return NO_ERROR;
-}
-
-status_t MediaPlayerService::AudioCache::getFramesWritten(uint32_t *written) const
-{
-    if (written == 0) return BAD_VALUE;
-    *written = mSize / mFrameSize;
-    return NO_ERROR;
-}
-
 ////////////////////////////////////////////////////////////////////////////////
 
 struct CallbackThread : public Thread {
@@ -2058,136 +1873,6 @@ bool CallbackThread::threadLoop() {
 
 ////////////////////////////////////////////////////////////////////////////////
 
-status_t MediaPlayerService::AudioCache::open(
-        uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
-        audio_format_t format, int bufferCount,
-        AudioCallback cb, void *cookie, audio_output_flags_t /*flags*/,
-        const audio_offload_info_t* /*offloadInfo*/)
-{
-    ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
-    if (mHeap->getHeapID() < 0) {
-        return NO_INIT;
-    }
-
-    mSampleRate = sampleRate;
-    mChannelCount = (uint16_t)channelCount;
-    mFormat = format;
-    mMsecsPerFrame = 1.e3 / (float) sampleRate;
-    mFrameSize =  audio_is_linear_pcm(mFormat)
-            ? mChannelCount * audio_bytes_per_sample(mFormat) : 1;
-    mFrameCount = mHeap->getSize() / mFrameSize;
-
-    if (cb != NULL) {
-        mCallbackThread = new CallbackThread(this, cb, cookie);
-    }
-    return NO_ERROR;
-}
-
-status_t MediaPlayerService::AudioCache::start() {
-    if (mCallbackThread != NULL) {
-        mCallbackThread->run("AudioCache callback");
-    }
-    return NO_ERROR;
-}
-
-void MediaPlayerService::AudioCache::stop() {
-    if (mCallbackThread != NULL) {
-        mCallbackThread->requestExitAndWait();
-    }
-}
-
-ssize_t MediaPlayerService::AudioCache::write(const void* buffer, size_t size)
-{
-    ALOGV("write(%p, %u)", buffer, size);
-    if ((buffer == 0) || (size == 0)) return size;
-
-    uint8_t* p = static_cast<uint8_t*>(mHeap->getBase());
-    if (p == NULL) return NO_INIT;
-    p += mSize;
-    ALOGV("memcpy(%p, %p, %u)", p, buffer, size);
-
-    bool overflow = mSize + size > mHeap->getSize();
-    if (overflow) {
-        ALOGE("Heap size overflow! req size: %d, max size: %d", (mSize + size), mHeap->getSize());
-        size = mHeap->getSize() - mSize;
-    }
-    size -= size % mFrameSize; // consume only integral amounts of frame size
-    memcpy(p, buffer, size);
-    mSize += size;
-
-    if (overflow) {
-        // Signal heap filled here (last frame may be truncated).
-        // After this point, no more data should be written as the
-        // heap is filled and the AudioCache should be effectively
-        // immutable with respect to future writes.
-        //
-        // It is thus safe for another thread to read the AudioCache.
-        mCommandComplete = true;
-        mSignal.signal();
-    }
-    return size;
-}
-
-// call with lock held
-status_t MediaPlayerService::AudioCache::wait()
-{
-    Mutex::Autolock lock(mLock);
-    while (!mCommandComplete) {
-        mSignal.wait(mLock);
-    }
-    mCommandComplete = false;
-
-    if (mError == NO_ERROR) {
-        ALOGV("wait - success");
-    } else {
-        ALOGV("wait - error");
-    }
-    return mError;
-}
-
-void MediaPlayerService::AudioCache::notify(
-        void* cookie, int msg, int ext1, int ext2, const Parcel* /*obj*/)
-{
-    ALOGV("notify(%p, %d, %d, %d)", cookie, msg, ext1, ext2);
-    AudioCache* p = static_cast<AudioCache*>(cookie);
-
-    // ignore buffering messages
-    switch (msg)
-    {
-    case MEDIA_ERROR:
-        ALOGE("Error %d, %d occurred", ext1, ext2);
-        p->mError = ext1;
-        break;
-    case MEDIA_PREPARED:
-        ALOGV("prepared");
-        break;
-    case MEDIA_PLAYBACK_COMPLETE:
-        ALOGV("playback complete");
-        break;
-    default:
-        ALOGV("ignored");
-        return;
-    }
-
-    // wake up thread
-    Mutex::Autolock lock(p->mLock);
-    p->mCommandComplete = true;
-    p->mSignal.signal();
-}
-
-int MediaPlayerService::AudioCache::getSessionId() const
-{
-    return 0;
-}
-
-uint32_t MediaPlayerService::AudioCache::getSampleRate() const
-{
-    if (mMsecsPerFrame == 0) {
-        return 0;
-    }
-    return (uint32_t)(1.e3 / mMsecsPerFrame);
-}
-
 void MediaPlayerService::addBatteryData(uint32_t params)
 {
     Mutex::Autolock lock(mLock);
@@ -2231,7 +1916,7 @@ void MediaPlayerService::addBatteryData(uint32_t params)
         return;
     }
 
-    // an sudio stream is started
+    // an audio stream is started
     if (params & kBatteryDataAudioFlingerStart) {
         // record the start time only if currently no other audio
         // is being played
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 3b96e88..fad3447 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -77,7 +77,6 @@ class MediaPlayerService : public BnMediaPlayerService
         virtual                 ~AudioOutput();
 
         virtual bool            ready() const { return mTrack != 0; }
-        virtual bool            realtime() const { return true; }
         virtual ssize_t         bufferSize() const;
         virtual ssize_t         frameCount() const;
         virtual ssize_t         channelCount() const;
@@ -184,75 +183,6 @@ class MediaPlayerService : public BnMediaPlayerService
     }; // AudioOutput
 
 
-    class AudioCache : public MediaPlayerBase::AudioSink
-    {
-    public:
-                                AudioCache(const sp<IMemoryHeap>& heap);
-        virtual                 ~AudioCache() {}
-
-        virtual bool            ready() const { return (mChannelCount > 0) && (mHeap->getHeapID() > 0); }
-        virtual bool            realtime() const { return false; }
-        virtual ssize_t         bufferSize() const { return frameSize() * mFrameCount; }
-        virtual ssize_t         frameCount() const { return mFrameCount; }
-        virtual ssize_t         channelCount() const { return (ssize_t)mChannelCount; }
-        virtual ssize_t         frameSize() const { return (ssize_t)mFrameSize; }
-        virtual uint32_t        latency() const;
-        virtual float           msecsPerFrame() const;
-        virtual status_t        getPosition(uint32_t *position) const;
-        virtual status_t        getTimestamp(AudioTimestamp &ts) const;
-        virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
-        virtual int             getSessionId() const;
-        virtual uint32_t        getSampleRate() const;
-
-        virtual status_t        open(
-                uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
-                audio_format_t format, int bufferCount = 1,
-                AudioCallback cb = NULL, void *cookie = NULL,
-                audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
-                const audio_offload_info_t *offloadInfo = NULL);
-
-        virtual status_t        start();
-        virtual ssize_t         write(const void* buffer, size_t size);
-        virtual void            stop();
-        virtual void            flush() {}
-        virtual void            pause() {}
-        virtual void            close() {}
-                void            setAudioStreamType(audio_stream_type_t streamType __unused) {}
-                // stream type is not used for AudioCache
-        virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
-
-                void            setVolume(float left __unused, float right __unused) {}
-        virtual status_t        setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; }
-                uint32_t        sampleRate() const { return mSampleRate; }
-                audio_format_t  format() const { return mFormat; }
-                size_t          size() const { return mSize; }
-                status_t        wait();
-
-                sp<IMemoryHeap> getHeap() const { return mHeap; }
-
-        static  void            notify(void* cookie, int msg,
-                                       int ext1, int ext2, const Parcel *obj);
-        virtual status_t        dump(int fd, const Vector<String16>& args) const;
-
-    private:
-                                AudioCache();
-
-        Mutex               mLock;
-        Condition           mSignal;
-        sp<IMemoryHeap>     mHeap;
-        float               mMsecsPerFrame;
-        uint16_t            mChannelCount;
-        audio_format_t      mFormat;
-        ssize_t             mFrameCount;
-        uint32_t            mSampleRate;
-        uint32_t            mSize;
-        size_t              mFrameSize;
-        int                 mError;
-        bool                mCommandComplete;
-
-        sp<Thread>          mCallbackThread;
-    }; // AudioCache
-
 public:
     static  void                instantiate();
 
@@ -263,19 +193,6 @@ public:
 
     virtual sp<IMediaPlayer>    create(const sp<IMediaPlayerClient>& client, int audioSessionId);
 
-    virtual status_t            decode(
-            const sp<IMediaHTTPService> &httpService,
-            const char* url,
-            uint32_t *pSampleRate,
-            int* pNumChannels,
-            audio_format_t* pFormat,
-            const sp<IMemoryHeap>& heap,
-            size_t *pSize);
-
-    virtual status_t            decode(int fd, int64_t offset, int64_t length,
-                                       uint32_t *pSampleRate, int* pNumChannels,
-                                       audio_format_t* pFormat,
-                                       const sp<IMemoryHeap>& heap, size_t *pSize);
     virtual sp<IMediaCodecList> getCodecList() const;
     virtual sp<IOMX>            getOMX();
     virtual sp<ICrypto>         makeCrypto();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 4bff90f..c52d837 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -637,8 +637,17 @@ void NuPlayerDriver::notifyListener_l(
         case MEDIA_PLAYBACK_COMPLETE:
         {
             if (mState != STATE_RESET_IN_PROGRESS) {
-                if (mLooping || (mAutoLoop
-                        && (mAudioSink == NULL || mAudioSink->realtime()))) {
+                if (mAutoLoop) {
+                    audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
+                    if (mAudioSink != NULL) {
+                        streamType = mAudioSink->getAudioStreamType();
+                    }
+                    if (streamType == AUDIO_STREAM_NOTIFICATION) {
+                        ALOGW("disabling auto-loop for notification");
+                        mAutoLoop = false;
+                    }
+                }
+                if (mLooping || mAutoLoop) {
                     mPlayer->seekToAsync(0);
                     break;
                 }
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index ab8ac79..a31b317 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -879,10 +879,7 @@ void AwesomePlayer::onStreamDone() {
     }
 
     if ((mFlags & LOOPING)
-            || ((mFlags & AUTO_LOOPING)
-                && (mAudioSink == NULL || mAudioSink->realtime()))) {
-        // Don't AUTO_LOOP if we're being recorded, since that cannot be
-        // turned off and recording would go on indefinitely.
+            || (mFlags & AUTO_LOOPING)) {
 
         seekTo_l(0);