diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
new file mode 100644
index 0000000..fb21629
--- /dev/null
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -0,0 +1,1450 @@
+/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+**
+** Copyright 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_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <signal.h>
+#include <sys/time.h>
+#include <sys/resource.h>
+
+#include <utils/IServiceManager.h>
+#include <utils/Log.h>
+#include <utils/Parcel.h>
+#include <utils/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+
+#include <media/AudioTrack.h>
+#include <media/AudioRecord.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <hardware/AudioHardwareInterface.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+
+namespace android {
+
+static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const unsigned long kBufferRecoveryInUsecs = 2000;
+static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
+static const float MAX_GAIN = 4096.0f;
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+
+#define AUDIOFLINGER_SECURITY_ENABLED 1
+
+// ----------------------------------------------------------------------------
+
+static bool recordingAllowed() {
+#ifndef HAVE_ANDROID_OS
+    return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
+    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
+    return ok;
+#else
+    if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
+        LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
+    return true;
+#endif
+}
+
+static bool settingsAllowed() {
+#ifndef HAVE_ANDROID_OS
+    return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
+    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+    return ok;
+#else
+    if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
+        LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
+    return true;
+#endif
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioFlinger()
+    : BnAudioFlinger(), Thread(false),
+        mMasterVolume(0), mMasterMute(true),
+        mAudioMixer(0), mAudioHardware(0), mOutput(0), mAudioRecordThread(0),
+        mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0),
+        mMixBuffer(0), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0),
+        mStandby(false), mInWrite(false)
+{
+    mHardwareStatus = AUDIO_HW_IDLE;
+    mAudioHardware = AudioHardwareInterface::create();
+    mHardwareStatus = AUDIO_HW_INIT;
+    if (mAudioHardware->initCheck() == NO_ERROR) {
+        // open 16-bit output stream for s/w mixer
+        mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+        mOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT);
+        mHardwareStatus = AUDIO_HW_IDLE;
+        if (mOutput) {
+            mSampleRate = mOutput->sampleRate();
+            mChannelCount = mOutput->channelCount();
+            mFormat = mOutput->format();
+            mMixBufferSize = mOutput->bufferSize();
+            mFrameCount = mMixBufferSize / mChannelCount / sizeof(int16_t);
+            mMixBuffer = new int16_t[mFrameCount * mChannelCount];
+            memset(mMixBuffer, 0, mMixBufferSize);
+            mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+            // FIXME - this should come from settings
+            setMasterVolume(1.0f);
+            setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+            setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+            setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
+            setMode(AudioSystem::MODE_NORMAL);
+            mMasterMute = false;
+        } else {
+            LOGE("Failed to initialize output stream");
+        }
+    } else {
+        LOGE("Couldn't even initialize the stubbed audio hardware!");
+    }
+}
+
+AudioFlinger::~AudioFlinger()
+{
+    delete mOutput;
+    delete mAudioHardware;
+    delete [] mMixBuffer;
+    delete mAudioMixer;
+    mAudioRecordThread.clear();
+}
+
+status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    result.append("Clients:\n");
+    for (size_t i = 0; i < mClients.size(); ++i) {
+        wp<Client> wClient = mClients.valueAt(i);
+        if (wClient != 0) {
+            sp<Client> client = wClient.promote();
+            if (client != 0) {
+                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
+                result.append(buffer);
+            }
+        }
+    }
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    result.append("Tracks:\n");
+    result.append("   Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        wp<Track> wTrack = mTracks[i];
+        if (wTrack != 0) {
+            sp<Track> track = wTrack.promote();
+            if (track != 0) {
+                track->dump(buffer, SIZE);
+                result.append(buffer);
+            }
+        }
+    }
+
+    result.append("Active Tracks:\n");
+    result.append("   Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+        wp<Track> wTrack = mTracks[i];
+        if (wTrack != 0) {
+            sp<Track> track = wTrack.promote();
+            if (track != 0) {
+                track->dump(buffer, SIZE);
+                result.append(buffer);
+            }
+        }
+    }
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    
+    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", audioMixer().trackNames());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+    result.append(buffer);
+    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "Permission Denial: "
+            "can't dump AudioFlinger from pid=%d, uid=%d\n",
+            IPCThreadState::self()->getCallingPid(),
+            IPCThreadState::self()->getCallingUid());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
+{
+    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+        dumpPermissionDenial(fd, args);
+    } else {
+        AutoMutex lock(&mLock);
+
+        dumpClients(fd, args);
+        dumpTracks(fd, args);
+        dumpInternals(fd, args);
+        if (mAudioHardware) {
+            mAudioHardware->dumpState(fd, args);
+        }
+    }
+    return NO_ERROR;
+}
+
+// Thread virtuals
+bool AudioFlinger::threadLoop()
+{
+    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
+    unsigned long sleepTime = kBufferRecoveryInUsecs;
+    const size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
+    int16_t* curBuf = mMixBuffer;
+    Vector< sp<Track> > tracksToRemove;
+    size_t enabledTracks;
+    nsecs_t standbyTime = systemTime();
+
+    do {
+        enabledTracks = 0;
+        { // scope for the lock
+            Mutex::Autolock _l(mLock);
+            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
+                // wait until we have something to do...
+                LOGV("Audio hardware entering standby\n");
+                mHardwareStatus = AUDIO_HW_STANDBY;
+                if (!mStandby) {
+                    mAudioHardware->standby();
+                    mStandby = true;
+                }
+                mHardwareStatus = AUDIO_HW_IDLE;
+                // we're about to wait, flush the binder command buffer
+                IPCThreadState::self()->flushCommands();
+                mWaitWorkCV.wait(mLock);
+                LOGV("Audio hardware exiting standby\n");
+                standbyTime = systemTime() + kStandbyTimeInNsecs;
+                continue;
+            }
+
+            // find out which tracks need to be processed
+            size_t count = activeTracks.size();
+            for (size_t i=0 ; i<count ; i++) {
+                sp<Track> t = activeTracks[i].promote();
+                if (t == 0) continue;
+
+                Track* const track = t.get();
+                audio_track_cblk_t* cblk = track->cblk();
+                uint32_t u = cblk->user;
+                uint32_t s = cblk->server;
+
+                // The first time a track is added we wait
+                // for all its buffers to be filled before processing it
+                audioMixer().setActiveTrack(track->name());
+                if ((u > s) && (track->isReady(u, s) || track->isStopped()) &&
+                        !track->isPaused())
+                {
+                    //LOGD("u=%08x, s=%08x [OK]", u, s);
+
+                    // compute volume for this track
+                    int16_t left, right;
+                    if (track->isMuted() || mMasterMute || track->isPausing()) {
+                        left = right = 0;
+                        if (track->isPausing()) {
+                            LOGV("paused(%d)", track->name());
+                            track->setPaused();
+                        }
+                    } else {
+                        float typeVolume = mStreamTypes[track->type()].volume;
+                        float v = mMasterVolume * typeVolume;
+                        float v_clamped = v * cblk->volume[0];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        left = int16_t(v_clamped);
+                        v_clamped = v * cblk->volume[1];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        right = int16_t(v_clamped);
+                    }
+
+                    // XXX: these things DON'T need to be done each time
+                    AudioMixer& mixer(audioMixer());
+                    mixer.setBufferProvider(track);
+                    mixer.enable(AudioMixer::MIXING);
+
+                    int param;
+                    if ( track->mFillingUpStatus == Track::FS_FILLED) {
+                        // no ramp for the first volume setting
+                        track->mFillingUpStatus = Track::FS_ACTIVE;
+                        if (track->mState == TrackBase::RESUMING) {
+                            track->mState = TrackBase::ACTIVE;
+                            param = AudioMixer::RAMP_VOLUME;
+                        } else {
+                            param = AudioMixer::VOLUME;
+                        }
+                    } else {
+                        param = AudioMixer::RAMP_VOLUME;
+                    }
+                    mixer.setParameter(param, AudioMixer::VOLUME0, left);
+                    mixer.setParameter(param, AudioMixer::VOLUME1, right);
+                    mixer.setParameter(
+                        AudioMixer::TRACK,
+                        AudioMixer::FORMAT, track->format());
+                    mixer.setParameter(
+                        AudioMixer::TRACK,
+                        AudioMixer::CHANNEL_COUNT, track->channelCount());
+                    mixer.setParameter(
+                        AudioMixer::RESAMPLE,
+                        AudioMixer::SAMPLE_RATE,
+                        int(cblk->sampleRate));
+
+                    // reset retry count
+                    track->mRetryCount = kMaxTrackRetries;
+                    enabledTracks++;
+                } else {
+                    //LOGD("u=%08x, s=%08x [NOT READY]", u, s);
+                    if (track->isStopped()) {
+                        track->mFillingUpStatus = Track::FS_FILLING;
+                        track->mFlags = 0;    
+                    }
+                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+                        // We have consumed all the buffers of this track.
+                        // Remove it from the list of active tracks.
+                        LOGV("remove(%d) from active list", track->name());
+                        tracksToRemove.add(track);
+                    } else {
+                        // No buffers for this track. Give it a few chances to
+                        // fill a buffer, then remove it from active list.
+                        if (--(track->mRetryCount) <= 0) {
+                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+                            tracksToRemove.add(track);
+                        }
+                    }
+                    // LOGV("disable(%d)", track->name());
+                    audioMixer().disable(AudioMixer::MIXING);
+                }
+            }
+
+            // remove all the tracks that need to be...
+            count = tracksToRemove.size();
+            if (UNLIKELY(count)) {
+                for (size_t i=0 ; i<count ; i++) {
+                    const sp<Track>& track = tracksToRemove[i];
+                    mActiveTracks.remove(track);
+                    if (track->isTerminated()) {
+                        mTracks.remove(track);
+                        audioMixer().deleteTrackName(track->mName);
+                    }
+                }
+            }
+        }
+
+        if (LIKELY(enabledTracks)) {
+            // mix buffers...
+            audioMixer().process(curBuf);
+
+            // output audio to hardware
+            mLastWriteTime = systemTime();
+            mInWrite = true;
+            mOutput->write(curBuf, mixBufferSize);
+            mNumWrites++;
+            mInWrite = false;
+            mStandby = false;
+            nsecs_t temp = systemTime();
+            standbyTime = temp + kStandbyTimeInNsecs;
+            nsecs_t delta = temp - mLastWriteTime;
+            if (delta > maxPeriod) {
+                LOGW("write blocked for %llu msecs", ns2ms(delta));
+                mNumDelayedWrites++;
+            }
+            sleepTime = kBufferRecoveryInUsecs;
+        } else {
+            // There was nothing to mix this round, which means all
+            // active tracks were late. Sleep a little bit to give
+            // them another chance. If we're too late, the audio
+            // hardware will zero-fill for us.
+            LOGV("no buffers - usleep(%lu)", sleepTime);
+            usleep(sleepTime);
+            if (sleepTime < kMaxBufferRecoveryInUsecs) {
+                sleepTime += kBufferRecoveryInUsecs;
+            }
+        }
+
+        // finally let go of all our tracks, without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.
+        tracksToRemove.clear();
+    } while (true);
+
+    return false;
+}
+
+status_t AudioFlinger::readyToRun()
+{
+    if (mSampleRate == 0) {
+        LOGE("No working audio driver found.");
+        return NO_INIT;
+    }
+    LOGI("AudioFlinger's main thread ready to run.");
+    return NO_ERROR;
+}
+
+void AudioFlinger::onFirstRef()
+{
+    run("AudioFlinger", ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// IAudioFlinger interface
+sp<IAudioTrack> AudioFlinger::createTrack(
+        pid_t pid,
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int bufferCount,
+        uint32_t flags)
+{
+    if (streamType >= AudioTrack::NUM_STREAM_TYPES) {
+        LOGE("invalid stream type");
+        return NULL;
+    }
+
+    if (sampleRate > MAX_SAMPLE_RATE) {
+        LOGE("Sample rate out of range: %d", sampleRate);
+        return NULL;
+    }
+
+    sp<Track> track;
+    sp<TrackHandle> trackHandle;
+    Mutex::Autolock _l(mLock);
+
+    if (mSampleRate == 0) {
+        LOGE("Audio driver not initialized.");
+        return trackHandle;
+    }
+
+    sp<Client> client;
+    wp<Client> wclient = mClients.valueFor(pid);
+
+    if (wclient != NULL) {
+        client = wclient.promote();
+    } else {
+        client = new Client(this, pid);
+        mClients.add(pid, client);
+    }
+
+    // FIXME: Buffer size should be based on sample rate for consistent latency
+    track = new Track(this, client, streamType, sampleRate, format,
+            channelCount, bufferCount, channelCount == 1 ? mMixBufferSize>>1 : mMixBufferSize);
+    mTracks.add(track);
+    trackHandle = new TrackHandle(track);
+    return trackHandle;
+}
+
+uint32_t AudioFlinger::sampleRate() const
+{
+    return mSampleRate;
+}
+
+int AudioFlinger::channelCount() const
+{
+    return mChannelCount;
+}
+
+int AudioFlinger::format() const
+{
+    return mFormat;
+}
+
+size_t AudioFlinger::frameCount() const
+{
+    return mFrameCount;
+}
+
+status_t AudioFlinger::setMasterVolume(float value)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    // when hw supports master volume, don't scale in sw mixer
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+    if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
+        mMasterVolume = 1.0f;
+    }
+    else {
+        mMasterVolume = value;
+    }
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
+        LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
+        return BAD_VALUE;
+    }
+
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_GET_ROUTING;
+    uint32_t r;
+    uint32_t err = mAudioHardware->getRouting(mode, &r);
+    if (err == NO_ERROR) {
+        r = (r & ~mask) | (routes & mask);
+        mHardwareStatus = AUDIO_HW_SET_ROUTING;
+        err = mAudioHardware->setRouting(mode, r);
+    }
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return err;
+}
+
+uint32_t AudioFlinger::getRouting(int mode) const
+{
+    uint32_t routes = 0;
+    if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
+        mHardwareStatus = AUDIO_HW_GET_ROUTING;
+        mAudioHardware->getRouting(mode, &routes);
+        mHardwareStatus = AUDIO_HW_IDLE;
+    } else {
+        LOGW("Illegal value: getRouting(%d)", mode);
+    }
+    return routes;
+}
+
+status_t AudioFlinger::setMode(int mode)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+        LOGW("Illegal value: setMode(%d)", mode);
+        return BAD_VALUE;
+    }
+
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_SET_MODE;
+    status_t ret = mAudioHardware->setMode(mode);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return ret;
+}
+
+int AudioFlinger::getMode() const
+{
+    int mode = AudioSystem::MODE_INVALID;
+    mHardwareStatus = AUDIO_HW_SET_MODE;
+    mAudioHardware->getMode(&mode);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return mode;
+}
+
+status_t AudioFlinger::setMicMute(bool state)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
+    status_t ret = mAudioHardware->setMicMute(state);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return ret;
+}
+
+bool AudioFlinger::getMicMute() const
+{
+    bool state = AudioSystem::MODE_INVALID;
+    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
+    mAudioHardware->getMicMute(&state);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return state;
+}
+
+status_t AudioFlinger::setMasterMute(bool muted)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    mMasterMute = muted;
+    return NO_ERROR;
+}
+
+float AudioFlinger::masterVolume() const
+{
+    return mMasterVolume;
+}
+
+bool AudioFlinger::masterMute() const
+{
+    return mMasterMute;
+}
+
+status_t AudioFlinger::setStreamVolume(int stream, float value)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+    
+    mStreamTypes[stream].volume = value;
+    status_t ret = NO_ERROR;
+    if (stream == AudioTrack::VOICE_CALL) {
+        AutoMutex lock(mHardwareLock);
+        mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+        ret = mAudioHardware->setVoiceVolume(value);
+        mHardwareStatus = AUDIO_HW_IDLE;
+    }
+    return ret;
+}
+
+status_t AudioFlinger::setStreamMute(int stream, bool muted)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+    mStreamTypes[stream].mute = muted;
+    return NO_ERROR;
+}
+
+float AudioFlinger::streamVolume(int stream) const
+{
+    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+        return 0.0f;
+    }
+    return mStreamTypes[stream].volume;
+}
+
+bool AudioFlinger::streamMute(int stream) const
+{
+    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+        return true;
+    }
+    return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::isMusicActive() const
+{
+    size_t count = mActiveTracks.size();
+    for (size_t i = 0 ; i < count ; ++i) {
+        sp<Track> t = mActiveTracks[i].promote();
+        if (t == 0) continue;
+        Track* const track = t.get();
+        if (t->mStreamType == AudioTrack::MUSIC)
+            return true;
+    }
+    return false;
+}
+
+status_t AudioFlinger::setParameter(const char* key, const char* value)
+{
+    status_t result;
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_SET_PARAMETER;
+    result = mAudioHardware->setParameter(key, value);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return result;
+}
+
+void AudioFlinger::removeClient(pid_t pid)
+{
+    Mutex::Autolock _l(mLock);
+    mClients.removeItem(pid);
+}
+
+status_t AudioFlinger::addTrack(const sp<Track>& track)
+{
+    Mutex::Autolock _l(mLock);
+
+    // here the track could be either new, or restarted
+    // in both cases "unstop" the track
+    if (track->isPaused()) {
+        track->mState = TrackBase::RESUMING;
+        LOGV("PAUSED => RESUMING (%d)", track->name());
+    } else {
+        track->mState = TrackBase::ACTIVE;
+        LOGV("? => ACTIVE (%d)", track->name());
+    }
+    // set retry count for buffer fill
+    track->mRetryCount = kMaxTrackStartupRetries;
+    LOGV("mWaitWorkCV.broadcast");
+    mWaitWorkCV.broadcast();
+
+    if (mActiveTracks.indexOf(track) < 0) {
+        // the track is newly added, make sure it fills up all its
+        // buffers before playing. This is to ensure the client will
+        // effectively get the latency it requested.
+        track->mFillingUpStatus = Track::FS_FILLING;
+        mActiveTracks.add(track);
+        return NO_ERROR;
+    }
+    return ALREADY_EXISTS;
+}
+
+void AudioFlinger::removeTrack(wp<Track> track, int name)
+{
+    Mutex::Autolock _l(mLock);
+    sp<Track> t = track.promote();
+    if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
+        remove_track_l(track, name);
+    }
+}
+
+void AudioFlinger::remove_track_l(wp<Track> track, int name)
+{
+    sp<Track> t = track.promote();
+    if (t!=NULL) {
+        t->reset();
+    }
+    audioMixer().deleteTrackName(name);
+    mActiveTracks.remove(track);
+    mWaitWorkCV.broadcast();
+}
+
+void AudioFlinger::destroyTrack(const sp<Track>& track)
+{
+    // NOTE: We're acquiring a strong reference on the track before
+    // acquiring the lock, this is to make sure removing it from
+    // mTracks won't cause the destructor to be called while the lock is
+    // held (note that technically, 'track' could be a reference to an item
+    // in mTracks, which is why we need to do this).
+    sp<Track> keep(track);
+    Mutex::Autolock _l(mLock);
+    track->mState = TrackBase::TERMINATED;
+    if (mActiveTracks.indexOf(track) < 0) {
+        LOGV("remove track (%d) and delete from mixer", track->name());
+        mTracks.remove(track);
+        audioMixer().deleteTrackName(keep->name());
+    }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
+    :   RefBase(),
+        mAudioFlinger(audioFlinger),
+        mMemoryDealer(new MemoryDealer(1024*1024)),
+        mPid(pid)
+{
+    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
+}
+
+AudioFlinger::Client::~Client()
+{
+    mAudioFlinger->removeClient(mPid);
+}
+
+const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+{
+    return mMemoryDealer;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackBase::TrackBase(
+            const sp<AudioFlinger>& audioFlinger,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int bufferCount,
+            int bufferSize)
+    :   RefBase(),
+        mAudioFlinger(audioFlinger),
+        mClient(client),
+        mStreamType(streamType),
+        mFormat(format),
+        mChannelCount(channelCount),
+        mBufferCount(bufferCount),
+        mFlags(0),
+        mBufferSize(bufferSize),
+        mState(IDLE),
+        mClientTid(-1)
+{
+    mName = audioFlinger->audioMixer().getTrackName();
+    if (mName < 0) {
+        LOGE("no more track names availlable");
+        return;
+    }
+
+    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
+    size_t size = sizeof(audio_track_cblk_t) + bufferCount * bufferSize;
+    mCblkMemory = client->heap()->allocate(size);
+    if (mCblkMemory != 0) {
+        mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
+        if (mCblk) { // construct the shared structure in-place.
+            new(mCblk) audio_track_cblk_t();
+            // clear all buffers
+            mCblk->size = bufferSize;
+            mCblk->sampleRate = sampleRate;
+            mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+            memset(mBuffers, 0, bufferCount * bufferSize);
+        }
+    } else {
+        LOGE("not enough memory for AudioTrack size=%u", size);
+        client->heap()->dump("AudioTrack");
+        return;
+    }
+}
+
+AudioFlinger::TrackBase::~TrackBase()
+{
+    mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
+    mCblkMemory.clear();            // and free the shared memory
+    mClient.clear();
+}
+
+void AudioFlinger::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    buffer->raw = 0;
+    buffer->frameCount = 0;
+    step();
+}
+
+bool AudioFlinger::TrackBase::step() {
+    bool result;
+    audio_track_cblk_t* cblk = this->cblk();
+    
+    result = cblk->stepServer(bufferCount()); 
+    if (!result) {
+        LOGV("stepServer failed acquiring cblk mutex");
+        mFlags |= STEPSERVER_FAILED;
+    }
+    return result;
+}
+
+void AudioFlinger::TrackBase::reset() {
+    audio_track_cblk_t* cblk = this->cblk();
+
+    cblk->user = 0;
+    cblk->server = 0;
+    mFlags = 0;    
+}
+
+sp<IMemory> AudioFlinger::TrackBase::getCblk() const
+{
+    return mCblkMemory;
+}
+
+int AudioFlinger::TrackBase::sampleRate() const {
+    return mCblk->sampleRate;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::Track::Track(
+            const sp<AudioFlinger>& audioFlinger,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int bufferCount,
+            int bufferSize)
+    :   TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, bufferCount, bufferSize)
+{
+    mVolume[0] = 1.0f;
+    mVolume[1] = 1.0f;
+    mMute = false;
+}
+
+AudioFlinger::Track::~Track()
+{
+    wp<Track> weak(this); // never create a strong ref from the dtor
+    mState = TERMINATED;
+    mAudioFlinger->removeTrack(weak, mName);
+}
+
+void AudioFlinger::Track::destroy()
+{
+    mAudioFlinger->destroyTrack(this);
+}
+
+void AudioFlinger::Track::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "  %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
+            mName - AudioMixer::TRACK0,
+            mClient->pid(),
+            mStreamType,
+            mFormat,
+            mChannelCount,
+            mBufferCount,
+            mState,
+            mMute,
+            mFillingUpStatus,
+            mCblk->sampleRate,
+            mCblk->volume[0],
+            mCblk->volume[1],
+            mCblk->server,
+            mCblk->user);
+}
+
+status_t AudioFlinger::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+     audio_track_cblk_t* cblk = this->cblk();
+     uint32_t u = cblk->user;
+     uint32_t s = cblk->server;
+     
+     // Check if last stepServer failed, try to step now 
+     if (mFlags & TrackBase::STEPSERVER_FAILED) {
+         if (!step())  goto getNextBuffer_exit;
+         LOGV("stepServer recovered");
+         mFlags &= ~TrackBase::STEPSERVER_FAILED;
+     }
+
+     if (LIKELY(u > s)) {
+         int index = s & audio_track_cblk_t::BUFFER_MASK;
+         buffer->raw = getBuffer(index);
+         buffer->frameCount = mAudioFlinger->frameCount();
+         return NO_ERROR;
+     }
+getNextBuffer_exit:
+     buffer->raw = 0;
+     buffer->frameCount = 0;
+     return NOT_ENOUGH_DATA;
+}
+
+bool AudioFlinger::Track::isReady(uint32_t u, int32_t s) const {
+    if (mFillingUpStatus != FS_FILLING) return true;
+    const uint32_t u_seq = u & audio_track_cblk_t::SEQUENCE_MASK;
+    const uint32_t u_buf = u & audio_track_cblk_t::BUFFER_MASK;
+    const uint32_t s_seq = s & audio_track_cblk_t::SEQUENCE_MASK;
+    const uint32_t s_buf = s & audio_track_cblk_t::BUFFER_MASK;
+    if (u_seq > s_seq && u_buf == s_buf) {
+        mFillingUpStatus = FS_FILLED;
+        return true;
+    }
+    return false;
+}
+
+status_t AudioFlinger::Track::start()
+{
+    LOGV("start(%d)", mName);
+    mAudioFlinger->addTrack(this);
+    return NO_ERROR;
+}
+
+void AudioFlinger::Track::stop()
+{
+    LOGV("stop(%d)", mName);
+    Mutex::Autolock _l(mAudioFlinger->mLock);
+    if (mState > STOPPED) {
+        mState = STOPPED;
+        // If the track is not active (PAUSED and buffers full), flush buffers  
+        if (mAudioFlinger->mActiveTracks.indexOf(this) < 0) {
+            reset();
+        }
+        LOGV("(> STOPPED) => STOPPED (%d)", mName);
+    }
+}
+
+void AudioFlinger::Track::pause()
+{
+    LOGV("pause(%d)", mName);
+    Mutex::Autolock _l(mAudioFlinger->mLock);
+    if (mState == ACTIVE || mState == RESUMING) {
+        mState = PAUSING;
+        LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+    }
+}
+
+void AudioFlinger::Track::flush()
+{
+    LOGV("flush(%d)", mName);
+    Mutex::Autolock _l(mAudioFlinger->mLock);
+    if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+        return;
+    }
+    // No point remaining in PAUSED state after a flush => go to
+    // STOPPED state
+    mState = STOPPED;
+
+    // NOTE: reset() will reset cblk->user and cblk->server with
+    // the risk that at the same time, the AudioMixer is trying to read
+    // data. In this case, getNextBuffer() would return a NULL pointer
+    // as audio buffer => the AudioMixer code MUST always test that pointer 
+    // returned by getNextBuffer() is not NULL! 
+    reset();
+}
+
+void AudioFlinger::Track::reset()
+{
+    TrackBase::reset();
+    mFillingUpStatus = FS_FILLING;
+}
+
+void AudioFlinger::Track::mute(bool muted)
+{
+    mMute = muted;
+}
+
+void AudioFlinger::Track::setVolume(float left, float right)
+{
+    mVolume[0] = left;
+    mVolume[1] = right;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::Track>& track)
+    : BnAudioTrack(),
+      mTrack(track)
+{
+}
+
+AudioFlinger::TrackHandle::~TrackHandle() {
+    // just stop the track on deletion, associated resources
+    // will be freed from the main thread once all pending buffers have
+    // been played. Unless it's not in the active track list, in which
+    // case we free everything now...
+    mTrack->destroy();
+}
+
+status_t AudioFlinger::TrackHandle::start() {
+    return mTrack->start();
+}
+
+void AudioFlinger::TrackHandle::stop() {
+    mTrack->stop();
+}
+
+void AudioFlinger::TrackHandle::flush() {
+    mTrack->flush();
+}
+
+void AudioFlinger::TrackHandle::mute(bool e) {
+    mTrack->mute(e);
+}
+
+void AudioFlinger::TrackHandle::pause() {
+    mTrack->pause();
+}
+
+void AudioFlinger::TrackHandle::setVolume(float left, float right) {
+    mTrack->setVolume(left, right);
+}
+
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+    return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioTrack::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+sp<AudioFlinger::AudioRecordThread> AudioFlinger::audioRecordThread()
+{
+    Mutex::Autolock _l(mLock);
+    return mAudioRecordThread;
+}
+
+void AudioFlinger::endRecord()
+{
+    Mutex::Autolock _l(mLock);
+    mAudioRecordThread.clear();
+}
+
+sp<IAudioRecord> AudioFlinger::openRecord(
+        pid_t pid,
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int bufferCount,
+        uint32_t flags)
+{
+    sp<AudioRecordThread> thread;
+    sp<RecordTrack> recordTrack;
+    sp<RecordHandle> recordHandle;
+    sp<Client> client;
+    wp<Client> wclient;
+    AudioStreamIn* input = 0;
+
+    // check calling permissions
+    if (!recordingAllowed()) {
+        goto Exit;
+    }
+
+    if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+        LOGE("invalid stream type");
+        goto Exit;
+    }
+
+    if (sampleRate > MAX_SAMPLE_RATE) {
+        LOGE("Sample rate out of range");
+        goto Exit;
+    }
+
+    if (mSampleRate == 0) {
+        LOGE("Audio driver not initialized");
+        goto Exit;
+    }
+
+    // Create audio thread - take mutex to prevent race condition
+    {
+        Mutex::Autolock _l(mLock);
+        if (mAudioRecordThread != 0) {
+            LOGE("Record channel already open");
+            goto Exit;
+        }
+        thread = new AudioRecordThread(this);
+        mAudioRecordThread = thread;
+    }
+    // It's safe to release the mutex here since the client doesn't get a
+    // handle until we return from this call
+
+    // open driver, initialize h/w
+    input = mAudioHardware->openInputStream(
+            AudioSystem::PCM_16_BIT, channelCount, sampleRate);
+    if (!input) {
+        LOGE("Error opening input stream");
+        mAudioRecordThread.clear();
+        goto Exit;
+    }
+
+    // add client to list
+    {
+        Mutex::Autolock _l(mLock);
+        wclient = mClients.valueFor(pid);
+        if (wclient != NULL) {
+            client = wclient.promote();
+        } else {
+            client = new Client(this, pid);
+            mClients.add(pid, client);
+        }
+    }
+
+    // create new record track and pass to record thread
+    recordTrack = new RecordTrack(this, client, streamType, sampleRate,
+            format, channelCount, bufferCount, input->bufferSize());
+
+    // spin up record thread
+    thread->open(recordTrack, input);
+    thread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
+
+    // return to handle to client
+    recordHandle = new RecordHandle(recordTrack);
+
+Exit:
+    return recordHandle;
+}
+
+status_t AudioFlinger::startRecord() {
+    sp<AudioRecordThread> t = audioRecordThread();
+    if (t == 0) return NO_INIT;
+    return t->start();
+}
+
+void AudioFlinger::stopRecord() {
+    sp<AudioRecordThread> t = audioRecordThread();
+    if (t != 0) t->stop();
+}
+
+void AudioFlinger::exitRecord()
+{
+    sp<AudioRecordThread> t = audioRecordThread();
+    if (t != 0) t->exit();
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordTrack::RecordTrack(
+            const sp<AudioFlinger>& audioFlinger,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int bufferCount,
+            int bufferSize)
+    :   TrackBase(audioFlinger, client, streamType, sampleRate, format,
+            channelCount, bufferCount, bufferSize),
+            mOverflow(false)
+{
+}
+
+AudioFlinger::RecordTrack::~RecordTrack()
+{
+    mAudioFlinger->audioMixer().deleteTrackName(mName);
+    mAudioFlinger->exitRecord();
+}
+
+status_t AudioFlinger::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+     audio_track_cblk_t* cblk = this->cblk();
+     const uint32_t u_seq = cblk->user & audio_track_cblk_t::SEQUENCE_MASK;
+     const uint32_t u_buf = cblk->user & audio_track_cblk_t::BUFFER_MASK;
+     const uint32_t s_seq = cblk->server & audio_track_cblk_t::SEQUENCE_MASK;
+     const uint32_t s_buf = cblk->server & audio_track_cblk_t::BUFFER_MASK;
+     
+     // Check if last stepServer failed, try to step now 
+     if (mFlags & TrackBase::STEPSERVER_FAILED) {
+         if (!step())  goto getNextBuffer_exit;
+         LOGV("stepServer recovered");
+         mFlags &= ~TrackBase::STEPSERVER_FAILED;
+     }
+
+     if (LIKELY(s_seq == u_seq || s_buf != u_buf)) {
+         buffer->raw = getBuffer(s_buf);
+         buffer->frameCount = mAudioFlinger->frameCount();
+         return NO_ERROR;
+     }
+
+getNextBuffer_exit:     
+     buffer->raw = 0;
+     buffer->frameCount = 0;
+     return NOT_ENOUGH_DATA;
+}
+
+status_t AudioFlinger::RecordTrack::start()
+{
+    return mAudioFlinger->startRecord();
+}
+
+void AudioFlinger::RecordTrack::stop()
+{
+    mAudioFlinger->stopRecord();
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordTrack>& recordTrack)
+    : BnAudioRecord(),
+    mRecordTrack(recordTrack)
+{
+}
+
+AudioFlinger::RecordHandle::~RecordHandle() {}
+
+status_t AudioFlinger::RecordHandle::start() {
+    LOGV("RecordHandle::start()");
+    return mRecordTrack->start();
+}
+
+void AudioFlinger::RecordHandle::stop() {
+    LOGV("RecordHandle::stop()");
+    mRecordTrack->stop();
+}
+
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+    return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioRecord::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioRecordThread::AudioRecordThread(const sp<AudioFlinger>& audioFlinger) :
+    mAudioFlinger(audioFlinger),
+    mRecordTrack(0),
+    mInput(0),
+    mActive(false)
+{
+}
+
+AudioFlinger::AudioRecordThread::~AudioRecordThread()
+{
+}
+
+bool AudioFlinger::AudioRecordThread::threadLoop()
+{
+    LOGV("AudioRecordThread: start record loop");
+
+    // start recording
+    while (!exitPending()) {
+        if (!mActive) {
+            mLock.lock();
+            if (!mActive && !exitPending()) {
+                LOGV("AudioRecordThread: loop stopping");
+                mWaitWorkCV.wait(mLock);
+                LOGV("AudioRecordThread: loop starting");
+            }
+            mLock.unlock();
+        } else {
+            // promote strong ref so track isn't deleted while we access it
+            sp<RecordTrack> t = mRecordTrack.promote();
+
+            // if we lose the weak reference, client is gone.
+            if (t == 0) {
+                LOGV("AudioRecordThread: client deleted track");
+                break;
+            }
+
+            if (LIKELY(t->getNextBuffer(&mBuffer) == NO_ERROR)) {
+                if (mInput->read(mBuffer.raw, t->mBufferSize) < 0) {
+                    LOGE("Error reading audio input");
+                    sleep(1);
+                }
+                t->releaseBuffer(&mBuffer);
+            }
+
+            // client isn't retrieving buffers fast enough
+            else {
+                if (!t->setOverflow())
+                    LOGW("AudioRecordThread: buffer overflow");
+            }
+        }
+    };
+
+    // close hardware
+    close();
+
+    // delete this object - no more data references after this call
+    mAudioFlinger->endRecord();
+    return false;
+}
+
+status_t AudioFlinger::AudioRecordThread::open(const sp<RecordTrack>& recordTrack, AudioStreamIn *input) {
+    LOGV("AudioRecordThread::open");
+    // check for record channel already open
+    AutoMutex lock(&mLock);
+    if (mRecordTrack != NULL) {
+        LOGE("Record channel already open");
+        return ALREADY_EXISTS;
+    }
+    mRecordTrack = recordTrack;
+    mInput = input;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::AudioRecordThread::start()
+{
+    LOGV("AudioRecordThread::start");
+    AutoMutex lock(&mLock);
+    if (mActive) return -EBUSY;
+
+    sp<RecordTrack> t = mRecordTrack.promote();
+    if (t == 0) return UNKNOWN_ERROR;
+
+    // signal thread to start
+    LOGV("Signal record thread");
+    mActive = true;
+    mWaitWorkCV.signal();
+    return NO_ERROR;
+}
+
+void AudioFlinger::AudioRecordThread::stop() {
+    LOGV("AudioRecordThread::stop");
+    AutoMutex lock(&mLock);
+    if (mActive) {
+        mActive = false;
+        mWaitWorkCV.signal();
+    }
+}
+
+void AudioFlinger::AudioRecordThread::exit()
+{
+    LOGV("AudioRecordThread::exit");
+    AutoMutex lock(&mLock);
+    requestExit();
+    mWaitWorkCV.signal();
+}
+
+
+status_t AudioFlinger::AudioRecordThread::close()
+{
+    LOGV("AudioRecordThread::close");
+    AutoMutex lock(&mLock);
+    if (!mInput) return NO_INIT;
+    delete mInput;
+    mInput = 0;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::onTransact(
+        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioFlinger::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+void AudioFlinger::instantiate() {
+    defaultServiceManager()->addService(
+            String16("media.audio_flinger"), new AudioFlinger());
+}
+
+}; // namespace android